From 34226c4239861654c69888e6e98973b2cc908c0b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 4 Jan 2006 12:14:23 +0100 Subject: [ALSA] pcxhr - Suppress debug messages Modules: Digigram PCXHR driver Suppress debug messages. Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index fa0d27e2c79beb..fdc652c6992d02 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1176,7 +1176,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id, struct pt_regs *regs) mgr->dsp_time_last = dsp_time_new; if (timer_toggle == mgr->timer_toggle) - snd_printk(KERN_ERR "ERROR TIMER TOGGLE\n"); + snd_printdd("ERROR TIMER TOGGLE\n"); mgr->timer_toggle = timer_toggle; reg &= ~PCXHR_IRQ_TIMER; -- cgit 1.2.3-korg From e03173fce2f2c233b755f2d668d6d4247a717453 Mon Sep 17 00:00:00 2001 From: Brent Cook Date: Tue, 10 Jan 2006 14:27:20 +0100 Subject: [ALSA] Add support for EDIROL UM-3ex Modules: USB generic driver This is my naive attempt at adding ALSA device support. The attached patch provides support for the EDIROL UM-3ex. This is a 3-port USB midi interface with a built-in USB hub and the ability to chain 2 other UM-3x's in a master-slave configuration. I only have one, so I do not know how this works in practice. Though this is a 3-port device, I had to throw in that 4th 'Control' interface to the definition in order to make the 3rd port work. If I set in/out_cables to 0x000b, a 3rd interface appears on the driver, but it does nothing. Changing it to 0x000f allows the 3rd interface to work, but of course interface 4 does not work because it does not exist. Signed-off-by: Takashi Iwai --- sound/usb/usbmidi.c | 2 ++ sound/usb/usbquirks.h | 22 ++++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index f15b021c3ce89f..905e33e6068120 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1092,6 +1092,8 @@ static struct { { USB_ID(0x086a, 0x0001), 8, "%s Broadcast" }, { USB_ID(0x086a, 0x0002), 8, "%s Broadcast" }, { USB_ID(0x086a, 0x0003), 4, "%s Broadcast" }, + /* Edirol UM-3ex */ + { USB_ID(0x0582, 0x009a), 3, "%s Control" }, }; static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 6190ada00e3890..8f348611bb0082 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -203,6 +203,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + USB_DEVICE(0x0582, 0x009a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UM-3ex", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x000f, + .in_cables = 0x000f + } + }, + { + .ifnum = -1 + } + } + } +}, { USB_DEVICE(0x0582, 0x0002), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit 1.2.3-korg From 7c79b7682ee6318a8e47173b158846ed0edd8f1a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 10 Jan 2006 18:56:23 +0100 Subject: [ALSA] usb-audio: cosmetic changes Modules: USB generic driver Move some entries around to keep them sorted by ID. Signed-off-by: Clemens Ladisch --- sound/usb/usbmidi.c | 4 ++-- sound/usb/usbquirks.h | 52 +++++++++++++++++++++------------------------------ 2 files changed, 23 insertions(+), 33 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 905e33e6068120..7f7e371c5df19b 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1082,6 +1082,8 @@ static struct { { USB_ID(0x0582, 0x004d), 0, "%s MIDI" }, { USB_ID(0x0582, 0x004d), 1, "%s 1" }, { USB_ID(0x0582, 0x004d), 2, "%s 2" }, + /* Edirol UM-3EX */ + { USB_ID(0x0582, 0x009a), 3, "%s Control" }, /* M-Audio MidiSport 8x8 */ { USB_ID(0x0763, 0x1031), 8, "%s Control" }, { USB_ID(0x0763, 0x1033), 8, "%s Control" }, @@ -1092,8 +1094,6 @@ static struct { { USB_ID(0x086a, 0x0001), 8, "%s Broadcast" }, { USB_ID(0x086a, 0x0002), 8, "%s Broadcast" }, { USB_ID(0x086a, 0x0003), 4, "%s Broadcast" }, - /* Edirol UM-3ex */ - { USB_ID(0x0582, 0x009a), 3, "%s Control" }, }; static void snd_usbmidi_init_substream(struct snd_usb_midi* umidi, diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 8f348611bb0082..381d9a8ccee1b8 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -203,28 +203,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -{ - USB_DEVICE(0x0582, 0x009a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "EDIROL", - .product_name = "UM-3ex", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x000f, - .in_cables = 0x000f - } - }, - { - .ifnum = -1 - } - } - } -}, { USB_DEVICE(0x0582, 0x0002), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { @@ -1112,6 +1090,27 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, + /* TODO: add Edirol UA-101 support */ + /* TODO: add Roland G-70 support */ + /* TODO: add Roland V-SYNTH XT support */ + /* TODO: add BOSS GT-PRO support */ + /* TODO: add Edirol PC-50 support */ + /* TODO: add Edirol PC-80 support */ + /* TODO: add Edirol UA-1EX support */ +{ + USB_DEVICE(0x0582, 0x009a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "UM-3EX", + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x000f, + .in_cables = 0x000f + } + } +}, + /* TODO: add Edirol MD-P1 support */ /* Guillemot devices */ { @@ -1133,15 +1132,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - /* TODO: add Edirol UA-101 support */ - /* TODO: add Roland G-70 support */ - /* TODO: add Roland V-SYNTH XT support */ - /* TODO: add BOSS GT-PRO support */ - /* TODO: add Edirol PC-50 support */ - /* TODO: add Edirol PC-80 support */ - /* TODO: add Edirol UA-1EX support */ - /* TODO: add Edirol UM-3 support */ - /* TODO: add Edirol MD-P1 support */ /* Midiman/M-Audio devices */ { -- cgit 1.2.3-korg From 9568f461e50c023e45ec702027bb1a5f60e443dc Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 12 Jan 2006 08:19:21 +0100 Subject: [ALSA] usb-audio: factor out packet size calculation code Modules: USB generic driver Move the common packet size calculation code from prepare_startup_playback_urb() and prepare_playback_urb() to a new function. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 34 ++++++++++++++-------------------- 1 file changed, 14 insertions(+), 20 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index d5013383fad71d..f5aadb001986f2 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -475,6 +475,18 @@ static int retire_playback_sync_urb_hs(struct snd_usb_substream *subs, return 0; } +/* determine the number of frames in the next packet */ +static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) +{ + if (subs->fill_max) + return subs->maxframesize; + else { + subs->phase = (subs->phase & 0xffff) + + (subs->freqm << subs->datainterval); + return min(subs->phase >> 16, subs->maxframesize); + } +} + /* * Prepare urb for streaming before playback starts. * @@ -492,16 +504,7 @@ static int prepare_startup_playback_urb(struct snd_usb_substream *subs, urb->dev = ctx->subs->dev; urb->number_of_packets = subs->packs_per_ms; for (i = 0; i < subs->packs_per_ms; ++i) { - /* calculate the size of a packet */ - if (subs->fill_max) - counts = subs->maxframesize; /* fixed */ - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - counts = subs->phase >> 16; - if (counts > subs->maxframesize) - counts = subs->maxframesize; - } + counts = snd_usb_audio_next_packet_size(subs); urb->iso_frame_desc[i].offset = offs * stride; urb->iso_frame_desc[i].length = counts * stride; offs += counts; @@ -538,16 +541,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { - /* calculate the size of a packet */ - if (subs->fill_max) - counts = subs->maxframesize; /* fixed */ - else { - subs->phase = (subs->phase & 0xffff) - + (subs->freqm << subs->datainterval); - counts = subs->phase >> 16; - if (counts > subs->maxframesize) - counts = subs->maxframesize; - } + counts = snd_usb_audio_next_packet_size(subs); /* set up descriptor */ urb->iso_frame_desc[i].offset = offs * stride; urb->iso_frame_desc[i].length = counts * stride; -- cgit 1.2.3-korg From 15f500a6994e552b7772d8b8459ee3114c47897d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jan 2006 11:43:49 +0100 Subject: [ALSA] ens1370 - Fix resume Modules: ENS1370/1+ driver Fix resume of ens1370 driver. Signed-off-by: Takashi Iwai --- sound/pci/ens1370.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 55aaf110331a42..bee382995fe995 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -2061,6 +2061,13 @@ static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state) #ifdef CHIP1371 snd_ac97_suspend(ensoniq->u.es1371.ac97); #else + /* try to reset AK4531 */ + outw(ES_1370_CODEC_WRITE(AK4531_RESET, 0x02), ES_REG(ensoniq, 1370_CODEC)); + inw(ES_REG(ensoniq, 1370_CODEC)); + udelay(100); + outw(ES_1370_CODEC_WRITE(AK4531_RESET, 0x03), ES_REG(ensoniq, 1370_CODEC)); + inw(ES_REG(ensoniq, 1370_CODEC)); + udelay(100); snd_ak4531_suspend(ensoniq->u.es1370.ak4531); #endif pci_set_power_state(pci, PCI_D3hot); -- cgit 1.2.3-korg From 84a43bd523a63f1b53fd734c3798d71b7b53f123 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jan 2006 11:47:32 +0100 Subject: [ALSA] intel8x0 - Fix/cleanup detection of codecs on SIS7012 Modules: Intel8x0 driver Fix the detection of tertriary codec on SIS7012, including clean-ups of relevant codes. Signed-off-by: Takashi Iwai --- sound/pci/intel8x0.c | 138 +++++++++++++++++++++++++++++++-------------------- 1 file changed, 84 insertions(+), 54 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 174237f4a22cb2..da024ffe96c252 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -178,6 +178,8 @@ DEFINE_REGSET(SP, 0x60); /* SPDIF out */ #define ICH_SAMPLE_CAP 0x00c00000 /* ICH4: sample capability bits (RO) */ #define ICH_SAMPLE_16_20 0x00400000 /* ICH4: 16- and 20-bit samples */ #define ICH_MULTICHAN_CAP 0x00300000 /* ICH4: multi-channel capability bits (RO) */ +#define ICH_SIS_TRI 0x00080000 /* SIS: tertiary resume irq */ +#define ICH_SIS_TCR 0x00040000 /* SIS: tertiary codec ready */ #define ICH_MD3 0x00020000 /* modem power down semaphore */ #define ICH_AD3 0x00010000 /* audio power down semaphore */ #define ICH_RCS 0x00008000 /* read completion status */ @@ -398,6 +400,10 @@ struct intel8x0 { struct snd_ac97_bus *ac97_bus; struct snd_ac97 *ac97[3]; unsigned int ac97_sdin[3]; + unsigned int max_codecs, ncodecs; + unsigned int *codec_bit; + unsigned int codec_isr_bits; + unsigned int codec_ready_bits; spinlock_t reg_lock; @@ -516,18 +522,6 @@ static void iaputword(struct intel8x0 *chip, u32 offset, u16 val) * access to AC97 codec via normal i/o (for ICH and SIS7012) */ -/* return the GLOB_STA bit for the corresponding codec */ -static unsigned int get_ich_codec_bit(struct intel8x0 *chip, unsigned int codec) -{ - static unsigned int codec_bit[3] = { - ICH_PCR, ICH_SCR, ICH_TCR - }; - snd_assert(codec < 3, return ICH_PCR); - if (chip->device_type == DEVICE_INTEL_ICH4) - codec = chip->ac97_sdin[codec]; - return codec_bit[codec]; -} - static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int codec) { int time; @@ -537,9 +531,9 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code if (chip->in_sdin_init) { /* we don't know the ready bit assignment at the moment */ /* so we check any */ - codec = ICH_PCR | ICH_SCR | ICH_TCR; + codec = chip->codec_isr_bits; } else { - codec = get_ich_codec_bit(chip, codec); + codec = chip->codec_bit[chip->ac97_sdin[codec]]; } /* codec ready ? */ @@ -596,7 +590,7 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, if ((tmp = igetdword(chip, ICHREG(GLOB_STA))) & ICH_RCS) { /* reset RCS and preserve other R/WC bits */ iputdword(chip, ICHREG(GLOB_STA), tmp & - ~(ICH_SRI|ICH_PRI|ICH_TRI|ICH_GSCI)); + ~(chip->codec_ready_bits | ICH_GSCI)); if (! chip->in_ac97_init) snd_printk(KERN_ERR "codec_read %d: read timeout for register 0x%x\n", ac97->num, reg); res = 0xffff; @@ -605,7 +599,8 @@ static unsigned short snd_intel8x0_codec_read(struct snd_ac97 *ac97, return res; } -static void snd_intel8x0_codec_read_test(struct intel8x0 *chip, unsigned int codec) +static void __devinit snd_intel8x0_codec_read_test(struct intel8x0 *chip, + unsigned int codec) { unsigned int tmp; @@ -614,7 +609,7 @@ static void snd_intel8x0_codec_read_test(struct intel8x0 *chip, unsigned int cod if ((tmp = igetdword(chip, ICHREG(GLOB_STA))) & ICH_RCS) { /* reset RCS and preserve other R/WC bits */ iputdword(chip, ICHREG(GLOB_STA), tmp & - ~(ICH_SRI|ICH_PRI|ICH_TRI|ICH_GSCI)); + ~(chip->codec_ready_bits | ICH_GSCI)); } } } @@ -2078,23 +2073,24 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, if (chip->device_type != DEVICE_ALI) { glob_sta = igetdword(chip, ICHREG(GLOB_STA)); ops = &standard_bus_ops; - if (chip->device_type == DEVICE_INTEL_ICH4) { - codecs = 0; - if (glob_sta & ICH_PCR) - codecs++; - if (glob_sta & ICH_SCR) - codecs++; - if (glob_sta & ICH_TCR) - codecs++; - chip->in_sdin_init = 1; - for (i = 0; i < codecs; i++) { - snd_intel8x0_codec_read_test(chip, i); - chip->ac97_sdin[i] = igetbyte(chip, ICHREG(SDM)) & ICH_LDI_MASK; - } - chip->in_sdin_init = 0; - } else { - codecs = glob_sta & ICH_SCR ? 2 : 1; + chip->in_sdin_init = 1; + codecs = 0; + for (i = 0; i < chip->max_codecs; i++) { + if (! (glob_sta & chip->codec_bit[i])) + continue; + if (chip->device_type == DEVICE_INTEL_ICH4) { + snd_intel8x0_codec_read_test(chip, codecs); + chip->ac97_sdin[codecs] = + igetbyte(chip, ICHREG(SDM)) & ICH_LDI_MASK; + snd_assert(chip->ac97_sdin[codecs] < 3, + chip->ac97_sdin[codecs] = 0); + } else + chip->ac97_sdin[codecs] = i; + codecs++; } + chip->in_sdin_init = 0; + if (! codecs) + codecs = 1; } else { ops = &ali_bus_ops; codecs = 1; @@ -2120,6 +2116,7 @@ static int __devinit snd_intel8x0_mixer(struct intel8x0 *chip, int ac97_clock, else pbus->dra = 1; chip->ac97_bus = pbus; + chip->ncodecs = codecs; ac97.pci = chip->pci; for (i = 0; i < codecs; i++) { @@ -2264,7 +2261,7 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) end_time = jiffies + HZ; do { status = igetdword(chip, ICHREG(GLOB_STA)) & - (ICH_PCR | ICH_SCR | ICH_TCR); + chip->codec_isr_bits; if (status) break; schedule_timeout_uninterruptible(1); @@ -2276,32 +2273,27 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) return -EIO; } - if (chip->device_type == DEVICE_INTEL_ICH4) - /* ICH4 can have three codecs */ - nstatus = ICH_PCR | ICH_SCR | ICH_TCR; - else - /* others up to two codecs */ - nstatus = ICH_PCR | ICH_SCR; - /* wait for other codecs ready status. */ end_time = jiffies + HZ / 4; - while (status != nstatus && time_after_eq(end_time, jiffies)) { + while (status != chip->codec_isr_bits && + time_after_eq(end_time, jiffies)) { schedule_timeout_uninterruptible(1); - status |= igetdword(chip, ICHREG(GLOB_STA)) & nstatus; + status |= igetdword(chip, ICHREG(GLOB_STA)) & + chip->codec_isr_bits; } } else { /* resume phase */ int i; status = 0; - for (i = 0; i < 3; i++) + for (i = 0; i < chip->ncodecs; i++) if (chip->ac97[i]) - status |= get_ich_codec_bit(chip, i); + status |= chip->codec_bit[chip->ac97_sdin[i]]; /* wait until all the probed codecs are ready */ end_time = jiffies + HZ; do { nstatus = igetdword(chip, ICHREG(GLOB_STA)) & - (ICH_PCR | ICH_SCR | ICH_TCR); + chip->codec_isr_bits; if (status == nstatus) break; schedule_timeout_uninterruptible(1); @@ -2447,7 +2439,7 @@ static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state) } } } - for (i = 0; i < 3; i++) + for (i = 0; i < chip->ncodecs; i++) snd_ac97_suspend(chip->ac97[i]); if (chip->device_type == DEVICE_INTEL_ICH4) chip->sdm_saved = igetbyte(chip, ICHREG(SDM)); @@ -2488,7 +2480,7 @@ static int intel8x0_resume(struct pci_dev *pci) if (chip->fix_nocache) fill_nocache(chip->bdbars.area, chip->bdbars.bytes, 1); - for (i = 0; i < 3; i++) + for (i = 0; i < chip->ncodecs; i++) snd_ac97_resume(chip->ac97[i]); /* refill nocache */ @@ -2619,12 +2611,20 @@ static void snd_intel8x0_proc_read(struct snd_info_entry * entry, snd_iprintf(buffer, "Global status : 0x%08x\n", tmp); if (chip->device_type == DEVICE_INTEL_ICH4) snd_iprintf(buffer, "SDM : 0x%08x\n", igetdword(chip, ICHREG(SDM))); - snd_iprintf(buffer, "AC'97 codecs ready :%s%s%s%s\n", - tmp & ICH_PCR ? " primary" : "", - tmp & ICH_SCR ? " secondary" : "", - tmp & ICH_TCR ? " tertiary" : "", - (tmp & (ICH_PCR | ICH_SCR | ICH_TCR)) == 0 ? " none" : ""); - if (chip->device_type == DEVICE_INTEL_ICH4) + snd_iprintf(buffer, "AC'97 codecs ready :"); + if (tmp & chip->codec_isr_bits) { + int i; + static const char *codecs[3] = { + "primary", "secondary", "tertiary" + }; + for (i = 0; i < chip->max_codecs; i++) + if (tmp & chip->codec_bit[i]) + snd_iprintf(buffer, " %s", codecs[i]); + } else + snd_iprintf(buffer, " none"); + snd_iprintf(buffer, "\n"); + if (chip->device_type == DEVICE_INTEL_ICH4 || + chip->device_type == DEVICE_SIS) snd_iprintf(buffer, "AC'97 codecs SDIN : %i %i %i\n", chip->ac97_sdin[0], chip->ac97_sdin[1], @@ -2653,6 +2653,13 @@ struct ich_reg_info { unsigned int offset; }; +static unsigned int ich_codec_bits[3] = { + ICH_PCR, ICH_SCR, ICH_TCR +}; +static unsigned int sis_codec_bits[3] = { + ICH_PCR, ICH_SCR, ICH_SIS_TCR +}; + static int __devinit snd_intel8x0_create(struct snd_card *card, struct pci_dev *pci, unsigned long device_type, @@ -2835,6 +2842,29 @@ static int __devinit snd_intel8x0_create(struct snd_card *card, pci_set_master(pci); synchronize_irq(chip->irq); + switch(chip->device_type) { + case DEVICE_INTEL_ICH4: + /* ICH4 can have three codecs */ + chip->max_codecs = 3; + chip->codec_bit = ich_codec_bits; + chip->codec_ready_bits = ICH_PRI | ICH_SRI | ICH_TRI; + break; + case DEVICE_SIS: + /* recent SIS7012 can have three codecs */ + chip->max_codecs = 3; + chip->codec_bit = sis_codec_bits; + chip->codec_ready_bits = ICH_PRI | ICH_SRI | ICH_SIS_TRI; + break; + default: + /* others up to two codecs */ + chip->max_codecs = 2; + chip->codec_bit = ich_codec_bits; + chip->codec_ready_bits = ICH_PRI | ICH_SRI; + break; + } + for (i = 0; i < chip->max_codecs; i++) + chip->codec_isr_bits |= chip->codec_bit[i]; + if ((err = snd_intel8x0_chip_init(chip, 1)) < 0) { snd_intel8x0_free(chip); return err; -- cgit 1.2.3-korg From 27346166a9b3b9eee586bce212502cddf9685a07 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Jan 2006 18:28:44 +0100 Subject: [ALSA] hda-intel - Add single_cmd option for debugging Modules: Documentation,HDA Intel driver Added single_cmd module option for debugging in the case CORB/RIRB doesn't work well (e.g. due to wrong irq routings). Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 13 ++++++ sound/pci/hda/hda_intel.c | 57 +++++++++++-------------- 2 files changed, 39 insertions(+), 31 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 36b511c7cade9a..cc8a701871998c 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -671,6 +671,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. model - force the model name position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size) + single_cmd - Use single immediate commands to communicate with + codecs (for debugging only) This module supports one card and autoprobe. @@ -723,6 +725,17 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. (Usually SD_LPLIB register is more accurate than the position buffer.) + NB: If you get many "azx_get_response timeout" messages at + loading, it's likely a problem of interrupts (e.g. ACPI irq + routing). Try to boot with options like "pci=noacpi". Also, you + can try "single_cmd=1" module option. This will switch the + communication method between HDA controller and codecs to the + single immediate commands instead of CORB/RIRB. Basically, the + single command mode is provided only for BIOS, and you won't get + unsolicited events, too. But, at least, this works independently + from the irq. Remember this is a last resort, and should be + avoided as much as possible... + The power-management is supported. Module snd-hdsp diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fd12b6991fe403..b3f37e7b33c047 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -53,6 +53,7 @@ static char *id = SNDRV_DEFAULT_STR1; static char *model; static int position_fix; static int probe_mask = -1; +static int single_cmd; module_param(index, int, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -64,6 +65,8 @@ module_param(position_fix, int, 0444); MODULE_PARM_DESC(position_fix, "Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)."); module_param(probe_mask, int, 0444); MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); +module_param(single_cmd, bool, 0444); +MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs (for debugging only)."); /* just for backward compatibility */ @@ -234,12 +237,6 @@ enum { #define NVIDIA_HDA_TRANSREG_ADDR 0x4e #define NVIDIA_HDA_ENABLE_COHBITS 0x0f -/* - * Use CORB/RIRB for communication from/to codecs. - * This is the way recommended by Intel (see below). - */ -#define USE_CORB_RIRB - /* */ @@ -325,6 +322,7 @@ struct azx { /* flags */ int position_fix; unsigned int initialized: 1; + unsigned int single_cmd: 1; }; /* driver types */ @@ -388,7 +386,6 @@ static char *driver_short_names[] __devinitdata = { * Interface for HD codec */ -#ifdef USE_CORB_RIRB /* * CORB / RIRB interface */ @@ -436,11 +433,7 @@ static void azx_init_cmd_io(struct azx *chip) /* set N=1, get RIRB response interrupt for new entry */ azx_writew(chip, RINTCNT, 1); /* enable rirb dma and response irq */ -#ifdef USE_CORB_RIRB azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN); -#else - azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN); -#endif chip->rirb.rp = chip->rirb.cmds = 0; } @@ -528,7 +521,6 @@ static unsigned int azx_get_response(struct hda_codec *codec) return chip->rirb.res; /* the last value */ } -#else /* * Use the single immediate command instead of CORB/RIRB for simplicity * @@ -539,13 +531,10 @@ static unsigned int azx_get_response(struct hda_codec *codec) * I left the codes, however, for debugging/testing purposes. */ -#define azx_alloc_cmd_io(chip) 0 -#define azx_init_cmd_io(chip) -#define azx_free_cmd_io(chip) - /* send a command */ -static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int para) +static int azx_single_send_cmd(struct hda_codec *codec, hda_nid_t nid, + int direct, unsigned int verb, + unsigned int para) { struct azx *chip = codec->bus->private_data; u32 val; @@ -573,7 +562,7 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, } /* receive a response */ -static unsigned int azx_get_response(struct hda_codec *codec) +static unsigned int azx_single_get_response(struct hda_codec *codec) { struct azx *chip = codec->bus->private_data; int timeout = 50; @@ -588,10 +577,6 @@ static unsigned int azx_get_response(struct hda_codec *codec) return (unsigned int)-1; } -#define azx_update_rirb(chip) - -#endif /* USE_CORB_RIRB */ - /* reset codec link */ static int azx_reset(struct azx *chip) { @@ -737,7 +722,8 @@ static void azx_init_chip(struct azx *chip) azx_int_enable(chip); /* initialize the codec command I/O */ - azx_init_cmd_io(chip); + if (! chip->single_cmd) + azx_init_cmd_io(chip); /* program the position buffer */ azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr); @@ -796,7 +782,7 @@ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { - if (status & RIRB_INT_RESPONSE) + if (! chip->single_cmd && (status & RIRB_INT_RESPONSE)) azx_update_rirb(chip); azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } @@ -913,8 +899,13 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) bus_temp.private_data = chip; bus_temp.modelname = model; bus_temp.pci = chip->pci; - bus_temp.ops.command = azx_send_cmd; - bus_temp.ops.get_response = azx_get_response; + if (chip->single_cmd) { + bus_temp.ops.command = azx_single_send_cmd; + bus_temp.ops.get_response = azx_single_get_response; + } else { + bus_temp.ops.command = azx_send_cmd; + bus_temp.ops.get_response = azx_get_response; + } if ((err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus)) < 0) return err; @@ -1316,7 +1307,8 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus, state); - azx_free_cmd_io(chip); + if (! chip->single_cmd) + azx_free_cmd_io(chip); pci_disable_device(pci); pci_save_state(pci); return 0; @@ -1354,7 +1346,8 @@ static int azx_free(struct azx *chip) azx_int_clear(chip); /* disable CORB/RIRB */ - azx_free_cmd_io(chip); + if (! chip->single_cmd) + azx_free_cmd_io(chip); /* disable position buffer */ azx_writel(chip, DPLBASE, 0); @@ -1422,6 +1415,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->driver_type = driver_type; chip->position_fix = position_fix ? position_fix : POS_FIX_POSBUF; + chip->single_cmd = single_cmd; #if BITS_PER_LONG != 64 /* Fix up base address on ULI M5461 */ @@ -1492,8 +1486,9 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, goto errout; } /* allocate CORB/RIRB */ - if ((err = azx_alloc_cmd_io(chip)) < 0) - goto errout; + if (! chip->single_cmd) + if ((err = azx_alloc_cmd_io(chip)) < 0) + goto errout; /* initialize streams */ azx_init_stream(chip); -- cgit 1.2.3-korg From 153abaa4ff0500070a282dd0868367d69388d527 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 13 Jan 2006 07:48:59 +0100 Subject: [ALSA] ymfpci: fix swapped channels in SPDIF output Modules: YMFPCI driver The routing of the effect 2/3 channels to the digital output is the opposite of the rear analog output (left/right swapped). We make the order correct for the digital output (which will make the analog rear have the channels swapped) to make AC3 output work. Signed-off-by: Clemens Ladisch --- sound/pci/ymfpci/ymfpci_main.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index a1aa74b79b3d9c..30ee53adb49413 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -536,13 +536,17 @@ static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int } } if (ypcm->output_rear) { + /* The SPDIF out channels seem to be swapped, so we have + * to swap them here, too. The rear analog out channels + * will be wrong, but otherwise AC3 would not work. + */ if (use_left) { - bank->eff2_gain = - bank->eff2_gain_end = vol_left; + bank->eff3_gain = + bank->eff3_gain_end = vol_left; } if (use_right) { - bank->eff3_gain = - bank->eff3_gain_end = vol_right; + bank->eff2_gain = + bank->eff2_gain_end = vol_right; } } } -- cgit 1.2.3-korg From 21a3479a0b606d36fe24093f70a1c27328cec286 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 13 Jan 2006 09:12:11 +0100 Subject: [ALSA] PCM midlevel & PCM OSS - make procfs & OSS plugin code optional Modules: ALSA Core,PCM Midlevel,ALSA<-OSS emulation,USB generic driver 1) The verbose procfs code for the PCM midlevel and usb audio can be removed now (more patches will follow). CONFIG_SND_VERBOSE_PROCFS 2) The PCM OSS plugin system can be also compiled optionaly. CONFIG_SND_PCM_OSS_PLUGINS Signed-off-by: Jaroslav Kysela --- include/sound/pcm_oss.h | 2 ++ sound/core/Kconfig | 18 ++++++++++++++++++ sound/core/oss/copy.c | 5 +++++ sound/core/oss/io.c | 5 +++++ sound/core/oss/linear.c | 5 +++++ sound/core/oss/mulaw.c | 5 +++++ sound/core/oss/pcm_oss.c | 20 ++++++++++++++++++-- sound/core/oss/pcm_plugin.c | 5 +++++ sound/core/oss/pcm_plugin.h | 10 ++++++++++ sound/core/oss/rate.c | 5 +++++ sound/core/oss/route.c | 5 +++++ sound/core/pcm.c | 4 +++- sound/usb/usbaudio.c | 13 +++++++++++-- 13 files changed, 97 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index fddaddde47b4f1..d6ec886637db38 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -56,8 +56,10 @@ struct snd_pcm_oss_runtime { size_t mmap_bytes; char *buffer; /* vmallocated period */ size_t buffer_used; /* used length from period buffer */ +#ifdef CONFIG_SND_PCM_OSS_PLUGINS struct snd_pcm_plugin *plugin_first; struct snd_pcm_plugin *plugin_last; +#endif unsigned int prev_hw_ptr_interrupt; }; diff --git a/sound/core/Kconfig b/sound/core/Kconfig index f79755f77a81f5..9dd121bb563808 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -73,6 +73,15 @@ config SND_PCM_OSS To compile this driver as a module, choose M here: the module will be called snd-pcm-oss. +config SND_PCM_OSS_PLUGINS + bool "OSS PCM (digital audio) API - Include plugin system" + depends on SND_PCM_OSS + default y + help + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + config SND_SEQUENCER_OSS bool "OSS Sequencer API" depends on SND && SND_SEQUENCER @@ -130,6 +139,15 @@ config SND_SUPPORT_OLD_API Say Y here to support the obsolete ALSA PCM API (ver.0.9.0 rc3 or older). +config SND_VERBOSE_PROCFS + bool "Verbose procfs contents" + depends on SND + default y + help + Say Y here to include code for verbose procfs contents (provides + usefull information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. + config SND_VERBOSE_PRINTK bool "Verbose printk" depends on SND diff --git a/sound/core/oss/copy.c b/sound/core/oss/copy.c index d6a04c2d5a75a7..7c008c4c0dd135 100644 --- a/sound/core/oss/copy.c +++ b/sound/core/oss/copy.c @@ -20,6 +20,9 @@ */ #include + +#ifdef SND_PCM_OSS_PLUGINS + #include #include #include @@ -85,3 +88,5 @@ int snd_pcm_plugin_build_copy(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } + +#endif diff --git a/sound/core/oss/io.c b/sound/core/oss/io.c index 322702e05f3eca..b6e7ce30e5a35d 100644 --- a/sound/core/oss/io.c +++ b/sound/core/oss/io.c @@ -20,6 +20,9 @@ */ #include + +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include #include #include @@ -132,3 +135,5 @@ int snd_pcm_plugin_build_io(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } + +#endif diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index 8cbfa415ce409f..ef331230b3a6ba 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -21,6 +21,9 @@ */ #include + +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include #include #include @@ -156,3 +159,5 @@ int snd_pcm_plugin_build_linear(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } + +#endif diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 14f5578ec7a784..ed12c81fcf1b94 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -22,6 +22,9 @@ */ #include + +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include #include #include @@ -306,3 +309,5 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } + +#endif diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 7fd072392c7e89..bc24d028f5181e 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -78,6 +78,7 @@ static inline void snd_leave_user(mm_segment_t fs) set_fs(fs); } +#ifdef CONFIG_SND_PCM_OSS_PLUGINS static int snd_pcm_oss_plugin_clear(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -122,6 +123,7 @@ int snd_pcm_plugin_append(struct snd_pcm_plugin *plugin) } return 0; } +#endif /* CONFIG_SND_PCM_OSS_PLUGINS */ static long snd_pcm_oss_bytes(struct snd_pcm_substream *substream, long frames) { @@ -412,6 +414,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) oss_frame_size = snd_pcm_format_physical_width(params_format(params)) * params_channels(params) / 8; +#ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_oss_plugin_clear(substream); if (!direct) { /* add necessary plugins */ @@ -441,6 +444,7 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) } } } +#endif err = snd_pcm_oss_period_size(substream, params, sparams); if (err < 0) @@ -498,11 +502,13 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream) runtime->oss.periods = params_periods(sparams); oss_period_size = snd_pcm_plug_client_size(substream, params_period_size(sparams)); snd_assert(oss_period_size >= 0, err = -EINVAL; goto failure); +#ifdef CONFIG_SND_PCM_OSS_PLUGINS if (runtime->oss.plugin_first) { err = snd_pcm_plug_alloc(substream, oss_period_size); if (err < 0) goto failure; } +#endif oss_period_size *= oss_frame_size; oss_buffer_size = oss_period_size * runtime->oss.periods; @@ -784,6 +790,7 @@ static ssize_t snd_pcm_oss_write2(struct snd_pcm_substream *substream, const cha { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t frames, frames1; +#ifdef CONFIG_SND_PCM_OSS_PLUGINS if (runtime->oss.plugin_first) { struct snd_pcm_plugin_channel *channels; size_t oss_frame_bytes = (runtime->oss.plugin_first->src_width * runtime->oss.plugin_first->src_format.channels) / 8; @@ -800,7 +807,9 @@ static ssize_t snd_pcm_oss_write2(struct snd_pcm_substream *substream, const cha if (frames1 <= 0) return frames1; bytes = frames1 * oss_frame_bytes; - } else { + } else +#endif + { frames = bytes_to_frames(runtime, bytes); frames1 = snd_pcm_oss_write3(substream, buf, frames, in_kernel); if (frames1 <= 0) @@ -871,6 +880,7 @@ static ssize_t snd_pcm_oss_read2(struct snd_pcm_substream *substream, char *buf, { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_sframes_t frames, frames1; +#ifdef CONFIG_SND_PCM_OSS_PLUGINS char __user *final_dst = (char __user *)buf; if (runtime->oss.plugin_first) { struct snd_pcm_plugin_channel *channels; @@ -887,7 +897,9 @@ static ssize_t snd_pcm_oss_read2(struct snd_pcm_substream *substream, char *buf, bytes = frames1 * oss_frame_bytes; if (!in_kernel && copy_to_user(final_dst, buf, bytes)) return -EFAULT; - } else { + } else +#endif + { frames = bytes_to_frames(runtime, bytes); frames1 = snd_pcm_oss_read3(substream, buf, frames, in_kernel); if (frames1 <= 0) @@ -1692,7 +1704,9 @@ static void snd_pcm_oss_release_substream(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime; runtime = substream->runtime; vfree(runtime->oss.buffer); +#ifdef CONFIG_SND_PCM_OSS_PLUGINS snd_pcm_oss_plugin_clear(substream); +#endif substream->oss.file = NULL; substream->oss.oss = 0; } @@ -2246,8 +2260,10 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) if ((err = snd_pcm_oss_change_params(substream)) < 0) return err; } +#ifdef CONFIG_SND_PCM_OSS_PLUGINS if (runtime->oss.plugin_first != NULL) return -EIO; +#endif if (area->vm_pgoff != 0) return -EINVAL; diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 7e8676880dde16..89bc8f96504397 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -25,6 +25,9 @@ #endif #include + +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include #include #include @@ -916,3 +919,5 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_area, size_t src_of } return 0; } + +#endif diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 29198da615cd1f..a8a4f958043565 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -22,6 +22,8 @@ * */ +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include static inline unsigned long *bitmap_alloc(unsigned int nbits) @@ -191,6 +193,14 @@ void zero_channel(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *dst_channel, size_t samples); +#else + +static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } +static inline snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size) { return clt_size; } +static inline int snd_pcm_plug_slave_format(int format, struct snd_mask *format_mask) { return format; } + +#endif + #ifdef PLUGIN_DEBUG #define pdprintf( fmt, args... ) printk( "plugin: " fmt, ##args) #else diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 4854cef6fb4f69..c4b75bff0ee521 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -20,6 +20,9 @@ */ #include + +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include #include #include @@ -377,3 +380,5 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } + +#endif diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index 726c5caa3fdb29..f99a54e6551a67 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -20,6 +20,9 @@ */ #include + +#ifdef CONFIG_SND_PCM_OSS_PLUGINS + #include #include #include @@ -519,3 +522,5 @@ int snd_pcm_plugin_build_route(struct snd_pcm_substream *plug, *r_plugin = plugin; return 0; } + +#endif diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 28ca61eb0b0d99..d92c3ce4a4c9ca 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -140,6 +140,9 @@ static int snd_pcm_control_ioctl(struct snd_card *card, } return -ENOIOCTLCMD; } + +#if defined(CONFIG_PROC_FS) && defined(CONFIG_SND_VERBOSE_PROCFS) + #define STATE(v) [SNDRV_PCM_STATE_##v] = #v #define STREAM(v) [SNDRV_PCM_STREAM_##v] = #v #define READY(v) [SNDRV_PCM_READY_##v] = #v @@ -197,7 +200,6 @@ const char *snd_pcm_format_name(snd_pcm_format_t format) return snd_pcm_format_names[format]; } -#ifdef CONFIG_PROC_FS static char *snd_pcm_stream_names[] = { STREAM(PLAYBACK), STREAM(CAPTURE), diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f5aadb001986f2..c9476c237c4222 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1384,8 +1384,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, channels = params_channels(hw_params); fmt = find_format(subs, format, rate, channels); if (! fmt) { - snd_printd(KERN_DEBUG "cannot set format: format = %s, rate = %d, channels = %d\n", - snd_pcm_format_name(format), rate, channels); + snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n", + format, rate, channels); return -EINVAL; } @@ -2011,6 +2011,8 @@ static struct usb_driver usb_audio_driver = { }; +#if defined(CONFIG_PROCFS) && defined(CONFIG_SND_VERBOSE_PROCFS) + /* * proc interface for list the supported pcm formats */ @@ -2101,6 +2103,13 @@ static void proc_pcm_format_add(struct snd_usb_stream *stream) snd_info_set_text_ops(entry, stream, 1024, proc_pcm_format_read); } +#else + +static inline void proc_pcm_format_add(struct snd_usb_stream *stream) +{ +} + +#endif /* * initialize the substream instance. -- cgit 1.2.3-korg From 9d83911ac082c6d63c1c08f235349113d8c1d0a3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jan 2006 11:32:52 +0100 Subject: [ALSA] Fix a typo Modules: ALSA<-OSS emulation Fixed a typo. Signed-off-by: Takashi Iwai --- sound/core/oss/copy.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/copy.c b/sound/core/oss/copy.c index 7c008c4c0dd135..6658facc5cdaa2 100644 --- a/sound/core/oss/copy.c +++ b/sound/core/oss/copy.c @@ -21,7 +21,7 @@ #include -#ifdef SND_PCM_OSS_PLUGINS +#ifdef CONFIG_SND_PCM_OSS_PLUGINS #include #include -- cgit 1.2.3-korg From 0534ab4279bd25e2d0a888af07466446dac05d74 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jan 2006 12:09:12 +0100 Subject: [ALSA] Clean up pcm-oss plugins Modules: ALSA<-OSS emulation Clean up pcm-oss plugin codes. Removed dead codes, and simplified route/rate plugins. Signed-off-by: Takashi Iwai --- sound/core/oss/linear.c | 2 +- sound/core/oss/mulaw.c | 19 ++ sound/core/oss/pcm_plugin.c | 264 ++++-------------------- sound/core/oss/pcm_plugin.h | 26 --- sound/core/oss/plugin_ops.h | 166 --------------- sound/core/oss/rate.c | 80 ++------ sound/core/oss/route.c | 484 ++++---------------------------------------- 7 files changed, 117 insertions(+), 924 deletions(-) diff --git a/sound/core/oss/linear.c b/sound/core/oss/linear.c index ef331230b3a6ba..5b1bcdc64779d7 100644 --- a/sound/core/oss/linear.c +++ b/sound/core/oss/linear.c @@ -106,7 +106,7 @@ static snd_pcm_sframes_t linear_transfer(struct snd_pcm_plugin *plugin, return frames; } -int conv_index(int src_format, int dst_format) +static int conv_index(int src_format, int dst_format) { int src_endian, dst_endian, sign, src_width, dst_width; diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index ed12c81fcf1b94..2eb18807e6d046 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -265,6 +265,25 @@ static snd_pcm_sframes_t mulaw_transfer(struct snd_pcm_plugin *plugin, return frames; } +static int getput_index(int format) +{ + int sign, width, endian; + sign = !snd_pcm_format_signed(format); + width = snd_pcm_format_width(format) / 8 - 1; + if (width < 0 || width > 3) { + snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format); + width = 0; + } +#ifdef SNDRV_LITTLE_ENDIAN + endian = snd_pcm_format_big_endian(format); +#else + endian = snd_pcm_format_little_endian(format); +#endif + if (endian < 0) + endian = 0; + return width * 4 + endian * 2 + sign; +} + int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, struct snd_pcm_plugin_format *src_format, struct snd_pcm_plugin_format *dst_format, diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index 89bc8f96504397..cec2774425ec4b 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -39,26 +39,6 @@ #define snd_pcm_plug_first(plug) ((plug)->runtime->oss.plugin_first) #define snd_pcm_plug_last(plug) ((plug)->runtime->oss.plugin_last) -static int snd_pcm_plugin_src_channels_mask(struct snd_pcm_plugin *plugin, - unsigned long *dst_vmask, - unsigned long **src_vmask) -{ - unsigned long *vmask = plugin->src_vmask; - bitmap_copy(vmask, dst_vmask, plugin->src_format.channels); - *src_vmask = vmask; - return 0; -} - -static int snd_pcm_plugin_dst_channels_mask(struct snd_pcm_plugin *plugin, - unsigned long *src_vmask, - unsigned long **dst_vmask) -{ - unsigned long *vmask = plugin->dst_vmask; - bitmap_copy(vmask, src_vmask, plugin->dst_format.channels); - *dst_vmask = vmask; - return 0; -} - /* * because some cards might have rates "very close", we ignore * all "resampling" requests within +-5% @@ -196,19 +176,7 @@ int snd_pcm_plugin_build(struct snd_pcm_substream *plug, snd_pcm_plugin_free(plugin); return -ENOMEM; } - plugin->src_vmask = bitmap_alloc(src_format->channels); - if (plugin->src_vmask == NULL) { - snd_pcm_plugin_free(plugin); - return -ENOMEM; - } - plugin->dst_vmask = bitmap_alloc(dst_format->channels); - if (plugin->dst_vmask == NULL) { - snd_pcm_plugin_free(plugin); - return -ENOMEM; - } plugin->client_channels = snd_pcm_plugin_client_channels; - plugin->src_channels_mask = snd_pcm_plugin_src_channels_mask; - plugin->dst_channels_mask = snd_pcm_plugin_dst_channels_mask; *ret = plugin; return 0; } @@ -221,8 +189,6 @@ int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin) plugin->private_free(plugin); kfree(plugin->buf_channels); vfree(plugin->buf); - kfree(plugin->src_vmask); - kfree(plugin->dst_vmask); kfree(plugin); return 0; } @@ -432,24 +398,14 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, dstformat.channels); /* Format change (linearization) */ - if ((srcformat.format != dstformat.format || - !rate_match(srcformat.rate, dstformat.rate) || - srcformat.channels != dstformat.channels) && - !snd_pcm_format_linear(srcformat.format)) { - if (snd_pcm_format_linear(dstformat.format)) - tmpformat.format = dstformat.format; - else - tmpformat.format = SNDRV_PCM_FORMAT_S16; - switch (srcformat.format) { - case SNDRV_PCM_FORMAT_MU_LAW: - err = snd_pcm_plugin_build_mulaw(plug, - &srcformat, &tmpformat, - &plugin); - break; - default: + if (! rate_match(srcformat.rate, dstformat.rate) && + ! snd_pcm_format_linear(srcformat.format)) { + if (srcformat.format != SNDRV_PCM_FORMAT_MU_LAW) return -EINVAL; - } - pdprintf("format change: src=%i, dst=%i returns %i\n", srcformat.format, tmpformat.format, err); + tmpformat.format = SNDRV_PCM_FORMAT_S16; + err = snd_pcm_plugin_build_mulaw(plug, + &srcformat, &tmpformat, + &plugin); if (err < 0) return err; err = snd_pcm_plugin_append(plugin); @@ -463,35 +419,11 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, /* channels reduction */ if (srcformat.channels > dstformat.channels) { - int sv = srcformat.channels; - int dv = dstformat.channels; - int *ttable = kcalloc(dv * sv, sizeof(*ttable), GFP_KERNEL); - if (ttable == NULL) - return -ENOMEM; -#if 1 - if (sv == 2 && dv == 1) { - ttable[0] = HALF; - ttable[1] = HALF; - } else -#endif - { - int v; - for (v = 0; v < dv; ++v) - ttable[v * sv + v] = FULL; - } tmpformat.channels = dstformat.channels; - if (rate_match(srcformat.rate, dstformat.rate) && - snd_pcm_format_linear(dstformat.format)) - tmpformat.format = dstformat.format; - err = snd_pcm_plugin_build_route(plug, - &srcformat, &tmpformat, - ttable, &plugin); - kfree(ttable); + err = snd_pcm_plugin_build_route(plug, &srcformat, &tmpformat, &plugin); pdprintf("channels reduction: src=%i, dst=%i returns %i\n", srcformat.channels, tmpformat.channels, err); - if (err < 0) { - snd_pcm_plugin_free(plugin); + if (err < 0) return err; - } err = snd_pcm_plugin_append(plugin); if (err < 0) { snd_pcm_plugin_free(plugin); @@ -503,18 +435,29 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, /* rate resampling */ if (!rate_match(srcformat.rate, dstformat.rate)) { + if (srcformat.format != SNDRV_PCM_FORMAT_S16) { + /* convert to S16 for resampling */ + tmpformat.format = SNDRV_PCM_FORMAT_S16; + err = snd_pcm_plugin_build_linear(plug, + &srcformat, &tmpformat, + &plugin); + if (err < 0) + return err; + err = snd_pcm_plugin_append(plugin); + if (err < 0) { + snd_pcm_plugin_free(plugin); + return err; + } + srcformat = tmpformat; + src_access = dst_access; + } tmpformat.rate = dstformat.rate; - if (srcformat.channels == dstformat.channels && - snd_pcm_format_linear(dstformat.format)) - tmpformat.format = dstformat.format; err = snd_pcm_plugin_build_rate(plug, &srcformat, &tmpformat, &plugin); pdprintf("rate down resampling: src=%i, dst=%i returns %i\n", srcformat.rate, tmpformat.rate, err); - if (err < 0) { - snd_pcm_plugin_free(plugin); + if (err < 0) return err; - } err = snd_pcm_plugin_append(plugin); if (err < 0) { snd_pcm_plugin_free(plugin); @@ -524,52 +467,6 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, src_access = dst_access; } - /* channels extension */ - if (srcformat.channels < dstformat.channels) { - int sv = srcformat.channels; - int dv = dstformat.channels; - int *ttable = kcalloc(dv * sv, sizeof(*ttable), GFP_KERNEL); - if (ttable == NULL) - return -ENOMEM; -#if 0 - { - int v; - for (v = 0; v < sv; ++v) - ttable[v * sv + v] = FULL; - } -#else - { - /* Playback is spreaded on all channels */ - int vd, vs; - for (vd = 0, vs = 0; vd < dv; ++vd) { - ttable[vd * sv + vs] = FULL; - vs++; - if (vs == sv) - vs = 0; - } - } -#endif - tmpformat.channels = dstformat.channels; - if (snd_pcm_format_linear(dstformat.format)) - tmpformat.format = dstformat.format; - err = snd_pcm_plugin_build_route(plug, - &srcformat, &tmpformat, - ttable, &plugin); - kfree(ttable); - pdprintf("channels extension: src=%i, dst=%i returns %i\n", srcformat.channels, tmpformat.channels, err); - if (err < 0) { - snd_pcm_plugin_free(plugin); - return err; - } - err = snd_pcm_plugin_append(plugin); - if (err < 0) { - snd_pcm_plugin_free(plugin); - return err; - } - srcformat = tmpformat; - src_access = dst_access; - } - /* format change */ if (srcformat.format != dstformat.format) { tmpformat.format = dstformat.format; @@ -598,6 +495,22 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, src_access = dst_access; } + /* channels extension */ + if (srcformat.channels < dstformat.channels) { + tmpformat.channels = dstformat.channels; + err = snd_pcm_plugin_build_route(plug, &srcformat, &tmpformat, &plugin); + pdprintf("channels extension: src=%i, dst=%i returns %i\n", srcformat.channels, tmpformat.channels, err); + if (err < 0) + return err; + err = snd_pcm_plugin_append(plugin); + if (err < 0) { + snd_pcm_plugin_free(plugin); + return err; + } + srcformat = tmpformat; + src_access = dst_access; + } + /* de-interleave */ if (src_access != dst_access) { err = snd_pcm_plugin_build_copy(plug, @@ -653,92 +566,6 @@ snd_pcm_sframes_t snd_pcm_plug_client_channels_buf(struct snd_pcm_substream *plu return count; } -static int snd_pcm_plug_playback_channels_mask(struct snd_pcm_substream *plug, - unsigned long *client_vmask) -{ - struct snd_pcm_plugin *plugin = snd_pcm_plug_last(plug); - if (plugin == NULL) { - return 0; - } else { - int schannels = plugin->dst_format.channels; - DECLARE_BITMAP(bs, schannels); - unsigned long *srcmask; - unsigned long *dstmask = bs; - int err; - bitmap_fill(dstmask, schannels); - - while (1) { - err = plugin->src_channels_mask(plugin, dstmask, &srcmask); - if (err < 0) - return err; - dstmask = srcmask; - if (plugin->prev == NULL) - break; - plugin = plugin->prev; - } - bitmap_and(client_vmask, client_vmask, dstmask, plugin->src_format.channels); - return 0; - } -} - -static int snd_pcm_plug_playback_disable_useless_channels(struct snd_pcm_substream *plug, - struct snd_pcm_plugin_channel *src_channels) -{ - struct snd_pcm_plugin *plugin = snd_pcm_plug_first(plug); - unsigned int nchannels = plugin->src_format.channels; - DECLARE_BITMAP(bs, nchannels); - unsigned long *srcmask = bs; - int err; - unsigned int channel; - for (channel = 0; channel < nchannels; channel++) { - if (src_channels[channel].enabled) - set_bit(channel, srcmask); - else - clear_bit(channel, srcmask); - } - err = snd_pcm_plug_playback_channels_mask(plug, srcmask); - if (err < 0) - return err; - for (channel = 0; channel < nchannels; channel++) { - if (!test_bit(channel, srcmask)) - src_channels[channel].enabled = 0; - } - return 0; -} - -static int snd_pcm_plug_capture_disable_useless_channels(struct snd_pcm_substream *plug, - struct snd_pcm_plugin_channel *src_channels, - struct snd_pcm_plugin_channel *client_channels) -{ - struct snd_pcm_plugin *plugin = snd_pcm_plug_last(plug); - unsigned int nchannels = plugin->dst_format.channels; - DECLARE_BITMAP(bs, nchannels); - unsigned long *dstmask = bs; - unsigned long *srcmask; - int err; - unsigned int channel; - for (channel = 0; channel < nchannels; channel++) { - if (client_channels[channel].enabled) - set_bit(channel, dstmask); - else - clear_bit(channel, dstmask); - } - while (plugin) { - err = plugin->src_channels_mask(plugin, dstmask, &srcmask); - if (err < 0) - return err; - dstmask = srcmask; - plugin = plugin->prev; - } - plugin = snd_pcm_plug_first(plug); - nchannels = plugin->src_format.channels; - for (channel = 0; channel < nchannels; channel++) { - if (!test_bit(channel, dstmask)) - src_channels[channel].enabled = 0; - } - return 0; -} - snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, struct snd_pcm_plugin_channel *src_channels, snd_pcm_uframes_t size) { struct snd_pcm_plugin *plugin, *next; @@ -746,9 +573,6 @@ snd_pcm_sframes_t snd_pcm_plug_write_transfer(struct snd_pcm_substream *plug, st int err; snd_pcm_sframes_t frames = size; - if ((err = snd_pcm_plug_playback_disable_useless_channels(plug, src_channels)) < 0) - return err; - plugin = snd_pcm_plug_first(plug); while (plugin && frames > 0) { if ((next = plugin->next) != NULL) { @@ -793,10 +617,6 @@ snd_pcm_sframes_t snd_pcm_plug_read_transfer(struct snd_pcm_substream *plug, str return err; } frames = err; - if (!plugin->prev) { - if ((err = snd_pcm_plug_capture_disable_useless_channels(plug, dst_channels, dst_channels_final)) < 0) - return err; - } } else { dst_channels = dst_channels_final; } diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index a8a4f958043565..3be91b3d53777c 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -24,13 +24,6 @@ #ifdef CONFIG_SND_PCM_OSS_PLUGINS -#include - -static inline unsigned long *bitmap_alloc(unsigned int nbits) -{ - return kmalloc(BITS_TO_LONGS(nbits), GFP_KERNEL); -} - #define snd_pcm_plug_stream(plug) ((plug)->stream) enum snd_pcm_plugin_action { @@ -71,12 +64,6 @@ struct snd_pcm_plugin { snd_pcm_sframes_t (*client_channels)(struct snd_pcm_plugin *plugin, snd_pcm_uframes_t frames, struct snd_pcm_plugin_channel **channels); - int (*src_channels_mask)(struct snd_pcm_plugin *plugin, - unsigned long *dst_vmask, - unsigned long **src_vmask); - int (*dst_channels_mask)(struct snd_pcm_plugin *plugin, - unsigned long *src_vmask, - unsigned long **dst_vmask); snd_pcm_sframes_t (*transfer)(struct snd_pcm_plugin *plugin, const struct snd_pcm_plugin_channel *src_channels, struct snd_pcm_plugin_channel *dst_channels, @@ -92,8 +79,6 @@ struct snd_pcm_plugin { char *buf; snd_pcm_uframes_t buf_frames; struct snd_pcm_plugin_channel *buf_channels; - unsigned long *src_vmask; - unsigned long *dst_vmask; char extra_data[0]; }; @@ -130,7 +115,6 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *handle, int snd_pcm_plugin_build_route(struct snd_pcm_substream *handle, struct snd_pcm_plugin_format *src_format, struct snd_pcm_plugin_format *dst_format, - int *ttable, struct snd_pcm_plugin **r_plugin); int snd_pcm_plugin_build_copy(struct snd_pcm_substream *handle, struct snd_pcm_plugin_format *src_format, @@ -183,16 +167,6 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames, int in_kernel); -#define ROUTE_PLUGIN_RESOLUTION 16 - -int getput_index(int format); -int copy_index(int format); -int conv_index(int src_format, int dst_format); - -void zero_channel(struct snd_pcm_plugin *plugin, - const struct snd_pcm_plugin_channel *dst_channel, - size_t samples); - #else static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } diff --git a/sound/core/oss/plugin_ops.h b/sound/core/oss/plugin_ops.h index 0607e956608471..1f5bde4631f10a 100644 --- a/sound/core/oss/plugin_ops.h +++ b/sound/core/oss/plugin_ops.h @@ -362,172 +362,6 @@ put_s16_xx12_0029: as_u32(dst) = (u_int32_t)swab16(sample) ^ 0x80; goto PUT_S16_ } #endif -#if 0 -#ifdef GET32_LABELS -/* src_wid src_endswap unsigned */ -static void *get32_labels[4 * 2 * 2] = { - &&get32_xxx1_1000, /* 8h -> 32h */ - &&get32_xxx1_9000, /* 8h ^> 32h */ - &&get32_xxx1_1000, /* 8s -> 32h */ - &&get32_xxx1_9000, /* 8s ^> 32h */ - &&get32_xx12_1200, /* 16h -> 32h */ - &&get32_xx12_9200, /* 16h ^> 32h */ - &&get32_xx12_2100, /* 16s -> 32h */ - &&get32_xx12_A100, /* 16s ^> 32h */ - &&get32_x123_1230, /* 24h -> 32h */ - &&get32_x123_9230, /* 24h ^> 32h */ - &&get32_123x_3210, /* 24s -> 32h */ - &&get32_123x_B210, /* 24s ^> 32h */ - &&get32_1234_1234, /* 32h -> 32h */ - &&get32_1234_9234, /* 32h ^> 32h */ - &&get32_1234_4321, /* 32s -> 32h */ - &&get32_1234_C321, /* 32s ^> 32h */ -}; -#endif - -#ifdef GET32_END -while (0) { -get32_xxx1_1000: sample = (u_int32_t)as_u8(src) << 24; goto GET32_END; -get32_xxx1_9000: sample = (u_int32_t)(as_u8(src) ^ 0x80) << 24; goto GET32_END; -get32_xx12_1200: sample = (u_int32_t)as_u16(src) << 16; goto GET32_END; -get32_xx12_9200: sample = (u_int32_t)(as_u16(src) ^ 0x8000) << 16; goto GET32_END; -get32_xx12_2100: sample = (u_int32_t)swab16(as_u16(src)) << 16; goto GET32_END; -get32_xx12_A100: sample = (u_int32_t)swab16(as_u16(src) ^ 0x80) << 16; goto GET32_END; -get32_x123_1230: sample = as_u32(src) << 8; goto GET32_END; -get32_x123_9230: sample = (as_u32(src) << 8) ^ 0x80000000; goto GET32_END; -get32_123x_3210: sample = swab32(as_u32(src) >> 8); goto GET32_END; -get32_123x_B210: sample = swab32((as_u32(src) >> 8) ^ 0x80); goto GET32_END; -get32_1234_1234: sample = as_u32(src); goto GET32_END; -get32_1234_9234: sample = as_u32(src) ^ 0x80000000; goto GET32_END; -get32_1234_4321: sample = swab32(as_u32(src)); goto GET32_END; -get32_1234_C321: sample = swab32(as_u32(src) ^ 0x80); goto GET32_END; -} -#endif -#endif - -#ifdef PUT_U32_LABELS -/* dst_wid dst_endswap unsigned */ -static void *put_u32_labels[4 * 2 * 2] = { - &&put_u32_1234_xxx9, /* u32h -> s8h */ - &&put_u32_1234_xxx1, /* u32h -> u8h */ - &&put_u32_1234_xxx9, /* u32h -> s8s */ - &&put_u32_1234_xxx1, /* u32h -> u8s */ - &&put_u32_1234_xx92, /* u32h -> s16h */ - &&put_u32_1234_xx12, /* u32h -> u16h */ - &&put_u32_1234_xx29, /* u32h -> s16s */ - &&put_u32_1234_xx21, /* u32h -> u16s */ - &&put_u32_1234_x923, /* u32h -> s24h */ - &&put_u32_1234_x123, /* u32h -> u24h */ - &&put_u32_1234_329x, /* u32h -> s24s */ - &&put_u32_1234_321x, /* u32h -> u24s */ - &&put_u32_1234_9234, /* u32h -> s32h */ - &&put_u32_1234_1234, /* u32h -> u32h */ - &&put_u32_1234_4329, /* u32h -> s32s */ - &&put_u32_1234_4321, /* u32h -> u32s */ -}; -#endif - -#ifdef PUT_U32_END -while (0) { -put_u32_1234_xxx1: as_u8(dst) = sample >> 24; goto PUT_U32_END; -put_u32_1234_xxx9: as_u8(dst) = (sample >> 24) ^ 0x80; goto PUT_U32_END; -put_u32_1234_xx12: as_u16(dst) = sample >> 16; goto PUT_U32_END; -put_u32_1234_xx92: as_u16(dst) = (sample >> 16) ^ 0x8000; goto PUT_U32_END; -put_u32_1234_xx21: as_u16(dst) = swab16(sample >> 16); goto PUT_U32_END; -put_u32_1234_xx29: as_u16(dst) = swab16(sample >> 16) ^ 0x80; goto PUT_U32_END; -put_u32_1234_x123: as_u32(dst) = sample >> 8; goto PUT_U32_END; -put_u32_1234_x923: as_u32(dst) = (sample >> 8) ^ 0x800000; goto PUT_U32_END; -put_u32_1234_321x: as_u32(dst) = swab32(sample) << 8; goto PUT_U32_END; -put_u32_1234_329x: as_u32(dst) = (swab32(sample) ^ 0x80) << 8; goto PUT_U32_END; -put_u32_1234_1234: as_u32(dst) = sample; goto PUT_U32_END; -put_u32_1234_9234: as_u32(dst) = sample ^ 0x80000000; goto PUT_U32_END; -put_u32_1234_4321: as_u32(dst) = swab32(sample); goto PUT_U32_END; -put_u32_1234_4329: as_u32(dst) = swab32(sample) ^ 0x80; goto PUT_U32_END; -} -#endif - -#ifdef GET_U_LABELS -/* width endswap unsigned*/ -static void *get_u_labels[4 * 2 * 2] = { - &&get_u_s8, /* s8 -> u8 */ - &&get_u_u8, /* u8 -> u8 */ - &&get_u_s8, /* s8 -> u8 */ - &&get_u_u8, /* u8 -> u8 */ - &&get_u_s16h, /* s16h -> u16h */ - &&get_u_u16h, /* u16h -> u16h */ - &&get_u_s16s, /* s16s -> u16h */ - &&get_u_u16s, /* u16s -> u16h */ - &&get_u_s24h, /* s24h -> u32h */ - &&get_u_u24h, /* u24h -> u32h */ - &&get_u_s24s, /* s24s -> u32h */ - &&get_u_u24s, /* u24s -> u32h */ - &&get_u_s32h, /* s32h -> u32h */ - &&get_u_u32h, /* u32h -> u32h */ - &&get_u_s32s, /* s32s -> u32h */ - &&get_u_u32s, /* u32s -> u32h */ -}; -#endif - -#ifdef GET_U_END -while (0) { -get_u_s8: sample = as_u8(src) ^ 0x80; goto GET_U_END; -get_u_u8: sample = as_u8(src); goto GET_U_END; -get_u_s16h: sample = as_u16(src) ^ 0x8000; goto GET_U_END; -get_u_u16h: sample = as_u16(src); goto GET_U_END; -get_u_s16s: sample = swab16(as_u16(src) ^ 0x80); goto GET_U_END; -get_u_u16s: sample = swab16(as_u16(src)); goto GET_U_END; -get_u_s24h: sample = (as_u32(src) ^ 0x800000); goto GET_U_END; -get_u_u24h: sample = as_u32(src); goto GET_U_END; -get_u_s24s: sample = swab32(as_u32(src) ^ 0x800000); goto GET_U_END; -get_u_u24s: sample = swab32(as_u32(src)); goto GET_U_END; -get_u_s32h: sample = as_u32(src) ^ 0x80000000; goto GET_U_END; -get_u_u32h: sample = as_u32(src); goto GET_U_END; -get_u_s32s: sample = swab32(as_u32(src) ^ 0x80); goto GET_U_END; -get_u_u32s: sample = swab32(as_u32(src)); goto GET_U_END; -} -#endif - -#if 0 -#ifdef PUT_LABELS -/* width endswap unsigned */ -static void *put_labels[4 * 2 * 2] = { - &&put_s8, /* s8 -> s8 */ - &&put_u8, /* u8 -> s8 */ - &&put_s8, /* s8 -> s8 */ - &&put_u8, /* u8 -> s8 */ - &&put_s16h, /* s16h -> s16h */ - &&put_u16h, /* u16h -> s16h */ - &&put_s16s, /* s16s -> s16h */ - &&put_u16s, /* u16s -> s16h */ - &&put_s24h, /* s24h -> s32h */ - &&put_u24h, /* u24h -> s32h */ - &&put_s24s, /* s24s -> s32h */ - &&put_u24s, /* u24s -> s32h */ - &&put_s32h, /* s32h -> s32h */ - &&put_u32h, /* u32h -> s32h */ - &&put_s32s, /* s32s -> s32h */ - &&put_u32s, /* u32s -> s32h */ -}; -#endif - -#ifdef PUT_END -put_s8: as_s8(dst) = sample; goto PUT_END; -put_u8: as_u8(dst) = sample ^ 0x80; goto PUT_END; -put_s16h: as_s16(dst) = sample; goto PUT_END; -put_u16h: as_u16(dst) = sample ^ 0x8000; goto PUT_END; -put_s16s: as_s16(dst) = swab16(sample); goto PUT_END; -put_u16s: as_u16(dst) = swab16(sample ^ 0x80); goto PUT_END; -put_s24h: as_s24(dst) = sample & 0xffffff; goto PUT_END; -put_u24h: as_u24(dst) = sample ^ 0x80000000; goto PUT_END; -put_s24s: as_s24(dst) = swab32(sample & 0xffffff); goto PUT_END; -put_u24s: as_u24(dst) = swab32(sample ^ 0x80); goto PUT_END; -put_s32h: as_s32(dst) = sample; goto PUT_END; -put_u32h: as_u32(dst) = sample ^ 0x80000000; goto PUT_END; -put_s32s: as_s32(dst) = swab32(sample); goto PUT_END; -put_u32s: as_u32(dst) = swab32(sample ^ 0x80); goto PUT_END; -#endif -#endif - #undef as_u8 #undef as_u16 #undef as_u32 diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index c4b75bff0ee521..18d8a0f4e8168a 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -50,7 +50,6 @@ struct rate_priv { unsigned int pitch; unsigned int pos; rate_f func; - int get, put; snd_pcm_sframes_t old_src_frames, old_dst_frames; struct rate_channel channels[0]; }; @@ -74,21 +73,12 @@ static void resample_expand(struct snd_pcm_plugin *plugin, unsigned int pos = 0; signed int val; signed short S1, S2; - char *src, *dst; + signed short *src, *dst; unsigned int channel; int src_step, dst_step; int src_frames1, dst_frames1; struct rate_priv *data = (struct rate_priv *)plugin->extra_data; struct rate_channel *rchannels = data->channels; - -#define GET_S16_LABELS -#define PUT_S16_LABELS -#include "plugin_ops.h" -#undef GET_S16_LABELS -#undef PUT_S16_LABELS - void *get = get_s16_labels[data->get]; - void *put = put_s16_labels[data->put]; - signed short sample = 0; for (channel = 0; channel < plugin->src_format.channels; channel++) { pos = data->pos; @@ -101,10 +91,12 @@ static void resample_expand(struct snd_pcm_plugin *plugin, continue; } dst_channels[channel].enabled = 1; - src = (char *)src_channels[channel].area.addr + src_channels[channel].area.first / 8; - dst = (char *)dst_channels[channel].area.addr + dst_channels[channel].area.first / 8; - src_step = src_channels[channel].area.step / 8; - dst_step = dst_channels[channel].area.step / 8; + src = (signed short *)src_channels[channel].area.addr + + src_channels[channel].area.first / 8 / 2; + dst = (signed short *)dst_channels[channel].area.addr + + dst_channels[channel].area.first / 8 / 2; + src_step = src_channels[channel].area.step / 8 / 2; + dst_step = dst_channels[channel].area.step / 8 / 2; src_frames1 = src_frames; dst_frames1 = dst_frames; while (dst_frames1-- > 0) { @@ -112,12 +104,7 @@ static void resample_expand(struct snd_pcm_plugin *plugin, pos &= R_MASK; S1 = S2; if (src_frames1-- > 0) { - goto *get; -#define GET_S16_END after_get -#include "plugin_ops.h" -#undef GET_S16_END - after_get: - S2 = sample; + S2 = *src; src += src_step; } } @@ -126,12 +113,7 @@ static void resample_expand(struct snd_pcm_plugin *plugin, val = -32768; else if (val > 32767) val = 32767; - sample = val; - goto *put; -#define PUT_S16_END after_put -#include "plugin_ops.h" -#undef PUT_S16_END - after_put: + *dst = val; dst += dst_step; pos += data->pitch; } @@ -150,21 +132,12 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, unsigned int pos = 0; signed int val; signed short S1, S2; - char *src, *dst; + signed short *src, *dst; unsigned int channel; int src_step, dst_step; int src_frames1, dst_frames1; struct rate_priv *data = (struct rate_priv *)plugin->extra_data; struct rate_channel *rchannels = data->channels; - -#define GET_S16_LABELS -#define PUT_S16_LABELS -#include "plugin_ops.h" -#undef GET_S16_LABELS -#undef PUT_S16_LABELS - void *get = get_s16_labels[data->get]; - void *put = put_s16_labels[data->put]; - signed short sample = 0; for (channel = 0; channel < plugin->src_format.channels; ++channel) { pos = data->pos; @@ -177,21 +150,18 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, continue; } dst_channels[channel].enabled = 1; - src = (char *)src_channels[channel].area.addr + src_channels[channel].area.first / 8; - dst = (char *)dst_channels[channel].area.addr + dst_channels[channel].area.first / 8; - src_step = src_channels[channel].area.step / 8; - dst_step = dst_channels[channel].area.step / 8; + src = (signed short *)src_channels[channel].area.addr + + src_channels[channel].area.first / 8 / 2; + dst = (signed short *)dst_channels[channel].area.addr + + dst_channels[channel].area.first / 8 / 2; + src_step = src_channels[channel].area.step / 8 / 2; + dst_step = dst_channels[channel].area.step / 8 / 2; src_frames1 = src_frames; dst_frames1 = dst_frames; while (dst_frames1 > 0) { S1 = S2; if (src_frames1-- > 0) { - goto *get; -#define GET_S16_END after_get -#include "plugin_ops.h" -#undef GET_S16_END - after_get: - S2 = sample; + S1 = *src; src += src_step; } if (pos & ~R_MASK) { @@ -201,12 +171,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, val = -32768; else if (val > 32767) val = 32767; - sample = val; - goto *put; -#define PUT_S16_END after_put -#include "plugin_ops.h" -#undef PUT_S16_END - after_put: + *dst = val; dst += dst_step; dst_frames1--; } @@ -346,8 +311,8 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, snd_assert(src_format->channels == dst_format->channels, return -ENXIO); snd_assert(src_format->channels > 0, return -ENXIO); - snd_assert(snd_pcm_format_linear(src_format->format) != 0, return -ENXIO); - snd_assert(snd_pcm_format_linear(dst_format->format) != 0, return -ENXIO); + snd_assert(src_format->format == SNDRV_PCM_FORMAT_S16, return -ENXIO); + snd_assert(dst_format->format == SNDRV_PCM_FORMAT_S16, return -ENXIO); snd_assert(src_format->rate != dst_format->rate, return -ENXIO); err = snd_pcm_plugin_build(plug, "rate conversion", @@ -358,11 +323,6 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, if (err < 0) return err; data = (struct rate_priv *)plugin->extra_data; - data->get = getput_index(src_format->format); - snd_assert(data->get >= 0 && data->get < 4*2*2, return -EINVAL); - data->put = getput_index(dst_format->format); - snd_assert(data->put >= 0 && data->put < 4*2*2, return -EINVAL); - if (src_format->rate < dst_format->rate) { data->pitch = ((src_format->rate << SHIFT) + (dst_format->rate >> 1)) / dst_format->rate; data->func = resample_expand; diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index f99a54e6551a67..46917dc0196b3b 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -1,5 +1,5 @@ /* - * Attenuated route Plug-In + * Route Plug-In * Copyright (c) 2000 by Abramo Bagnara * * @@ -29,496 +29,82 @@ #include #include "pcm_plugin.h" -/* The best possible hack to support missing optimization in gcc 2.7.2.3 */ -#if ROUTE_PLUGIN_RESOLUTION & (ROUTE_PLUGIN_RESOLUTION - 1) != 0 -#define div(a) a /= ROUTE_PLUGIN_RESOLUTION -#elif ROUTE_PLUGIN_RESOLUTION == 16 -#define div(a) a >>= 4 -#else -#error "Add some code here" -#endif - -struct ttable_dst; - -typedef void (*route_channel_f)(struct snd_pcm_plugin *plugin, - const struct snd_pcm_plugin_channel *src_channels, - struct snd_pcm_plugin_channel *dst_channel, - struct ttable_dst *ttable, snd_pcm_uframes_t frames); - -struct ttable_src { - int channel; - int as_int; -}; - -struct ttable_dst { - int att; /* Attenuated */ - unsigned int nsrcs; - struct ttable_src *srcs; - route_channel_f func; -}; - -struct route_priv { - enum {R_UINT32=0, R_UINT64=1} sum_type; - int get, put; - int conv; - int src_sample_size; - struct ttable_dst ttable[0]; -}; - -union sum { - u_int32_t as_uint32; - u_int64_t as_uint64; -}; - - -static void route_to_channel_from_zero(struct snd_pcm_plugin *plugin, - const struct snd_pcm_plugin_channel *src_channels, - struct snd_pcm_plugin_channel *dst_channel, - struct ttable_dst *ttable, - snd_pcm_uframes_t frames) +static void zero_areas(struct snd_pcm_plugin_channel *dvp, int ndsts, + snd_pcm_uframes_t frames, int format) { - if (dst_channel->wanted) - snd_pcm_area_silence(&dst_channel->area, 0, frames, plugin->dst_format.format); - dst_channel->enabled = 0; -} - -static void route_to_channel_from_one(struct snd_pcm_plugin *plugin, - const struct snd_pcm_plugin_channel *src_channels, - struct snd_pcm_plugin_channel *dst_channel, - struct ttable_dst *ttable, - snd_pcm_uframes_t frames) -{ -#define CONV_LABELS -#include "plugin_ops.h" -#undef CONV_LABELS - struct route_priv *data = (struct route_priv *)plugin->extra_data; - void *conv; - const struct snd_pcm_plugin_channel *src_channel = NULL; - unsigned int srcidx; - char *src, *dst; - int src_step, dst_step; - for (srcidx = 0; srcidx < ttable->nsrcs; ++srcidx) { - src_channel = &src_channels[ttable->srcs[srcidx].channel]; - if (src_channel->area.addr != NULL) - break; - } - if (srcidx == ttable->nsrcs) { - route_to_channel_from_zero(plugin, src_channels, dst_channel, ttable, frames); - return; - } - - dst_channel->enabled = 1; - conv = conv_labels[data->conv]; - src = src_channel->area.addr + src_channel->area.first / 8; - src_step = src_channel->area.step / 8; - dst = dst_channel->area.addr + dst_channel->area.first / 8; - dst_step = dst_channel->area.step / 8; - while (frames-- > 0) { - goto *conv; -#define CONV_END after -#include "plugin_ops.h" -#undef CONV_END - after: - src += src_step; - dst += dst_step; + int dst = 0; + for (; dst < ndsts; ++dst) { + if (dvp->wanted) + snd_pcm_area_silence(&dvp->area, 0, frames, format); + dvp->enabled = 0; + dvp++; } } -static void route_to_channel(struct snd_pcm_plugin *plugin, - const struct snd_pcm_plugin_channel *src_channels, +static inline void copy_area(const struct snd_pcm_plugin_channel *src_channel, struct snd_pcm_plugin_channel *dst_channel, - struct ttable_dst *ttable, snd_pcm_uframes_t frames) + snd_pcm_uframes_t frames, int format) { -#define GET_U_LABELS -#define PUT_U32_LABELS -#include "plugin_ops.h" -#undef GET_U_LABELS -#undef PUT_U32_LABELS - static void *zero_labels[2] = { &&zero_int32, &&zero_int64 }; - /* sum_type att */ - static void *add_labels[2 * 2] = { &&add_int32_noatt, &&add_int32_att, - &&add_int64_noatt, &&add_int64_att, - }; - /* sum_type att shift */ - static void *norm_labels[2 * 2 * 4] = { NULL, - &&norm_int32_8_noatt, - &&norm_int32_16_noatt, - &&norm_int32_24_noatt, - NULL, - &&norm_int32_8_att, - &&norm_int32_16_att, - &&norm_int32_24_att, - &&norm_int64_0_noatt, - &&norm_int64_8_noatt, - &&norm_int64_16_noatt, - &&norm_int64_24_noatt, - &&norm_int64_0_att, - &&norm_int64_8_att, - &&norm_int64_16_att, - &&norm_int64_24_att, - }; - struct route_priv *data = (struct route_priv *)plugin->extra_data; - void *zero, *get, *add, *norm, *put_u32; - int nsrcs = ttable->nsrcs; - char *dst; - int dst_step; - char *srcs[nsrcs]; - int src_steps[nsrcs]; - struct ttable_src src_tt[nsrcs]; - u_int32_t sample = 0; - int srcidx, srcidx1 = 0; - for (srcidx = 0; srcidx < nsrcs; ++srcidx) { - const struct snd_pcm_plugin_channel *src_channel = &src_channels[ttable->srcs[srcidx].channel]; - if (!src_channel->enabled) - continue; - srcs[srcidx1] = src_channel->area.addr + src_channel->area.first / 8; - src_steps[srcidx1] = src_channel->area.step / 8; - src_tt[srcidx1] = ttable->srcs[srcidx]; - srcidx1++; - } - nsrcs = srcidx1; - if (nsrcs == 0) { - route_to_channel_from_zero(plugin, src_channels, dst_channel, ttable, frames); - return; - } else if (nsrcs == 1 && src_tt[0].as_int == ROUTE_PLUGIN_RESOLUTION) { - route_to_channel_from_one(plugin, src_channels, dst_channel, ttable, frames); - return; - } - dst_channel->enabled = 1; - zero = zero_labels[data->sum_type]; - get = get_u_labels[data->get]; - add = add_labels[data->sum_type * 2 + ttable->att]; - norm = norm_labels[data->sum_type * 8 + ttable->att * 4 + 4 - data->src_sample_size]; - put_u32 = put_u32_labels[data->put]; - dst = dst_channel->area.addr + dst_channel->area.first / 8; - dst_step = dst_channel->area.step / 8; - - while (frames-- > 0) { - struct ttable_src *ttp = src_tt; - union sum sum; - - /* Zero sum */ - goto *zero; - zero_int32: - sum.as_uint32 = 0; - goto zero_end; - zero_int64: - sum.as_uint64 = 0; - goto zero_end; - zero_end: - for (srcidx = 0; srcidx < nsrcs; ++srcidx) { - char *src = srcs[srcidx]; - - /* Get sample */ - goto *get; -#define GET_U_END after_get -#include "plugin_ops.h" -#undef GET_U_END - after_get: - - /* Sum */ - goto *add; - add_int32_att: - sum.as_uint32 += sample * ttp->as_int; - goto after_sum; - add_int32_noatt: - if (ttp->as_int) - sum.as_uint32 += sample; - goto after_sum; - add_int64_att: - sum.as_uint64 += (u_int64_t) sample * ttp->as_int; - goto after_sum; - add_int64_noatt: - if (ttp->as_int) - sum.as_uint64 += sample; - goto after_sum; - after_sum: - srcs[srcidx] += src_steps[srcidx]; - ttp++; - } - - /* Normalization */ - goto *norm; - norm_int32_8_att: - sum.as_uint64 = sum.as_uint32; - norm_int64_8_att: - sum.as_uint64 <<= 8; - norm_int64_0_att: - div(sum.as_uint64); - goto norm_int; - - norm_int32_16_att: - sum.as_uint64 = sum.as_uint32; - norm_int64_16_att: - sum.as_uint64 <<= 16; - div(sum.as_uint64); - goto norm_int; - - norm_int32_24_att: - sum.as_uint64 = sum.as_uint32; - norm_int64_24_att: - sum.as_uint64 <<= 24; - div(sum.as_uint64); - goto norm_int; - - norm_int32_8_noatt: - sum.as_uint64 = sum.as_uint32; - norm_int64_8_noatt: - sum.as_uint64 <<= 8; - goto norm_int; - - norm_int32_16_noatt: - sum.as_uint64 = sum.as_uint32; - norm_int64_16_noatt: - sum.as_uint64 <<= 16; - goto norm_int; - - norm_int32_24_noatt: - sum.as_uint64 = sum.as_uint32; - norm_int64_24_noatt: - sum.as_uint64 <<= 24; - goto norm_int; - - norm_int64_0_noatt: - norm_int: - if (sum.as_uint64 > (u_int32_t)0xffffffff) - sample = (u_int32_t)0xffffffff; - else - sample = sum.as_uint64; - goto after_norm; - - after_norm: - - /* Put sample */ - goto *put_u32; -#define PUT_U32_END after_put_u32 -#include "plugin_ops.h" -#undef PUT_U32_END - after_put_u32: - - dst += dst_step; - } -} - -static int route_src_channels_mask(struct snd_pcm_plugin *plugin, - unsigned long *dst_vmask, - unsigned long **src_vmask) -{ - struct route_priv *data = (struct route_priv *)plugin->extra_data; - int schannels = plugin->src_format.channels; - int dchannels = plugin->dst_format.channels; - unsigned long *vmask = plugin->src_vmask; - int channel; - struct ttable_dst *dp = data->ttable; - bitmap_zero(vmask, schannels); - for (channel = 0; channel < dchannels; channel++, dp++) { - unsigned int src; - struct ttable_src *sp; - if (!test_bit(channel, dst_vmask)) - continue; - sp = dp->srcs; - for (src = 0; src < dp->nsrcs; src++, sp++) - set_bit(sp->channel, vmask); - } - *src_vmask = vmask; - return 0; -} - -static int route_dst_channels_mask(struct snd_pcm_plugin *plugin, - unsigned long *src_vmask, - unsigned long **dst_vmask) -{ - struct route_priv *data = (struct route_priv *)plugin->extra_data; - int dchannels = plugin->dst_format.channels; - unsigned long *vmask = plugin->dst_vmask; - int channel; - struct ttable_dst *dp = data->ttable; - bitmap_zero(vmask, dchannels); - for (channel = 0; channel < dchannels; channel++, dp++) { - unsigned int src; - struct ttable_src *sp; - sp = dp->srcs; - for (src = 0; src < dp->nsrcs; src++, sp++) { - if (test_bit(sp->channel, src_vmask)) { - set_bit(channel, vmask); - break; - } - } - } - *dst_vmask = vmask; - return 0; -} - -static void route_free(struct snd_pcm_plugin *plugin) -{ - struct route_priv *data = (struct route_priv *)plugin->extra_data; - unsigned int dst_channel; - for (dst_channel = 0; dst_channel < plugin->dst_format.channels; ++dst_channel) { - kfree(data->ttable[dst_channel].srcs); - } -} - -static int route_load_ttable(struct snd_pcm_plugin *plugin, - const int *src_ttable) -{ - struct route_priv *data; - unsigned int src_channel, dst_channel; - const int *sptr; - struct ttable_dst *dptr; - if (src_ttable == NULL) - return 0; - data = (struct route_priv *)plugin->extra_data; - dptr = data->ttable; - sptr = src_ttable; - plugin->private_free = route_free; - for (dst_channel = 0; dst_channel < plugin->dst_format.channels; ++dst_channel) { - int t = 0; - int att = 0; - int nsrcs = 0; - struct ttable_src srcs[plugin->src_format.channels]; - for (src_channel = 0; src_channel < plugin->src_format.channels; ++src_channel) { - snd_assert(*sptr >= 0 || *sptr <= FULL, return -ENXIO); - if (*sptr != 0) { - srcs[nsrcs].channel = src_channel; - srcs[nsrcs].as_int = *sptr; - if (*sptr != FULL) - att = 1; - t += *sptr; - nsrcs++; - } - sptr++; - } - dptr->att = att; - dptr->nsrcs = nsrcs; - if (nsrcs == 0) - dptr->func = route_to_channel_from_zero; - else if (nsrcs == 1 && !att) - dptr->func = route_to_channel_from_one; - else - dptr->func = route_to_channel; - if (nsrcs > 0) { - int srcidx; - dptr->srcs = kcalloc(nsrcs, sizeof(*srcs), GFP_KERNEL); - for(srcidx = 0; srcidx < nsrcs; srcidx++) - dptr->srcs[srcidx] = srcs[srcidx]; - } else - dptr->srcs = NULL; - dptr++; - } - return 0; + snd_pcm_area_copy(&src_channel->area, 0, &dst_channel->area, 0, frames, format); } static snd_pcm_sframes_t route_transfer(struct snd_pcm_plugin *plugin, - const struct snd_pcm_plugin_channel *src_channels, - struct snd_pcm_plugin_channel *dst_channels, - snd_pcm_uframes_t frames) + const struct snd_pcm_plugin_channel *src_channels, + struct snd_pcm_plugin_channel *dst_channels, + snd_pcm_uframes_t frames) { - struct route_priv *data; - int src_nchannels, dst_nchannels; - int dst_channel; - struct ttable_dst *ttp; + int nsrcs, ndsts, dst; struct snd_pcm_plugin_channel *dvp; + int format; snd_assert(plugin != NULL && src_channels != NULL && dst_channels != NULL, return -ENXIO); if (frames == 0) return 0; - data = (struct route_priv *)plugin->extra_data; - src_nchannels = plugin->src_format.channels; - dst_nchannels = plugin->dst_format.channels; + nsrcs = plugin->src_format.channels; + ndsts = plugin->dst_format.channels; -#ifdef CONFIG_SND_DEBUG - { - int src_channel; - for (src_channel = 0; src_channel < src_nchannels; ++src_channel) { - snd_assert(src_channels[src_channel].area.first % 8 == 0 || - src_channels[src_channel].area.step % 8 == 0, - return -ENXIO); - } - for (dst_channel = 0; dst_channel < dst_nchannels; ++dst_channel) { - snd_assert(dst_channels[dst_channel].area.first % 8 == 0 || - dst_channels[dst_channel].area.step % 8 == 0, - return -ENXIO); + format = plugin->dst_format.format; + dvp = dst_channels; + if (nsrcs <= 1) { + /* expand to all channels */ + for (dst = 0; dst < ndsts; ++dst) { + copy_area(src_channels, dvp, frames, format); + dvp++; } + return frames; } -#endif - ttp = data->ttable; - dvp = dst_channels; - for (dst_channel = 0; dst_channel < dst_nchannels; ++dst_channel) { - ttp->func(plugin, src_channels, dvp, ttp, frames); + for (dst = 0; dst < ndsts && dst < nsrcs; ++dst) { + copy_area(src_channels, dvp, frames, format); dvp++; - ttp++; + src_channels++; } + if (dst < ndsts) + zero_areas(dvp, ndsts - dst, frames, format); return frames; } -int getput_index(int format) -{ - int sign, width, endian; - sign = !snd_pcm_format_signed(format); - width = snd_pcm_format_width(format) / 8 - 1; - if (width < 0 || width > 3) { - snd_printk(KERN_ERR "snd-pcm-oss: invalid format %d\n", format); - width = 0; - } -#ifdef SNDRV_LITTLE_ENDIAN - endian = snd_pcm_format_big_endian(format); -#else - endian = snd_pcm_format_little_endian(format); -#endif - if (endian < 0) - endian = 0; - return width * 4 + endian * 2 + sign; -} - int snd_pcm_plugin_build_route(struct snd_pcm_substream *plug, struct snd_pcm_plugin_format *src_format, struct snd_pcm_plugin_format *dst_format, - int *ttable, struct snd_pcm_plugin **r_plugin) { - struct route_priv *data; struct snd_pcm_plugin *plugin; int err; snd_assert(r_plugin != NULL, return -ENXIO); *r_plugin = NULL; snd_assert(src_format->rate == dst_format->rate, return -ENXIO); - snd_assert(snd_pcm_format_linear(src_format->format) != 0 && - snd_pcm_format_linear(dst_format->format) != 0, - return -ENXIO); + snd_assert(src_format->format == dst_format->format, return -ENXIO); - err = snd_pcm_plugin_build(plug, "attenuated route conversion", - src_format, dst_format, - sizeof(struct route_priv) + - sizeof(data->ttable[0]) * dst_format->channels, - &plugin); + err = snd_pcm_plugin_build(plug, "route conversion", + src_format, dst_format, 0, &plugin); if (err < 0) return err; - data = (struct route_priv *)plugin->extra_data; - - data->get = getput_index(src_format->format); - snd_assert(data->get >= 0 && data->get < 4*2*2, return -EINVAL); - data->put = getput_index(dst_format->format); - snd_assert(data->get >= 0 && data->get < 4*2*2, return -EINVAL); - data->conv = conv_index(src_format->format, dst_format->format); - - if (snd_pcm_format_width(src_format->format) == 32) - data->sum_type = R_UINT64; - else - data->sum_type = R_UINT32; - data->src_sample_size = snd_pcm_format_width(src_format->format) / 8; - - if ((err = route_load_ttable(plugin, ttable)) < 0) { - snd_pcm_plugin_free(plugin); - return err; - } plugin->transfer = route_transfer; - plugin->src_channels_mask = route_src_channels_mask; - plugin->dst_channels_mask = route_dst_channels_mask; *r_plugin = plugin; return 0; } -- cgit 1.2.3-korg From 45fe722ba814dc50410729a473022c550dd96910 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jan 2006 13:50:16 +0100 Subject: [ALSA] ice1724 - Add support of Prodigy-7.1LT Modules: ICE1712 driver Added the support of Prodigy-7.1LT board. The patch taken from ALSA BTS bug#1090. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/aureon.c | 78 ++++++++++++++++++++++++++++++++++++---------- sound/pci/ice1712/aureon.h | 8 ++++- 2 files changed, 68 insertions(+), 18 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 8809812a1c2241..2175f6721347bb 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -322,36 +322,48 @@ static void aureon_spi_write(struct snd_ice1712 *ice, unsigned int cs, unsigned { unsigned int tmp; int i; + unsigned int mosi, clk; tmp = snd_ice1712_gpio_read(ice); - snd_ice1712_gpio_set_mask(ice, ~(AUREON_WM_RW|AUREON_SPI_MOSI|AUREON_SPI_CLK| - AUREON_WM_CS|AUREON_CS8415_CS)); - tmp |= AUREON_WM_RW; + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT) { + snd_ice1712_gpio_set_mask(ice, ~(PRODIGY_SPI_MOSI|PRODIGY_SPI_CLK|PRODIGY_WM_CS)); + mosi = PRODIGY_SPI_MOSI; + clk = PRODIGY_SPI_CLK; + } + else { + snd_ice1712_gpio_set_mask(ice, ~(AUREON_WM_RW|AUREON_SPI_MOSI|AUREON_SPI_CLK| + AUREON_WM_CS|AUREON_CS8415_CS)); + mosi = AUREON_SPI_MOSI; + clk = AUREON_SPI_CLK; + + tmp |= AUREON_WM_RW; + } + tmp &= ~cs; snd_ice1712_gpio_write(ice, tmp); udelay(1); for (i = bits - 1; i >= 0; i--) { - tmp &= ~AUREON_SPI_CLK; + tmp &= ~clk; snd_ice1712_gpio_write(ice, tmp); udelay(1); if (data & (1 << i)) - tmp |= AUREON_SPI_MOSI; + tmp |= mosi; else - tmp &= ~AUREON_SPI_MOSI; + tmp &= ~mosi; snd_ice1712_gpio_write(ice, tmp); udelay(1); - tmp |= AUREON_SPI_CLK; + tmp |= clk; snd_ice1712_gpio_write(ice, tmp); udelay(1); } - tmp &= ~AUREON_SPI_CLK; + tmp &= ~clk; tmp |= cs; snd_ice1712_gpio_write(ice, tmp); udelay(1); - tmp |= AUREON_SPI_CLK; + tmp |= clk; snd_ice1712_gpio_write(ice, tmp); udelay(1); } @@ -440,7 +452,9 @@ static unsigned short wm_get(struct snd_ice1712 *ice, int reg) */ static void wm_put_nocache(struct snd_ice1712 *ice, int reg, unsigned short val) { - aureon_spi_write(ice, AUREON_WM_CS, (reg << 9) | (val & 0x1ff), 16); + aureon_spi_write(ice, + (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT ? PRODIGY_WM_CS : AUREON_WM_CS), + (reg << 9) | (val & 0x1ff), 16); } /* @@ -1659,7 +1673,7 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) return err; } } - else { + else if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) { for (i = 0; i < ARRAY_SIZE(ac97_controls); i++) { err = snd_ctl_add(ice->card, snd_ctl_new1(&ac97_controls[i], ice)); if (err < 0) @@ -1667,7 +1681,7 @@ static int __devinit aureon_add_controls(struct snd_ice1712 *ice) } } - { + if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) { unsigned char id; snd_ice1712_save_gpio_status(ice); id = aureon_cs8415_get(ice, CS8415_ID); @@ -1822,7 +1836,8 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) udelay(1); /* initialize WM8770 codec */ - if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71) + if (ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71 || + ice->eeprom.subvendor == VT1724_SUBDEVICE_PRODIGY71LT) p = wm_inits_prodigy; else p = wm_inits_aureon; @@ -1830,11 +1845,13 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) wm_put(ice, p[0], p[1]); /* initialize CS8415A codec */ - for (p = cs_inits; *p != (unsigned short)-1; p++) - aureon_spi_write(ice, AUREON_CS8415_CS, *p | 0x200000, 24); - ice->spec.aureon.cs8415_mux = 1; + if (ice->eeprom.subvendor != VT1724_SUBDEVICE_PRODIGY71LT) { + for (p = cs_inits; *p != (unsigned short)-1; p++) + aureon_spi_write(ice, AUREON_CS8415_CS, *p | 0x200000, 24); + ice->spec.aureon.cs8415_mux = 1; - aureon_set_headphone_amp(ice, 1); + aureon_set_headphone_amp(ice, 1); + } snd_ice1712_restore_gpio_status(ice); @@ -1902,6 +1919,23 @@ static unsigned char prodigy71_eeprom[] __devinitdata = { 0x00, /* GPIO_STATE2 */ }; +static unsigned char prodigy71lt_eeprom[] __devinitdata = { + 0x0b, /* SYSCINF: clock 512, spdif-in/ADC, 4DACs */ + 0x80, /* ACLINK: I2S */ + 0xfc, /* I2S: vol, 96k, 24bit, 192k */ + 0xc3, /* SPDUF: out-en, out-int */ + 0x00, /* GPIO_DIR */ + 0x07, /* GPIO_DIR1 */ + 0x00, /* GPIO_DIR2 */ + 0xff, /* GPIO_MASK */ + 0xf8, /* GPIO_MASK1 */ + 0xff, /* GPIO_MASK2 */ + 0x00, /* GPIO_STATE */ + 0x00, /* GPIO_STATE1 */ + 0x00, /* GPIO_STATE2 */ +}; + + /* entry point */ struct snd_ice1712_card_info snd_vt1724_aureon_cards[] __devinitdata = { { @@ -1944,5 +1978,15 @@ struct snd_ice1712_card_info snd_vt1724_aureon_cards[] __devinitdata = { .eeprom_data = prodigy71_eeprom, .driver = "Prodigy71", /* should be identical with Aureon71 */ }, + { + .subvendor = VT1724_SUBDEVICE_PRODIGY71LT, + .name = "Audiotrak Prodigy 7.1 LT", + .model = "prodigy71lt", + .chip_init = aureon_init, + .build_controls = aureon_add_controls, + .eeprom_size = sizeof(prodigy71lt_eeprom), + .eeprom_data = prodigy71lt_eeprom, + .driver = "Prodigy71LT", + }, { } /* terminator */ }; diff --git a/sound/pci/ice1712/aureon.h b/sound/pci/ice1712/aureon.h index 95d515f36f23e3..98a6752280f26c 100644 --- a/sound/pci/ice1712/aureon.h +++ b/sound/pci/ice1712/aureon.h @@ -27,12 +27,14 @@ #define AUREON_DEVICE_DESC "{Terratec,Aureon 5.1 Sky},"\ "{Terratec,Aureon 7.1 Space},"\ "{Terratec,Aureon 7.1 Universe}," \ - "{AudioTrak,Prodigy 7.1}," + "{AudioTrak,Prodigy 7.1}," \ + "{AudioTrak,Prodigy 7.1 LT}," #define VT1724_SUBDEVICE_AUREON51_SKY 0x3b154711 /* Aureon 5.1 Sky */ #define VT1724_SUBDEVICE_AUREON71_SPACE 0x3b154511 /* Aureon 7.1 Space */ #define VT1724_SUBDEVICE_AUREON71_UNIVERSE 0x3b155311 /* Aureon 7.1 Universe */ #define VT1724_SUBDEVICE_PRODIGY71 0x33495345 /* PRODIGY 7.1 */ +#define VT1724_SUBDEVICE_PRODIGY71LT 0x32315441 /* PRODIGY 7.1 LT */ extern struct snd_ice1712_card_info snd_vt1724_aureon_cards[]; @@ -53,4 +55,8 @@ extern struct snd_ice1712_card_info snd_vt1724_aureon_cards[]; #define AUREON_AC97_DATA_HIGH (1 << 8) #define AUREON_AC97_DATA_MASK 0xFF +#define PRODIGY_WM_CS (1 << 8) +#define PRODIGY_SPI_MOSI (1 << 10) +#define PRODIGY_SPI_CLK (1 << 9) + #endif /* __SOUND_AUREON_H */ -- cgit 1.2.3-korg From 7a6c8ff1ef83df4ce44b586999e54966d8e5bda8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jan 2006 13:56:33 +0100 Subject: [ALSA] Update description of ice1724 driver Modules: Documentation Updated the description of ice1724 driver. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index cc8a701871998c..b45f7be47816d5 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -815,6 +815,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------ Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards. + * MidiMan M Audio Revolution 5.1 * MidiMan M Audio Revolution 7.1 * AMP Ltd AUDIO2000 * TerraTec Aureon 5.1 Sky @@ -823,6 +824,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * TerraTec Phase 22 * TerraTec Phase 28 * AudioTrak Prodigy 7.1 + * AudioTrak Prodigy 7.1LT * AudioTrak Prodigy 192 * Pontis MS300 * Albatron K8X800 Pro II @@ -833,9 +835,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. * Shuttle SN25P model - Use the given board model, one of the following: - revo71, amp2000, prodigy71, prodigy192, aureon51, - aureon71, universe, k8x800, phase22, phase28, ms300, - av710 + revo51, revo71, amp2000, prodigy71, prodigy71lt, + prodigy192, aureon51, aureon71, universe, + k8x800, phase22, phase28, ms300, av710 This module supports multiple cards and autoprobe. -- cgit 1.2.3-korg From 97c67b65cbdfd19887450ae1b80ddbb54de9559d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jan 2006 17:16:29 +0100 Subject: [ALSA] au88x0 - 64bit arch fixes Modules: au88x0 driver Fix the driver codes to run on 64bit architectures. The patch taken from ALSA BTS bug#1047. Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0.c | 10 ++++++++-- sound/pci/au88x0/au88x0.h | 12 ++++++------ sound/pci/au88x0/au88x0_core.c | 12 ++++++------ sound/pci/au88x0/au88x0_eq.c | 31 +++++++++++++++---------------- sound/pci/au88x0/au88x0_eq.h | 30 +++++++++++++++--------------- sound/pci/au88x0/au88x0_eqdata.c | 6 +++++- sound/pci/au88x0/au88x0_mpu401.c | 4 ++-- sound/pci/au88x0/au88x0_synth.c | 10 +++++----- sound/pci/au88x0/au88x0_wt.h | 10 +++++----- sound/pci/au88x0/au88x0_xtalk.c | 16 ++++++++-------- sound/pci/au88x0/au88x0_xtalk.h | 12 ++++++------ 11 files changed, 81 insertions(+), 72 deletions(-) diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 7d9184f7367a87..126870ec063a96 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -151,14 +151,18 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) // check PCI availability (DMA). if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK)) { + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { printk(KERN_ERR "error to set DMA mask\n"); + pci_disable_device(pci); return -ENXIO; } chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) + if (chip == NULL) { + pci_disable_device(pci); return -ENOMEM; + } chip->card = card; @@ -208,6 +212,8 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) goto alloc_out; } + snd_card_set_dev(card, &pci->dev); + *rchip = chip; return 0; diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index c2ad2674bea714..d65ccb1866a020 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -39,8 +39,8 @@ #include "au88x0_wt.h" #endif -#define hwread(x,y) readl((x)+((y)>>2)) -#define hwwrite(x,y,z) writel((z),(x)+((y)>>2)) +#define hwread(x,y) readl((x)+(y)) +#define hwwrite(x,y,z) writel((z),(x)+(y)) /* Vortex MPU401 defines. */ #define MIDI_CLOCK_DIV 0x61 @@ -113,7 +113,7 @@ typedef struct { //int this_08; /* Still unknown */ int fifo_enabled; /* this_24 */ int fifo_status; /* this_1c */ - int dma_ctrl; /* this_78 (ADB), this_7c (WT) */ + u32 dma_ctrl; /* this_78 (ADB), this_7c (WT) */ int dma_unknown; /* this_74 (ADB), this_78 (WT). WDM: +8 */ int cfg0; int cfg1; @@ -178,7 +178,7 @@ struct snd_vortex { /* PCI hardware resources */ unsigned long io; - unsigned long __iomem *mmio; + void __iomem *mmio; unsigned int irq; spinlock_t lock; @@ -201,14 +201,14 @@ static void vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, int count); static void vortex_adbdma_setmode(vortex_t * vortex, int adbdma, int ie, int dir, int fmt, int d, - unsigned long offset); + u32 offset); static void vortex_adbdma_setstartbuffer(vortex_t * vortex, int adbdma, int sb); #ifndef CHIP_AU8810 static void vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma, struct snd_sg_buf * sgbuf, int size, int count); static void vortex_wtdma_setmode(vortex_t * vortex, int wtdma, int ie, int fmt, int d, /*int e, */ - unsigned long offset); + u32 offset); static void vortex_wtdma_setstartbuffer(vortex_t * vortex, int wtdma, int sb); #endif diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index e3394fe632539d..9cac02e93b2561 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -376,7 +376,7 @@ vortex_mixer_delWTD(vortex_t * vortex, unsigned char mix, unsigned char ch) static void vortex_mixer_init(vortex_t * vortex) { - unsigned long addr; + u32 addr; int x; // FIXME: get rid of this crap. @@ -639,7 +639,7 @@ static void vortex_src_setupchannel(vortex_t * card, unsigned char src, static void vortex_srcblock_init(vortex_t * vortex) { - unsigned long addr; + u32 addr; int x; hwwrite(vortex->mmio, VORTEX_SRC_SOURCESIZE, 0x1ff); /* @@ -1035,7 +1035,7 @@ vortex_fifo_setwtctrl(vortex_t * vortex, int fifo, int ctrl, int priority, static void vortex_fifo_init(vortex_t * vortex) { int x; - unsigned long addr; + u32 addr; /* ADB DMA channels fifos. */ addr = VORTEX_FIFO_ADBCTRL + ((NR_ADB - 1) * 4); @@ -1054,7 +1054,7 @@ static void vortex_fifo_init(vortex_t * vortex) hwwrite(vortex->mmio, addr, FIFO_U0); if (hwread(vortex->mmio, addr) != FIFO_U0) printk(KERN_ERR - "bad wt fifo reset (0x%08lx, 0x%08x)!\n", + "bad wt fifo reset (0x%08x, 0x%08x)!\n", addr, hwread(vortex->mmio, addr)); vortex_fifo_clearwtdata(vortex, x, FIFO_SIZE); addr -= 4; @@ -1152,7 +1152,7 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, static void vortex_adbdma_setmode(vortex_t * vortex, int adbdma, int ie, int dir, - int fmt, int d, unsigned long offset) + int fmt, int d, u32 offset) { stream_t *dma = &vortex->dma_adb[adbdma]; @@ -1411,7 +1411,7 @@ vortex_wtdma_setbuffers(vortex_t * vortex, int wtdma, static void vortex_wtdma_setmode(vortex_t * vortex, int wtdma, int ie, int fmt, int d, - /*int e, */ unsigned long offset) + /*int e, */ u32 offset) { stream_t *dma = &vortex->dma_wt[wtdma]; diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index c8280f82eb5ad4..64fbfbbaf816f6 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -377,23 +377,23 @@ static void vortex_EqHw_GetLevels(vortex_t * vortex, u16 a[]) #endif /* Global Control */ -static void vortex_EqHw_SetControlReg(vortex_t * vortex, unsigned long reg) +static void vortex_EqHw_SetControlReg(vortex_t * vortex, u32 reg) { hwwrite(vortex->mmio, 0x2b440, reg); } -static void vortex_EqHw_SetSampleRate(vortex_t * vortex, int sr) +static void vortex_EqHw_SetSampleRate(vortex_t * vortex, u32 sr) { hwwrite(vortex->mmio, 0x2b440, ((sr & 0x1f) << 3) | 0xb800); } #if 0 -static void vortex_EqHw_GetControlReg(vortex_t * vortex, unsigned long *reg) +static void vortex_EqHw_GetControlReg(vortex_t * vortex, u32 *reg) { *reg = hwread(vortex->mmio, 0x2b440); } -static void vortex_EqHw_GetSampleRate(vortex_t * vortex, int *sr) +static void vortex_EqHw_GetSampleRate(vortex_t * vortex, u32 *sr) { *sr = (hwread(vortex->mmio, 0x2b440) >> 3) & 0x1f; } @@ -554,7 +554,7 @@ static void vortex_Eqlzr_SetRightGain(vortex_t * vortex, u16 index, u16 gain) #if 0 static int -vortex_Eqlzr_GetAllBands(vortex_t * vortex, u16 * gains, unsigned long *cnt) +vortex_Eqlzr_GetAllBands(vortex_t * vortex, u16 * gains, s32 *cnt) { eqlzr_t *eq = &(vortex->eq); int si = 0; @@ -586,7 +586,7 @@ static int vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex_t * vortex) } static int -vortex_Eqlzr_SetAllBands(vortex_t * vortex, u16 gains[], unsigned long count) +vortex_Eqlzr_SetAllBands(vortex_t * vortex, u16 gains[], s32 count) { eqlzr_t *eq = &(vortex->eq); int i; @@ -604,11 +604,10 @@ vortex_Eqlzr_SetAllBands(vortex_t * vortex, u16 gains[], unsigned long count) } static void -vortex_Eqlzr_SetA3dBypassGain(vortex_t * vortex, unsigned long a, - unsigned long b) +vortex_Eqlzr_SetA3dBypassGain(vortex_t * vortex, u32 a, u32 b) { eqlzr_t *eq = &(vortex->eq); - int eax, ebx; + u32 eax, ebx; eq->this58 = a; eq->this5c = b; @@ -624,7 +623,7 @@ vortex_Eqlzr_SetA3dBypassGain(vortex_t * vortex, unsigned long a, static void vortex_Eqlzr_ProgramA3dBypassGain(vortex_t * vortex) { eqlzr_t *eq = &(vortex->eq); - int eax, ebx; + u32 eax, ebx; if (eq->this54) eax = eq->this0e; @@ -641,7 +640,7 @@ static void vortex_Eqlzr_ShutDownA3d(vortex_t * vortex) vortex_EqHw_ZeroA3DIO(vortex); } -static void vortex_Eqlzr_SetBypass(vortex_t * vortex, long bp) +static void vortex_Eqlzr_SetBypass(vortex_t * vortex, u32 bp) { eqlzr_t *eq = &(vortex->eq); @@ -651,8 +650,8 @@ static void vortex_Eqlzr_SetBypass(vortex_t * vortex, long bp) vortex_EqHw_SetBypassGain(vortex, eq->this08, eq->this08); } else { /* EQ disabled. */ - vortex_EqHw_SetLeftGainsTarget(vortex, (u16 *) (eq->this14)); - vortex_EqHw_SetRightGainsTarget(vortex, (u16 *) (eq->this14)); + vortex_EqHw_SetLeftGainsTarget(vortex, eq->this14_array); + vortex_EqHw_SetRightGainsTarget(vortex, eq->this14_array); vortex_EqHw_SetBypassGain(vortex, eq->this0c, eq->this0c); } vortex_Eqlzr_ProgramA3dBypassGain(vortex); @@ -706,7 +705,7 @@ static void vortex_Eqlzr_init(vortex_t * vortex) eq->this5c = 0xffff; /* Set gains. */ - memset(eq->this14, 0, 2 * 10); + memset(eq->this14_array, 0, sizeof(eq->this14_array)); /* Actual init. */ vortex_EqHw_ZeroState(vortex); @@ -792,7 +791,7 @@ snd_vortex_eq_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucon { vortex_t *vortex = snd_kcontrol_chip(kcontrol); int i = kcontrol->private_value; - u16 gainL, gainR; + u16 gainL = 0, gainR = 0; vortex_Eqlzr_GetLeftGain(vortex, i, &gainL); vortex_Eqlzr_GetRightGain(vortex, i, &gainR); @@ -806,7 +805,7 @@ snd_vortex_eq_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucon { vortex_t *vortex = snd_kcontrol_chip(kcontrol); int changed = 0, i = kcontrol->private_value; - u16 gainL, gainR; + u16 gainL = 0, gainR = 0; vortex_Eqlzr_GetLeftGain(vortex, i, &gainL); vortex_Eqlzr_GetRightGain(vortex, i, &gainR); diff --git a/sound/pci/au88x0/au88x0_eq.h b/sound/pci/au88x0/au88x0_eq.h index e49bc625c873ae..717544af58adae 100644 --- a/sound/pci/au88x0/au88x0_eq.h +++ b/sound/pci/au88x0/au88x0_eq.h @@ -18,26 +18,26 @@ typedef struct { } auxxEqCoeffSet_t; typedef struct { - unsigned int *this00; /*CAsp4HwIO */ - long this04; /* How many filters for each side (default = 10) */ - long this08; /* inited to cero. Stereo flag? */ + u32 ptr_this00; /*CAsp4HwIO */ + s32 this04; /* How many filters for each side (default = 10) */ + s32 this08; /* inited to cero. Stereo flag? */ } eqhw_t; typedef struct { - unsigned int *this00; /*CAsp4Core */ + u32 ptr_this00; /*CAsp4Core */ eqhw_t this04; /* CHwEq */ - short this08; /* Bad codec flag ? SetBypassGain: bypass gain */ - short this0a; - short this0c; /* SetBypassGain: bypass gain when this28 is not set. */ - short this0e; + u16 this08; /* Bad codec flag ? SetBypassGain: bypass gain */ + u16 this0a; + u16 this0c; /* SetBypassGain: bypass gain when this28 is not set. */ + u16 this0e; - long this10; /* How many gains are used for each side (right or left). */ - u16 this14[32]; /* SetLeftGainsTarget: Left (and right?) EQ gains */ - long this24; - long this28; /* flag related to EQ enabled or not. Gang flag ? */ - long this54; /* SetBypass */ - long this58; - long this5c; + s32 this10; /* How many gains are used for each side (right or left). */ + u16 this14_array[32]; /* SetLeftGainsTarget: Left (and right?) EQ gains */ + s32 this24; + s32 this28; /* flag related to EQ enabled or not. Gang flag ? */ + s32 this54; /* SetBypass */ + s32 this58; + s32 this5c; /*0x60 */ auxxEqCoeffSet_t coefset; /* 50 u16 word each channel. */ u16 this130[20]; /* Left and Right gains */ diff --git a/sound/pci/au88x0/au88x0_eqdata.c b/sound/pci/au88x0/au88x0_eqdata.c index abf8d6ac4c157d..ce8dca8ce1e2dd 100644 --- a/sound/pci/au88x0/au88x0_eqdata.c +++ b/sound/pci/au88x0/au88x0_eqdata.c @@ -104,7 +104,11 @@ static u16 asEqOutStateZeros[48] = { }; /*_rodataba0:*/ -static long eq_levels[32] = { +static u16 eq_levels[64] = { + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, 0x0000, diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 8ba6dd36222b46..873f486b07b88e 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -95,7 +95,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) return temp; } #else - port = (unsigned long)(vortex->mmio + (VORTEX_MIDI_DATA >> 2)); + port = (unsigned long)(vortex->mmio + VORTEX_MIDI_DATA); if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port, 1, 0, 0, &rmidi)) != 0) { @@ -105,7 +105,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) return temp; } mpu = rmidi->private_data; - mpu->cport = (unsigned long)(vortex->mmio + (VORTEX_MIDI_CMD >> 2)); + mpu->cport = (unsigned long)(vortex->mmio + VORTEX_MIDI_CMD); #endif vortex->rmidi = rmidi; return 0; diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 65f375bad43aca..d3e662a1285ddd 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -32,7 +32,7 @@ static void vortex_connection_mixin_mix(vortex_t * vortex, int en, unsigned char mix, int a); static void vortex_fifo_wtinitialize(vortex_t * vortex, int fifo, int j); static int vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, - unsigned long val); + u32 val); /* WT */ @@ -166,7 +166,7 @@ static int vortex_wt_GetReg(vortex_t * vortex, char reg, int wt) /* WT hardware abstraction layer generic register interface. */ static int vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt, - unsigned short val) + u16 val) { /* int eax, edx; @@ -190,7 +190,7 @@ vortex_wt_SetReg2(vortex_t * vortex, unsigned char reg, int wt, #endif static int vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, - unsigned long val) + u32 val) { int ecx; @@ -279,7 +279,7 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, static void vortex_wt_init(vortex_t * vortex) { - int var4, var8, varc, var10 = 0, edi; + u32 var4, var8, varc, var10 = 0, edi; var10 &= 0xFFFFFFE3; var10 |= 0x22; @@ -353,7 +353,7 @@ static void vortex_wt_SetVolume(vortex_t * vortex, int wt, int vol[]) static void vortex_wt_SetFrequency(vortex_t * vortex, int wt, unsigned int sr) { wt_voice_t *voice = &(vortex->wt_voice[wt]); - long int eax, edx; + u32 eax, edx; //FIXME: 64 bit operation. eax = ((sr << 0xf) * 0x57619F1) & 0xffffffff; diff --git a/sound/pci/au88x0/au88x0_wt.h b/sound/pci/au88x0/au88x0_wt.h index d536c88b43bfb9..38d98f88a95c06 100644 --- a/sound/pci/au88x0/au88x0_wt.h +++ b/sound/pci/au88x0/au88x0_wt.h @@ -53,11 +53,11 @@ enum { #endif typedef struct { - unsigned int parm0; /* this_1E4 */ - unsigned int parm1; /* this_1E8 */ - unsigned int parm2; /* this_1EC */ - unsigned int parm3; /* this_1F0 */ - unsigned int this_1D0; + u32 parm0; /* this_1E4 */ + u32 parm1; /* this_1E8 */ + u32 parm2; /* this_1EC */ + u32 parm3; /* this_1F0 */ + u32 this_1D0; } wt_voice_t; #endif /* _AU88X0_WT_H */ diff --git a/sound/pci/au88x0/au88x0_xtalk.c b/sound/pci/au88x0/au88x0_xtalk.c index df915fa3f88d1f..4534e1882ada6f 100644 --- a/sound/pci/au88x0/au88x0_xtalk.c +++ b/sound/pci/au88x0/au88x0_xtalk.c @@ -562,7 +562,7 @@ static void vortex_XtalkHw_SetDelay(vortex_t * vortex, unsigned short right, unsigned short left) { - int esp0 = 0; + u32 esp0 = 0; esp0 &= 0x1FFFFFFF; esp0 |= 0xA0000000; @@ -632,18 +632,18 @@ static void vortex_XtalkHw_GetRightDline(vortex_t * vortex, xtalk_dline_t dline) /* Control/Global stuff */ #if 0 -static void vortex_XtalkHw_SetControlReg(vortex_t * vortex, unsigned long ctrl) +static void vortex_XtalkHw_SetControlReg(vortex_t * vortex, u32 ctrl) { hwwrite(vortex->mmio, 0x24660, ctrl); } -static void vortex_XtalkHw_GetControlReg(vortex_t * vortex, unsigned long *ctrl) +static void vortex_XtalkHw_GetControlReg(vortex_t * vortex, u32 *ctrl) { *ctrl = hwread(vortex->mmio, 0x24660); } #endif -static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, int sr) +static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, u32 sr) { - int temp; + u32 temp; temp = (hwread(vortex->mmio, 0x24660) & 0x1FFFFFFF) | 0xC0000000; temp = (temp & 0xffffff07) | ((sr & 0x1f) << 3); @@ -651,7 +651,7 @@ static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, int sr) } #if 0 -static void vortex_XtalkHw_GetSampleRate(vortex_t * vortex, int *sr) +static void vortex_XtalkHw_GetSampleRate(vortex_t * vortex, u32 *sr) { *sr = (hwread(vortex->mmio, 0x24660) >> 3) & 0x1f; } @@ -659,7 +659,7 @@ static void vortex_XtalkHw_GetSampleRate(vortex_t * vortex, int *sr) #endif static void vortex_XtalkHw_Enable(vortex_t * vortex) { - int temp; + u32 temp; temp = (hwread(vortex->mmio, 0x24660) & 0x1FFFFFFF) | 0xC0000000; temp |= 1; @@ -669,7 +669,7 @@ static void vortex_XtalkHw_Enable(vortex_t * vortex) static void vortex_XtalkHw_Disable(vortex_t * vortex) { - int temp; + u32 temp; temp = (hwread(vortex->mmio, 0x24660) & 0x1FFFFFFF) | 0xC0000000; temp &= 0xfffffffe; diff --git a/sound/pci/au88x0/au88x0_xtalk.h b/sound/pci/au88x0/au88x0_xtalk.h index 0b8d7b64012daf..7f4534b94d000d 100644 --- a/sound/pci/au88x0/au88x0_xtalk.h +++ b/sound/pci/au88x0/au88x0_xtalk.h @@ -39,16 +39,16 @@ #define XT_SPEAKER1 3 #define XT_DIAMOND 4 -typedef long xtalk_dline_t[XTDLINE_SZ]; -typedef short xtalk_gains_t[XTGAINS_SZ]; -typedef short xtalk_instate_t[XTINST_SZ]; -typedef short xtalk_coefs_t[5][5]; -typedef short xtalk_state_t[5][4]; +typedef u32 xtalk_dline_t[XTDLINE_SZ]; +typedef u16 xtalk_gains_t[XTGAINS_SZ]; +typedef u16 xtalk_instate_t[XTINST_SZ]; +typedef u16 xtalk_coefs_t[5][5]; +typedef u16 xtalk_state_t[5][4]; static void vortex_XtalkHw_SetGains(vortex_t * vortex, xtalk_gains_t const gains); static void vortex_XtalkHw_SetGainsAllChan(vortex_t * vortex); -static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, int sr); +static void vortex_XtalkHw_SetSampleRate(vortex_t * vortex, u32 sr); static void vortex_XtalkHw_ProgramPipe(vortex_t * vortex); static void vortex_XtalkHw_ProgramPipe(vortex_t * vortex); static void vortex_XtalkHw_ProgramXtalkWide(vortex_t * vortex); -- cgit 1.2.3-korg From f7cbb7fcd3bae5264a079e06411b35366da9bd4d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 13 Jan 2006 18:48:06 +0100 Subject: [ALSA] Fix snd_xxx_t typedefs Modules: PXA Mainstone driver,CX88 driver,SAA7134 driver Replace snd_xxx_t typedefs with explicit structs. Signed-off-by: Takashi Iwai --- arch/arm/mach-pxa/mainstone.c | 4 +- drivers/media/video/cx88/cx88-alsa.c | 54 +++++++++++++------------ drivers/media/video/saa7134/saa7134-alsa.c | 65 +++++++++++++++++------------- drivers/media/video/saa7134/saa7134.h | 2 +- include/asm-arm/arch-pxa/audio.h | 4 +- 5 files changed, 70 insertions(+), 59 deletions(-) diff --git a/arch/arm/mach-pxa/mainstone.c b/arch/arm/mach-pxa/mainstone.c index d5bda60209ecaf..98356f810007fa 100644 --- a/arch/arm/mach-pxa/mainstone.c +++ b/arch/arm/mach-pxa/mainstone.c @@ -157,14 +157,14 @@ static struct platform_device smc91x_device = { .resource = smc91x_resources, }; -static int mst_audio_startup(snd_pcm_substream_t *substream, void *priv) +static int mst_audio_startup(struct snd_pcm_substream *substream, void *priv) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF; return 0; } -static void mst_audio_shutdown(snd_pcm_substream_t *substream, void *priv) +static void mst_audio_shutdown(struct snd_pcm_substream *substream, void *priv) { if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF; diff --git a/drivers/media/video/cx88/cx88-alsa.c b/drivers/media/video/cx88/cx88-alsa.c index 2acccd6d49bca6..c841914c0244fd 100644 --- a/drivers/media/video/cx88/cx88-alsa.c +++ b/drivers/media/video/cx88/cx88-alsa.c @@ -63,7 +63,7 @@ struct cx88_audio_dev { /* audio controls */ int irq; - snd_card_t *card; + struct snd_card *card; spinlock_t reg_lock; @@ -82,7 +82,7 @@ struct cx88_audio_dev { struct cx88_buffer *buf; long opened; - snd_pcm_substream_t *substream; + struct snd_pcm_substream *substream; }; typedef struct cx88_audio_dev snd_cx88_card_t; @@ -96,7 +96,7 @@ typedef struct cx88_audio_dev snd_cx88_card_t; static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 1}; -static snd_card_t *snd_cx88_cards[SNDRV_CARDS]; +static struct snd_card *snd_cx88_cards[SNDRV_CARDS]; module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable cx88x soundcard. default enabled."); @@ -320,7 +320,7 @@ static int dsp_buffer_free(snd_cx88_card_t *chip) /* * Digital hardware definition */ -static snd_pcm_hardware_t snd_cx88_digital_hw = { +static struct snd_pcm_hardware snd_cx88_digital_hw = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -342,16 +342,16 @@ static snd_pcm_hardware_t snd_cx88_digital_hw = { /* * audio pcm capture runtime free */ -static void snd_card_cx88_runtime_free(snd_pcm_runtime_t *runtime) +static void snd_card_cx88_runtime_free(struct snd_pcm_runtime *runtime) { } /* * audio pcm capture open callback */ -static int snd_cx88_pcm_open(snd_pcm_substream_t *substream) +static int snd_cx88_pcm_open(struct snd_pcm_substream *substream) { snd_cx88_card_t *chip = snd_pcm_substream_chip(substream); - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; int err; if (test_and_set_bit(0, &chip->opened)) @@ -380,7 +380,7 @@ _error: /* * audio close callback */ -static int snd_cx88_close(snd_pcm_substream_t *substream) +static int snd_cx88_close(struct snd_pcm_substream *substream) { snd_cx88_card_t *chip = snd_pcm_substream_chip(substream); @@ -393,8 +393,8 @@ static int snd_cx88_close(snd_pcm_substream_t *substream) /* * hw_params callback */ -static int snd_cx88_hw_params(snd_pcm_substream_t * substream, - snd_pcm_hw_params_t * hw_params) +static int snd_cx88_hw_params(struct snd_pcm_substream * substream, + struct snd_pcm_hw_params * hw_params) { snd_cx88_card_t *chip = snd_pcm_substream_chip(substream); struct cx88_buffer *buf; @@ -453,7 +453,7 @@ static int snd_cx88_hw_params(snd_pcm_substream_t * substream, /* * hw free callback */ -static int snd_cx88_hw_free(snd_pcm_substream_t * substream) +static int snd_cx88_hw_free(struct snd_pcm_substream * substream) { snd_cx88_card_t *chip = snd_pcm_substream_chip(substream); @@ -469,7 +469,7 @@ static int snd_cx88_hw_free(snd_pcm_substream_t * substream) /* * prepare callback */ -static int snd_cx88_prepare(snd_pcm_substream_t *substream) +static int snd_cx88_prepare(struct snd_pcm_substream *substream) { return 0; } @@ -478,7 +478,7 @@ static int snd_cx88_prepare(snd_pcm_substream_t *substream) /* * trigger callback */ -static int snd_cx88_card_trigger(snd_pcm_substream_t *substream, int cmd) +static int snd_cx88_card_trigger(struct snd_pcm_substream *substream, int cmd) { snd_cx88_card_t *chip = snd_pcm_substream_chip(substream); int err; @@ -505,10 +505,10 @@ static int snd_cx88_card_trigger(snd_pcm_substream_t *substream, int cmd) /* * pointer callback */ -static snd_pcm_uframes_t snd_cx88_pointer(snd_pcm_substream_t *substream) +static snd_pcm_uframes_t snd_cx88_pointer(struct snd_pcm_substream *substream) { snd_cx88_card_t *chip = snd_pcm_substream_chip(substream); - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; if (chip->read_count) { chip->read_count -= snd_pcm_lib_period_bytes(substream); @@ -525,7 +525,7 @@ static snd_pcm_uframes_t snd_cx88_pointer(snd_pcm_substream_t *substream) /* * operators */ -static snd_pcm_ops_t snd_cx88_pcm_ops = { +static struct snd_pcm_ops snd_cx88_pcm_ops = { .open = snd_cx88_pcm_open, .close = snd_cx88_close, .ioctl = snd_pcm_lib_ioctl, @@ -542,7 +542,7 @@ static snd_pcm_ops_t snd_cx88_pcm_ops = { static int __devinit snd_cx88_pcm(snd_cx88_card_t *chip, int device, char *name) { int err; - snd_pcm_t *pcm; + struct snd_pcm *pcm; err = snd_pcm_new(chip->card, name, device, 0, 1, &pcm); if (err < 0) @@ -557,7 +557,8 @@ static int __devinit snd_cx88_pcm(snd_cx88_card_t *chip, int device, char *name) /**************************************************************************** CONTROL INTERFACE ****************************************************************************/ -static int snd_cx88_capture_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *info) +static int snd_cx88_capture_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) { info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; info->count = 1; @@ -568,7 +569,8 @@ static int snd_cx88_capture_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_i } /* OK - TODO: test it */ -static int snd_cx88_capture_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *value) +static int snd_cx88_capture_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) { snd_cx88_card_t *chip = snd_kcontrol_chip(kcontrol); struct cx88_core *core=chip->core; @@ -579,7 +581,8 @@ static int snd_cx88_capture_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_va } /* OK - TODO: test it */ -static int snd_cx88_capture_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *value) +static int snd_cx88_capture_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *value) { snd_cx88_card_t *chip = snd_kcontrol_chip(kcontrol); struct cx88_core *core=chip->core; @@ -595,7 +598,7 @@ static int snd_cx88_capture_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_va return v != old_control; } -static snd_kcontrol_new_t snd_cx88_capture_volume = { +static struct snd_kcontrol_new snd_cx88_capture_volume = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Volume", .info = snd_cx88_capture_volume_info, @@ -641,7 +644,7 @@ static int snd_cx88_free(snd_cx88_card_t *chip) /* * Component Destructor */ -static void snd_cx88_dev_free(snd_card_t * card) +static void snd_cx88_dev_free(struct snd_card * card) { snd_cx88_card_t *chip = card->private_data; @@ -654,8 +657,9 @@ static void snd_cx88_dev_free(snd_card_t * card) */ static int devno; -static int __devinit snd_cx88_create(snd_card_t *card, struct pci_dev *pci, - snd_cx88_card_t **rchip) +static int __devinit snd_cx88_create(struct snd_card *card, + struct pci_dev *pci, + snd_cx88_card_t **rchip) { snd_cx88_card_t *chip; struct cx88_core *core; @@ -726,7 +730,7 @@ static int __devinit snd_cx88_create(snd_card_t *card, struct pci_dev *pci, static int __devinit cx88_audio_initdev(struct pci_dev *pci, const struct pci_device_id *pci_id) { - snd_card_t *card; + struct snd_card *card; snd_cx88_card_t *chip; int err; diff --git a/drivers/media/video/saa7134/saa7134-alsa.c b/drivers/media/video/saa7134/saa7134-alsa.c index 7df5e0826e123f..e02e6ee31b7826 100644 --- a/drivers/media/video/saa7134/saa7134-alsa.c +++ b/drivers/media/video/saa7134/saa7134-alsa.c @@ -71,7 +71,7 @@ MODULE_PARM_DESC(enable, "Enable (or not) the SAA7134 capture interface(s)."); */ typedef struct snd_card_saa7134 { - snd_card_t *card; + struct snd_card *card; spinlock_t mixer_lock; int mixer_volume[MIXER_ADDR_LAST+1][2]; int capture_source[MIXER_ADDR_LAST+1][2]; @@ -95,10 +95,10 @@ typedef struct snd_card_saa7134_pcm { spinlock_t lock; - snd_pcm_substream_t *substream; + struct snd_pcm_substream *substream; } snd_card_saa7134_pcm_t; -static snd_card_t *snd_saa7134_cards[SNDRV_CARDS]; +static struct snd_card *snd_saa7134_cards[SNDRV_CARDS]; /* @@ -251,10 +251,10 @@ out: * */ -static int snd_card_saa7134_capture_trigger(snd_pcm_substream_t * substream, +static int snd_card_saa7134_capture_trigger(struct snd_pcm_substream * substream, int cmd) { - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; snd_card_saa7134_pcm_t *pcm = runtime->private_data; struct saa7134_dev *dev=pcm->dev; int err = 0; @@ -333,9 +333,9 @@ static int dsp_buffer_free(struct saa7134_dev *dev) * */ -static int snd_card_saa7134_capture_prepare(snd_pcm_substream_t * substream) +static int snd_card_saa7134_capture_prepare(struct snd_pcm_substream * substream) { - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; int bswap, sign; u32 fmt, control; snd_card_saa7134_t *saa7134 = snd_pcm_substream_chip(substream); @@ -422,9 +422,10 @@ static int snd_card_saa7134_capture_prepare(snd_pcm_substream_t * substream) * */ -static snd_pcm_uframes_t snd_card_saa7134_capture_pointer(snd_pcm_substream_t * substream) +static snd_pcm_uframes_t +snd_card_saa7134_capture_pointer(struct snd_pcm_substream * substream) { - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; snd_card_saa7134_pcm_t *pcm = runtime->private_data; struct saa7134_dev *dev=pcm->dev; @@ -442,7 +443,7 @@ static snd_pcm_uframes_t snd_card_saa7134_capture_pointer(snd_pcm_substream_t * * ALSA hardware capabilities definition */ -static snd_pcm_hardware_t snd_card_saa7134_capture = +static struct snd_pcm_hardware snd_card_saa7134_capture = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | @@ -465,7 +466,7 @@ static snd_pcm_hardware_t snd_card_saa7134_capture = .periods_max = 1024, }; -static void snd_card_saa7134_runtime_free(snd_pcm_runtime_t *runtime) +static void snd_card_saa7134_runtime_free(struct snd_pcm_runtime *runtime) { snd_card_saa7134_pcm_t *pcm = runtime->private_data; @@ -482,8 +483,8 @@ static void snd_card_saa7134_runtime_free(snd_pcm_runtime_t *runtime) * */ -static int snd_card_saa7134_hw_params(snd_pcm_substream_t * substream, - snd_pcm_hw_params_t * hw_params) +static int snd_card_saa7134_hw_params(struct snd_pcm_substream * substream, + struct snd_pcm_hw_params * hw_params) { snd_card_saa7134_t *saa7134 = snd_pcm_substream_chip(substream); struct saa7134_dev *dev; @@ -562,7 +563,7 @@ static int snd_card_saa7134_hw_params(snd_pcm_substream_t * substream, * */ -static int snd_card_saa7134_hw_free(snd_pcm_substream_t * substream) +static int snd_card_saa7134_hw_free(struct snd_pcm_substream * substream) { snd_card_saa7134_t *saa7134 = snd_pcm_substream_chip(substream); struct saa7134_dev *dev; @@ -588,7 +589,7 @@ static int snd_card_saa7134_hw_free(snd_pcm_substream_t * substream) * */ -static int snd_card_saa7134_capture_close(snd_pcm_substream_t * substream) +static int snd_card_saa7134_capture_close(struct snd_pcm_substream * substream) { return 0; } @@ -603,9 +604,9 @@ static int snd_card_saa7134_capture_close(snd_pcm_substream_t * substream) * */ -static int snd_card_saa7134_capture_open(snd_pcm_substream_t * substream) +static int snd_card_saa7134_capture_open(struct snd_pcm_substream * substream) { - snd_pcm_runtime_t *runtime = substream->runtime; + struct snd_pcm_runtime *runtime = substream->runtime; snd_card_saa7134_pcm_t *pcm; snd_card_saa7134_t *saa7134 = snd_pcm_substream_chip(substream); struct saa7134_dev *dev = saa7134->dev; @@ -641,7 +642,7 @@ static int snd_card_saa7134_capture_open(snd_pcm_substream_t * substream) * ALSA capture callbacks definition */ -static snd_pcm_ops_t snd_card_saa7134_capture_ops = { +static struct snd_pcm_ops snd_card_saa7134_capture_ops = { .open = snd_card_saa7134_capture_open, .close = snd_card_saa7134_capture_close, .ioctl = snd_pcm_lib_ioctl, @@ -662,7 +663,7 @@ static snd_pcm_ops_t snd_card_saa7134_capture_ops = { static int snd_card_saa7134_pcm(snd_card_saa7134_t *saa7134, int device) { - snd_pcm_t *pcm; + struct snd_pcm *pcm; int err; if ((err = snd_pcm_new(saa7134->card, "SAA7134 PCM", device, 0, 1, &pcm)) < 0) @@ -680,7 +681,8 @@ static int snd_card_saa7134_pcm(snd_card_saa7134_t *saa7134, int device) .get = snd_saa7134_volume_get, .put = snd_saa7134_volume_put, \ .private_value = addr } -static int snd_saa7134_volume_info(snd_kcontrol_t * kcontrol, snd_ctl_elem_info_t * uinfo) +static int snd_saa7134_volume_info(struct snd_kcontrol * kcontrol, + struct snd_ctl_elem_info * uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; @@ -689,7 +691,8 @@ static int snd_saa7134_volume_info(snd_kcontrol_t * kcontrol, snd_ctl_elem_info_ return 0; } -static int snd_saa7134_volume_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol) +static int snd_saa7134_volume_get(struct snd_kcontrol * kcontrol, + struct snd_ctl_elem_value * ucontrol) { snd_card_saa7134_t *chip = snd_kcontrol_chip(kcontrol); int addr = kcontrol->private_value; @@ -699,7 +702,8 @@ static int snd_saa7134_volume_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_ return 0; } -static int snd_saa7134_volume_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol) +static int snd_saa7134_volume_put(struct snd_kcontrol * kcontrol, + struct snd_ctl_elem_value * ucontrol) { snd_card_saa7134_t *chip = snd_kcontrol_chip(kcontrol); int change, addr = kcontrol->private_value; @@ -730,7 +734,8 @@ static int snd_saa7134_volume_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_ .get = snd_saa7134_capsrc_get, .put = snd_saa7134_capsrc_put, \ .private_value = addr } -static int snd_saa7134_capsrc_info(snd_kcontrol_t * kcontrol, snd_ctl_elem_info_t * uinfo) +static int snd_saa7134_capsrc_info(struct snd_kcontrol * kcontrol, + struct snd_ctl_elem_info * uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; uinfo->count = 2; @@ -739,7 +744,8 @@ static int snd_saa7134_capsrc_info(snd_kcontrol_t * kcontrol, snd_ctl_elem_info_ return 0; } -static int snd_saa7134_capsrc_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol) +static int snd_saa7134_capsrc_get(struct snd_kcontrol * kcontrol, + struct snd_ctl_elem_value * ucontrol) { snd_card_saa7134_t *chip = snd_kcontrol_chip(kcontrol); int addr = kcontrol->private_value; @@ -752,7 +758,8 @@ static int snd_saa7134_capsrc_get(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_ return 0; } -static int snd_saa7134_capsrc_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol) +static int snd_saa7134_capsrc_put(struct snd_kcontrol * kcontrol, + struct snd_ctl_elem_value * ucontrol) { snd_card_saa7134_t *chip = snd_kcontrol_chip(kcontrol); int change, addr = kcontrol->private_value; @@ -829,7 +836,7 @@ static int snd_saa7134_capsrc_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_ return change; } -static snd_kcontrol_new_t snd_saa7134_controls[] = { +static struct snd_kcontrol_new snd_saa7134_controls[] = { SAA713x_VOLUME("Video Volume", 0, MIXER_ADDR_TVTUNER), SAA713x_CAPSRC("Video Capture Switch", 0, MIXER_ADDR_TVTUNER), SAA713x_VOLUME("Line Volume", 1, MIXER_ADDR_LINE1), @@ -848,7 +855,7 @@ SAA713x_CAPSRC("Line Capture Switch", 2, MIXER_ADDR_LINE2), static int snd_card_saa7134_new_mixer(snd_card_saa7134_t * chip) { - snd_card_t *card = chip->card; + struct snd_card *card = chip->card; unsigned int idx; int err; @@ -862,7 +869,7 @@ static int snd_card_saa7134_new_mixer(snd_card_saa7134_t * chip) return 0; } -static void snd_saa7134_free(snd_card_t * card) +static void snd_saa7134_free(struct snd_card * card) { snd_card_saa7134_t *chip = card->private_data; @@ -889,7 +896,7 @@ static void snd_saa7134_free(snd_card_t * card) static int alsa_card_saa7134_create(struct saa7134_dev *dev, int devnum) { - snd_card_t *card; + struct snd_card *card; snd_card_saa7134_t *chip; int err; diff --git a/drivers/media/video/saa7134/saa7134.h b/drivers/media/video/saa7134/saa7134.h index 3261d8bebdd1ce..6873d9a85ef125 100644 --- a/drivers/media/video/saa7134/saa7134.h +++ b/drivers/media/video/saa7134/saa7134.h @@ -386,7 +386,7 @@ struct saa7134_dmasound { unsigned int read_offset; unsigned int read_count; void * priv_data; - snd_pcm_substream_t *substream; + struct snd_pcm_substream *substream; }; /* IR input */ diff --git a/include/asm-arm/arch-pxa/audio.h b/include/asm-arm/arch-pxa/audio.h index 60976f830e3f65..17eccd72013650 100644 --- a/include/asm-arm/arch-pxa/audio.h +++ b/include/asm-arm/arch-pxa/audio.h @@ -6,8 +6,8 @@ #include typedef struct { - int (*startup)(snd_pcm_substream_t *, void *); - void (*shutdown)(snd_pcm_substream_t *, void *); + int (*startup)(struct snd_pcm_substream *, void *); + void (*shutdown)(struct snd_pcm_substream *, void *); void (*suspend)(void *); void (*resume)(void *); void *priv; -- cgit 1.2.3-korg From 415b09e4559b0c95666af3cebe918386212aef98 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 16 Jan 2006 08:03:52 +0100 Subject: [ALSA] usb-audio: add Edirol PC-50 support Modules: USB generic driver Add a quirk entry for the Edirol PC-50. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 381d9a8ccee1b8..a433c6f2128d4b 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1094,7 +1094,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* TODO: add Roland G-70 support */ /* TODO: add Roland V-SYNTH XT support */ /* TODO: add BOSS GT-PRO support */ - /* TODO: add Edirol PC-50 support */ +{ + /* has ID 0x008c when not in "Advanced Driver" mode */ + USB_DEVICE(0x0582, 0x008b), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "EDIROL", + .product_name = "PC-50", + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + } +}, /* TODO: add Edirol PC-80 support */ /* TODO: add Edirol UA-1EX support */ { -- cgit 1.2.3-korg From c1f6d41593d3ef1f778fd6892cc4b916050c7cb9 Mon Sep 17 00:00:00 2001 From: Mark Salazar Date: Mon, 16 Jan 2006 11:29:09 +0100 Subject: [ALSA] #1/4 for Zoom Video - resolve common vs chipset specific mixer controls Modules: ES18xx driver First of 4 es18xx.c patches culminating in Zoom Video support. While adding support for Zoom Video to the es18xx driver I found some of the mixer controls were wrong. Since you guys went to the trouble of supplying the datasheets for the supported chipsets I did a review of all of them and tried to get es18xx.c to accurately reflect the proper mixer controls for each chipset. If the datasheets are wrong then so are my patches. This first patch moves some controls from the common-to-all-chipsets array 'snd_es18xx_base_controls' to a chipset-specific array and adds code to manage that new array. Also while testing on my ES1878 test machine I discovered it needed a couple of udelays in the identify function so those are in this patch as well. Testing: This work was initially done on the source from the Debian Sarge ALSA package, then tested on an ES1879 and an ES1878 machine. Patches were created against the Sarge code and then edited to apply correctly to the ALSA cvs code. Lastly the patched ALSA cvs code was test for successful compilation. No additional testing was done on the ALSA cvs version. Signed-off-by: Mark Salazar Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 49 ++++++++++++++++++++++++++++++++++++++----------- 1 file changed, 38 insertions(+), 11 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 08f032b5110716..bde9860327fef9 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1191,19 +1191,22 @@ static int snd_es18xx_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e return change; } +/* Mixer controls + * These arrays contain setup data for mixer controls. + * + * The controls that are universal to all chipsets are fully initialized + * here. + */ static struct snd_kcontrol_new snd_es18xx_base_controls[] = { ES18XX_DOUBLE("Master Playback Volume", 0, 0x60, 0x62, 0, 0, 63, 0), ES18XX_DOUBLE("Master Playback Switch", 0, 0x60, 0x62, 6, 6, 1, 1), ES18XX_DOUBLE("Line Playback Volume", 0, 0x3e, 0x3e, 4, 0, 15, 0), ES18XX_DOUBLE("CD Playback Volume", 0, 0x38, 0x38, 4, 0, 15, 0), ES18XX_DOUBLE("FM Playback Volume", 0, 0x36, 0x36, 4, 0, 15, 0), -ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mic Playback Volume", 0, 0x1a, 0x1a, 4, 0, 15, 0), ES18XX_DOUBLE("Aux Playback Volume", 0, 0x3a, 0x3a, 4, 0, 15, 0), -ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0), ES18XX_SINGLE("Record Monitor", 0, 0xa8, 3, 1, 0), ES18XX_DOUBLE("Capture Volume", 0, 0xb4, 0xb4, 4, 0, 15, 0), -ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -1213,19 +1216,27 @@ ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), } }; -static struct snd_kcontrol_new snd_es18xx_mono_in_control = -ES18XX_DOUBLE("Mono Input Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0); - static struct snd_kcontrol_new snd_es18xx_recmix_controls[] = { ES18XX_DOUBLE("PCM Capture Volume", 0, 0x69, 0x69, 4, 0, 15, 0), ES18XX_DOUBLE("Mic Capture Volume", 0, 0x68, 0x68, 4, 0, 15, 0), ES18XX_DOUBLE("Line Capture Volume", 0, 0x6e, 0x6e, 4, 0, 15, 0), ES18XX_DOUBLE("FM Capture Volume", 0, 0x6b, 0x6b, 4, 0, 15, 0), -ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0), ES18XX_DOUBLE("CD Capture Volume", 0, 0x6a, 0x6a, 4, 0, 15, 0), ES18XX_DOUBLE("Aux Capture Volume", 0, 0x6c, 0x6c, 4, 0, 15, 0) }; +/* + * The chipset specific mixer controls + */ +static struct snd_kcontrol_new snd_es18xx_opt_speaker = + ES18XX_SINGLE("PC Speaker Playback Volume", 0, 0x3c, 0, 7, 0); + +static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { +ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), +ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) +}; + static struct snd_kcontrol_new snd_es18xx_pcm1_controls[] = { ES18XX_DOUBLE("PCM Playback Volume", 0, 0x14, 0x14, 4, 0, 15, 0), }; @@ -1476,11 +1487,14 @@ static int __devinit snd_es18xx_identify(struct snd_es18xx *chip) } outb(0x40, chip->port + 0x04); + udelay(10); hi = inb(chip->port + 0x05); + udelay(10); lo = inb(chip->port + 0x05); if (hi != lo) { chip->version = hi << 8 | lo; chip->ctrl_port = inb(chip->port + 0x05) << 8; + udelay(10); chip->ctrl_port += inb(chip->port + 0x05); if ((chip->res_ctrl_port = request_region(chip->ctrl_port, 8, "ES18xx - CTRL")) == NULL) { @@ -1778,10 +1792,6 @@ static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) } } - if (chip->caps & ES18XX_MONO) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_mono_in_control, chip))) < 0) - return err; - } if (chip->caps & ES18XX_RECMIX) { for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_recmix_controls); idx++) { if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_recmix_controls[idx], chip))) < 0) @@ -1819,6 +1829,23 @@ static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) } } + /* finish initializing other chipset specific controls + */ + if (chip->version != 0x1868) { + err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_opt_speaker, + chip)); + if (err < 0) + return err; + } + if (chip->version == 0x1869) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_1869); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_1869[idx], + chip)); + if (err < 0) + return err; + } + } return 0; } -- cgit 1.2.3-korg From f4df221f8fe129ac2fa2a2a4306b7355cf7d05d6 Mon Sep 17 00:00:00 2001 From: Mark Salazar Date: Mon, 16 Jan 2006 11:31:14 +0100 Subject: [ALSA] #2/4 for Zoom Video - resolve number of record sources Modules: ES18xx driver Second of 4 es18xx.c patches culminating in Zoom Video support. This patch changes the 'record source' mux routines to reflect the fact that not all of the supported chipsets have 8 possible inputs. Some have 4 and some have 5. Testing: This work was initially done on the source from the Debian Sarge ALSA package, then tested on an ES1879 and an ES1878 machine. Patches were created against the Sarge code and then edited to apply correctly to the ALSA cvs code. Lastly the patched ALSA cvs code was test for successful compilation. No additional testing was done on the ALSA cvs version. Signed-off-by: Mark Salazar Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 99 +++++++++++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 90 insertions(+), 9 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index bde9860327fef9..0488eba051e9c0 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -939,37 +939,118 @@ static int snd_es18xx_capture_close(struct snd_pcm_substream *substream) * MIXER part */ +/* Record source mux routines: + * Depending on the chipset this mux switches between 4, 5, or 8 possible inputs. + * bit table for the 4/5 source mux: + * reg 1C: + * b2 b1 b0 muxSource + * x 0 x microphone + * 0 1 x CD + * 1 1 0 line + * 1 1 1 mixer + * if it's "mixer" and it's a 5 source mux chipset then reg 7A bit 3 determines + * either the play mixer or the capture mixer. + * + * "map4Source" translates from source number to reg bit pattern + * "invMap4Source" translates from reg bit pattern to source number + */ + static int snd_es18xx_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[8] = { + static char *texts4Source[4] = { + "Mic", "CD", "Line", "Master" + }; + static char *texts5Source[5] = { + "Mic", "CD", "Line", "Master", "Mix" + }; + static char *texts8Source[8] = { "Mic", "Mic Master", "CD", "AOUT", "Mic1", "Mix", "Line", "Master" }; + struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 8; - if (uinfo->value.enumerated.item > 7) - uinfo->value.enumerated.item = 7; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + switch (chip->version) { + case 0x1868: + case 0x1878: + uinfo->value.enumerated.items = 4; + if (uinfo->value.enumerated.item > 3) + uinfo->value.enumerated.item = 3; + strcpy(uinfo->value.enumerated.name, texts4Source[uinfo->value.enumerated.item]); + break; + case 0x1887: + case 0x1888: + uinfo->value.enumerated.items = 5; + if (uinfo->value.enumerated.item > 4) + uinfo->value.enumerated.item = 4; + strcpy(uinfo->value.enumerated.name, texts5Source[uinfo->value.enumerated.item]); + break; + case 0x1869: /* DS somewhat contradictory for 1869: could be be 5 or 8 */ + case 0x1879: + uinfo->value.enumerated.items = 8; + if (uinfo->value.enumerated.item > 7) + uinfo->value.enumerated.item = 7; + strcpy(uinfo->value.enumerated.name, texts8Source[uinfo->value.enumerated.item]); + break; + default: + return -EINVAL; + } return 0; } static int snd_es18xx_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + static unsigned char invMap4Source[8] = {0, 0, 1, 1, 0, 0, 2, 3}; struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.enumerated.item[0] = snd_es18xx_mixer_read(chip, 0x1c) & 0x07; + int muxSource = snd_es18xx_mixer_read(chip, 0x1c) & 0x07; + if (!(chip->version == 0x1869 || chip->version == 0x1879)) { + muxSource = invMap4Source[muxSource]; + if (muxSource==3 && + (chip->version == 0x1887 || chip->version == 0x1888) && + (snd_es18xx_mixer_read(chip, 0x7a) & 0x08) + ) + muxSource = 4; + } + ucontrol->value.enumerated.item[0] = muxSource; return 0; } static int snd_es18xx_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + static unsigned char map4Source[4] = {0, 2, 6, 7}; struct snd_es18xx *chip = snd_kcontrol_chip(kcontrol); unsigned char val = ucontrol->value.enumerated.item[0]; - - if (val > 7) + unsigned char retVal = 0; + + switch (chip->version) { + /* 5 source chips */ + case 0x1887: + case 0x1888: + if (val > 4) + return -EINVAL; + if (val == 4) { + retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x08) != 0x08; + val = 3; + } else + retVal = snd_es18xx_mixer_bits(chip, 0x7a, 0x08, 0x00) != 0x00; + /* 4 source chips */ + case 0x1868: + case 0x1878: + if (val > 3) + return -EINVAL; + val = map4Source[val]; + break; + /* 8 source chips */ + case 0x1869: + case 0x1879: + if (val > 7) + return -EINVAL; + break; + default: return -EINVAL; - return snd_es18xx_mixer_bits(chip, 0x1c, 0x07, val) != val; + } + return (snd_es18xx_mixer_bits(chip, 0x1c, 0x07, val) != val) || retVal; } static int snd_es18xx_info_spatializer_enable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -- cgit 1.2.3-korg From 14086771c04acecc81e3985ca3118b330324194d Mon Sep 17 00:00:00 2001 From: Mark Salazar Date: Mon, 16 Jan 2006 11:33:52 +0100 Subject: [ALSA] #3/4 for Zoom Video - change Hardware Volume interrupt handling Modules: ES18xx driver Third of 4 es18xx.c patches culminating in Zoom Video support. This patch changes the Hardware Volume support to reflect the fact that not all of the supported chipsets have seperate registers dedicated to the Hardware Volume inputs. Although all the chipsets can generate an HWV interrupt whenever a Hardware Volume input is received only those with seperate HWV registers can split the HWV registers from the Master volume registers. Testing: This work was initially done on the source from the Debian Sarge ALSA package, then tested on an ES1879 and an ES1878 machine. Patches were created against the Sarge code and then edited to apply correctly to the ALSA cvs code. Lastly the patched ALSA cvs code was test for successful compilation. No additional testing was done on the ALSA cvs version. Signed-off-by: Mark Salazar Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 0488eba051e9c0..93335000c51e6a 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -148,7 +148,7 @@ struct snd_audiodrive { #define ES18XX_DUPLEX_SAME 0x0010 /* Playback and record must share the same rate */ #define ES18XX_NEW_RATE 0x0020 /* More precise rate setting */ #define ES18XX_AUXB 0x0040 /* AuxB mixer control */ -#define ES18XX_HWV 0x0080 /* Has hardware volume */ +#define ES18XX_HWV 0x0080 /* Has seperate hardware volume mixer controls*/ #define ES18XX_MONO 0x0100 /* Mono_in mixer control */ #define ES18XX_I2S 0x0200 /* I2S mixer control */ #define ES18XX_MUTEREC 0x0400 /* Record source can be muted */ @@ -788,9 +788,12 @@ static irqreturn_t snd_es18xx_interrupt(int irq, void *dev_id, struct pt_regs *r /* Hardware volume */ if (status & HWV_IRQ) { - int split = snd_es18xx_mixer_read(chip, 0x64) & 0x80; - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id); - snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id); + int split = 0; + if (chip->caps & ES18XX_HWV) { + split = snd_es18xx_mixer_read(chip, 0x64) & 0x80; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_switch->id); + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->hw_volume->id); + } if (!split) { snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_switch->id); snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, &chip->master_volume->id); @@ -1614,22 +1617,22 @@ static int __devinit snd_es18xx_probe(struct snd_es18xx *chip) switch (chip->version) { case 0x1868: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_CONTROL; break; case 0x1869: chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_MONO | ES18XX_MUTEREC | ES18XX_CONTROL | ES18XX_HWV; break; case 0x1878: - chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV; + chip->caps = ES18XX_DUPLEX_MONO | ES18XX_DUPLEX_SAME | ES18XX_I2S | ES18XX_CONTROL; break; case 0x1879: chip->caps = ES18XX_PCM2 | ES18XX_SPATIALIZER | ES18XX_RECMIX | ES18XX_NEW_RATE | ES18XX_AUXB | ES18XX_I2S | ES18XX_CONTROL | ES18XX_HWV; break; case 0x1887: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; break; case 0x1888: - chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_HWV; + chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME; break; default: snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", -- cgit 1.2.3-korg From 95b712965f0a50365cc0128dacc27acf562f2ff1 Mon Sep 17 00:00:00 2001 From: Mark Salazar Date: Mon, 16 Jan 2006 11:35:40 +0100 Subject: [ALSA] #4/4 for Zoom Video - add Zoom Video support Modules: ES18xx driver Forth of 4 es18xx.c patches culminating in Zoom Video support. This patch adds Zoom Video support for those chipsets that support it. Testing: This work was initially done on the source from the Debian Sarge ALSA package, then tested on an ES1879. I could not test the Zoom Video function for an ES1878 or ES1869. Patches were created against the Sarge code and then edited to apply correctly to the ALSA cvs code. Lastly the patched ALSA cvs code was test for successful compilation. No additional testing was done on the ALSA cvs version. One quirk (noted in my comments below) is that apparently the datasheet is wrong for one of the ES1879 Zoom Video 'enable' bits, because 1) if you set this bit it messes up PCM playback (speaker_test play a lower frequency) 2) even if you don't set this bit Zoom Video still works. I added a control to toggle the bit on just in case there might be a version of the ES1879 that requires it, but I expect noone will need it. Signed-off-by: Mark Salazar Signed-off-by: Takashi Iwai --- sound/isa/es18xx.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 48 insertions(+), 3 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 93335000c51e6a..79785808dd60b9 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -49,6 +49,10 @@ * - contrarily to some pages in DS_1869.PDF the rates can be set * independently. * + * - Zoom Video is implemented by sharing the FM DAC, thus the user can + * have either FM playback or Video playback but not both simultaneously. + * The Video Playback Switch mixer control toggles this choice. + * * BUGS: * * - There is a major trouble I noted: @@ -63,7 +67,16 @@ * */ - +/* + * ES1879 NOTES: + * - When Zoom Video is enabled (reg 0x71 bit 6 toggled on) the PCM playback + * seems to be effected (speaker_test plays a lower frequency). Can't find + * anything in the datasheet to account for this, so a Video Playback Switch + * control has been included to allow ZV to be enabled only when necessary. + * Then again on at least one test system the 0x71 bit 6 enable bit is not + * needed for ZV, so maybe the datasheet is entirely wrong here. + */ + #include #include #include @@ -1317,10 +1330,20 @@ static struct snd_kcontrol_new snd_es18xx_opt_speaker = static struct snd_kcontrol_new snd_es18xx_opt_1869[] = { ES18XX_SINGLE("Capture Switch", 0, 0x1c, 4, 1, 1), +ES18XX_SINGLE("Video Playback Switch", 0, 0x7f, 0, 1, 0), ES18XX_DOUBLE("Mono Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), ES18XX_DOUBLE("Mono Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) }; +static struct snd_kcontrol_new snd_es18xx_opt_1878 = + ES18XX_DOUBLE("Video Playback Volume", 0, 0x68, 0x68, 4, 0, 15, 0); + +static struct snd_kcontrol_new snd_es18xx_opt_1879[] = { +ES18XX_SINGLE("Video Playback Switch", 0, 0x71, 6, 1, 0), +ES18XX_DOUBLE("Video Playback Volume", 0, 0x6d, 0x6d, 4, 0, 15, 0), +ES18XX_DOUBLE("Video Capture Volume", 0, 0x6f, 0x6f, 4, 0, 15, 0) +}; + static struct snd_kcontrol_new snd_es18xx_pcm1_controls[] = { ES18XX_DOUBLE("PCM Playback Volume", 0, 0x14, 0x14, 4, 0, 15, 0), }; @@ -1365,7 +1388,6 @@ static struct snd_kcontrol_new snd_es18xx_hw_volume_controls[] = { ES18XX_SINGLE("Hardware Master Volume Split", 0, 0x64, 7, 1, 0), }; -#if 0 static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned char reg) { int data; @@ -1376,7 +1398,6 @@ static int __devinit snd_es18xx_config_read(struct snd_es18xx *chip, unsigned ch spin_unlock_irqrestore(&chip->ctrl_lock, flags); return data; } -#endif static void __devinit snd_es18xx_config_write(struct snd_es18xx *chip, unsigned char reg, unsigned char data) @@ -1522,6 +1543,17 @@ static int __devinit snd_es18xx_initialize(struct snd_es18xx *chip) snd_es18xx_mixer_write(chip, 0x58, 0x94); snd_es18xx_mixer_write(chip, 0x5a, 0x80); } + /* Flip the "enable I2S" bits for those chipsets that need it */ + switch (chip->version) { + case 0x1879: + //Leaving I2S enabled on the 1879 screws up the PCM playback (rate effected somehow) + //so a Switch control has been added to toggle this 0x71 bit on/off: + //snd_es18xx_mixer_bits(chip, 0x71, 0x40, 0x40); + /* Note: we fall through on purpose here. */ + case 0x1878: + snd_es18xx_config_write(chip, 0x29, snd_es18xx_config_read(chip, 0x29) | 0x40); + break; + } /* Mute input source */ if (chip->caps & ES18XX_MUTEREC) mask = 0x10; @@ -1929,6 +1961,19 @@ static int __devinit snd_es18xx_mixer(struct snd_es18xx *chip) if (err < 0) return err; } + } else if (chip->version == 0x1878) { + err = snd_ctl_add(card, snd_ctl_new1(&snd_es18xx_opt_1878, + chip)); + if (err < 0) + return err; + } else if (chip->version == 0x1879) { + for (idx = 0; idx < ARRAY_SIZE(snd_es18xx_opt_1879); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_es18xx_opt_1879[idx], + chip)); + if (err < 0) + return err; + } } return 0; } -- cgit 1.2.3-korg From f0283f45a04d5cf31512e5e390a38504d97e7a97 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 Jan 2006 12:59:01 +0100 Subject: [ALSA] au88x0 - Fix structs for equalizer Modules: au88x0 driver Fixed the structures for equalizer. Signed-off-by: Takashi Iwai --- sound/pci/au88x0/au88x0_eq.h | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/pci/au88x0/au88x0_eq.h b/sound/pci/au88x0/au88x0_eq.h index 717544af58adae..474cd0046294e2 100644 --- a/sound/pci/au88x0/au88x0_eq.h +++ b/sound/pci/au88x0/au88x0_eq.h @@ -13,18 +13,16 @@ typedef struct { u16 LeftCoefs[50]; //0x4 u16 RightCoefs[50]; // 0x68 - u16 LeftGains[20]; //0xd0 - u16 RightGains[20]; //0xe4 + u16 LeftGains[10]; //0xd0 + u16 RightGains[10]; //0xe4 } auxxEqCoeffSet_t; typedef struct { - u32 ptr_this00; /*CAsp4HwIO */ s32 this04; /* How many filters for each side (default = 10) */ s32 this08; /* inited to cero. Stereo flag? */ } eqhw_t; typedef struct { - u32 ptr_this00; /*CAsp4Core */ eqhw_t this04; /* CHwEq */ u16 this08; /* Bad codec flag ? SetBypassGain: bypass gain */ u16 this0a; @@ -32,8 +30,7 @@ typedef struct { u16 this0e; s32 this10; /* How many gains are used for each side (right or left). */ - u16 this14_array[32]; /* SetLeftGainsTarget: Left (and right?) EQ gains */ - s32 this24; + u16 this14_array[10]; /* SetLeftGainsTarget: Left (and right?) EQ gains */ s32 this28; /* flag related to EQ enabled or not. Gang flag ? */ s32 this54; /* SetBypass */ s32 this58; -- cgit 1.2.3-korg From 1a60d4c5a0c4028559585a74e48593b16e1ca9b2 Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Mon, 16 Jan 2006 16:29:08 +0100 Subject: [ALSA] semaphore -> mutex (core part) Semaphore to mutex conversion. The conversion was generated via scripts, and the result was validated automatically via a script as well. Signed-off-by: Ingo Molnar Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- include/sound/core.h | 8 ++--- include/sound/hwdep.h | 2 +- include/sound/info.h | 2 +- include/sound/mixer_oss.h | 2 +- include/sound/pcm.h | 2 +- include/sound/pcm_oss.h | 2 +- include/sound/rawmidi.h | 4 +-- include/sound/seq_instr.h | 2 +- sound/core/hwdep.c | 43 +++++++++++------------ sound/core/info.c | 27 ++++++++------- sound/core/info_oss.c | 13 +++---- sound/core/init.c | 2 +- sound/core/memalloc.c | 22 ++++++------ sound/core/oss/mixer_oss.c | 14 ++++---- sound/core/oss/pcm_oss.c | 30 ++++++++-------- sound/core/pcm.c | 40 +++++++++++----------- sound/core/pcm_native.c | 12 +++---- sound/core/rawmidi.c | 57 ++++++++++++++++--------------- sound/core/seq/oss/seq_oss.c | 27 ++++++++------- sound/core/seq/seq_clientmgr.c | 40 +++++++++++----------- sound/core/seq/seq_clientmgr.h | 2 +- sound/core/seq/seq_device.c | 53 +++++++++++++++-------------- sound/core/seq/seq_instr.c | 6 ++-- sound/core/seq/seq_midi.c | 20 +++++------ sound/core/seq/seq_ports.c | 12 +++---- sound/core/seq/seq_queue.c | 6 ++-- sound/core/seq/seq_queue.h | 2 +- sound/core/sound.c | 23 +++++++------ sound/core/sound_oss.c | 23 +++++++------ sound/core/timer.c | 77 +++++++++++++++++++++--------------------- 30 files changed, 293 insertions(+), 282 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index 3093e3ddcf36cd..144bdc2f217fab 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -23,7 +23,7 @@ */ #include /* wake_up() */ -#include /* struct semaphore */ +#include /* struct mutex */ #include /* struct rw_semaphore */ #include /* struct workqueue_struct */ #include /* pm_message_t */ @@ -137,7 +137,7 @@ struct snd_card { #ifdef CONFIG_PM unsigned int power_state; /* power state */ - struct semaphore power_lock; /* power lock */ + struct mutex power_lock; /* power lock */ wait_queue_head_t power_sleep; #endif @@ -150,12 +150,12 @@ struct snd_card { #ifdef CONFIG_PM static inline void snd_power_lock(struct snd_card *card) { - down(&card->power_lock); + mutex_lock(&card->power_lock); } static inline void snd_power_unlock(struct snd_card *card) { - up(&card->power_lock); + mutex_unlock(&card->power_lock); } static inline unsigned int snd_power_get_state(struct snd_card *card) diff --git a/include/sound/hwdep.h b/include/sound/hwdep.h index c679e5b3111135..94c387b5d72429 100644 --- a/include/sound/hwdep.h +++ b/include/sound/hwdep.h @@ -60,7 +60,7 @@ struct snd_hwdep { void *private_data; void (*private_free) (struct snd_hwdep *hwdep); - struct semaphore open_mutex; + struct mutex open_mutex; int used; unsigned int dsp_loaded; unsigned int exclusive: 1; diff --git a/include/sound/info.h b/include/sound/info.h index 8ea5c7497c03df..f23d8381c216a0 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -84,7 +84,7 @@ struct snd_info_entry { void *private_data; void (*private_free)(struct snd_info_entry *entry); struct proc_dir_entry *p; - struct semaphore access; + struct mutex access; }; #if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS) diff --git a/include/sound/mixer_oss.h b/include/sound/mixer_oss.h index ca5b4822b62c8e..197b9e3d612b96 100644 --- a/include/sound/mixer_oss.h +++ b/include/sound/mixer_oss.h @@ -61,7 +61,7 @@ struct snd_mixer_oss { unsigned int active_index); void *private_data_recsrc; void (*private_free_recsrc)(struct snd_mixer_oss *mixer); - struct semaphore reg_mutex; + struct mutex reg_mutex; struct snd_info_entry *proc_entry; int oss_dev_alloc; /* --- */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 314268a1104830..15b885660bf056 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -420,7 +420,7 @@ struct snd_pcm { char id[64]; char name[80]; struct snd_pcm_str streams[2]; - struct semaphore open_mutex; + struct mutex open_mutex; wait_queue_head_t open_wait; void *private_data; void (*private_free) (struct snd_pcm *pcm); diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index d6ec886637db38..bff0778e196937 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -75,7 +75,7 @@ struct snd_pcm_oss_substream { struct snd_pcm_oss_stream { struct snd_pcm_oss_setup *setup_list; /* setup list */ - struct semaphore setup_mutex; + struct mutex setup_mutex; struct snd_info_entry *proc_entry; }; diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index d19bddfbf9953b..584e73dd479319 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -26,7 +26,7 @@ #include #include #include -#include +#include #if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) #include "seq_device.h" @@ -130,7 +130,7 @@ struct snd_rawmidi { void *private_data; void (*private_free) (struct snd_rawmidi *rmidi); - struct semaphore open_mutex; + struct mutex open_mutex; wait_queue_head_t open_wait; struct snd_info_entry *dev; diff --git a/include/sound/seq_instr.h b/include/sound/seq_instr.h index db764f09efb7e2..f2db03bfd74e59 100644 --- a/include/sound/seq_instr.h +++ b/include/sound/seq_instr.h @@ -64,7 +64,7 @@ struct snd_seq_kinstr_list { spinlock_t lock; spinlock_t ops_lock; - struct semaphore ops_mutex; + struct mutex ops_mutex; unsigned long ops_flags; }; diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 618c43be0bc36e..2524e66eccdd28 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -36,7 +37,7 @@ MODULE_DESCRIPTION("Hardware dependent layer"); MODULE_LICENSE("GPL"); static LIST_HEAD(snd_hwdep_devices); -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); static int snd_hwdep_free(struct snd_hwdep *hwdep); static int snd_hwdep_dev_free(struct snd_device *device); @@ -111,7 +112,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) init_waitqueue_entry(&wait, current); add_wait_queue(&hw->open_wait, &wait); - down(&hw->open_mutex); + mutex_lock(&hw->open_mutex); while (1) { if (hw->exclusive && hw->used > 0) { err = -EBUSY; @@ -128,9 +129,9 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) } else break; set_current_state(TASK_INTERRUPTIBLE); - up(&hw->open_mutex); + mutex_unlock(&hw->open_mutex); schedule(); - down(&hw->open_mutex); + mutex_lock(&hw->open_mutex); if (signal_pending(current)) { err = -ERESTARTSYS; break; @@ -147,7 +148,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) hw->ops.release(hw, file); } } - up(&hw->open_mutex); + mutex_unlock(&hw->open_mutex); if (err < 0) module_put(hw->card->module); return err; @@ -157,7 +158,7 @@ static int snd_hwdep_release(struct inode *inode, struct file * file) { int err = -ENXIO; struct snd_hwdep *hw = file->private_data; - down(&hw->open_mutex); + mutex_lock(&hw->open_mutex); if (hw->ops.release) { err = hw->ops.release(hw, file); wake_up(&hw->open_wait); @@ -165,7 +166,7 @@ static int snd_hwdep_release(struct inode *inode, struct file * file) if (hw->used > 0) hw->used--; snd_card_file_remove(hw->card, file); - up(&hw->open_mutex); + mutex_unlock(&hw->open_mutex); module_put(hw->card->module); return err; } @@ -272,7 +273,7 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_MINOR_HWDEPS) { if (snd_hwdep_search(card, device)) @@ -281,7 +282,7 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, } if (device >= SNDRV_MINOR_HWDEPS) device = -1; - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (put_user(device, (int __user *)arg)) return -EFAULT; return 0; @@ -294,13 +295,13 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, if (get_user(device, &info->device)) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); hwdep = snd_hwdep_search(card, device); if (hwdep) err = snd_hwdep_info(hwdep, info); else err = -ENXIO; - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } } @@ -375,7 +376,7 @@ int snd_hwdep_new(struct snd_card *card, char *id, int device, return err; } init_waitqueue_head(&hwdep->open_wait); - init_MUTEX(&hwdep->open_mutex); + mutex_init(&hwdep->open_mutex); *rhwdep = hwdep; return 0; } @@ -401,9 +402,9 @@ static int snd_hwdep_dev_register(struct snd_device *device) int err; char name[32]; - down(®ister_mutex); + mutex_lock(®ister_mutex); if (snd_hwdep_search(hwdep->card, hwdep->device)) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EBUSY; } list_add_tail(&hwdep->list, &snd_hwdep_devices); @@ -414,7 +415,7 @@ static int snd_hwdep_dev_register(struct snd_device *device) snd_printk(KERN_ERR "unable to register hardware dependent device %i:%i\n", hwdep->card->number, hwdep->device); list_del(&hwdep->list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } #ifdef CONFIG_SND_OSSEMUL @@ -434,7 +435,7 @@ static int snd_hwdep_dev_register(struct snd_device *device) } } #endif - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -443,9 +444,9 @@ static int snd_hwdep_dev_unregister(struct snd_device *device) struct snd_hwdep *hwdep = device->device_data; snd_assert(hwdep != NULL, return -ENXIO); - down(®ister_mutex); + mutex_lock(®ister_mutex); if (snd_hwdep_search(hwdep->card, hwdep->device) != hwdep) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EINVAL; } #ifdef CONFIG_SND_OSSEMUL @@ -454,7 +455,7 @@ static int snd_hwdep_dev_unregister(struct snd_device *device) #endif snd_unregister_device(SNDRV_DEVICE_TYPE_HWDEP, hwdep->card, hwdep->device); list_del(&hwdep->list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return snd_hwdep_free(hwdep); } @@ -469,13 +470,13 @@ static void snd_hwdep_proc_read(struct snd_info_entry *entry, struct list_head *p; struct snd_hwdep *hwdep; - down(®ister_mutex); + mutex_lock(®ister_mutex); list_for_each(p, &snd_hwdep_devices) { hwdep = list_entry(p, struct snd_hwdep, list); snd_iprintf(buffer, "%02i-%02i: %s\n", hwdep->card->number, hwdep->device, hwdep->name); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); } static struct snd_info_entry *snd_hwdep_proc_entry; diff --git a/sound/core/info.c b/sound/core/info.c index af123e3bdb2478..2582b74d319953 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -31,6 +31,7 @@ #include #include #include +#include #include /* @@ -68,7 +69,7 @@ int snd_info_check_reserved_words(const char *str) return 1; } -static DECLARE_MUTEX(info_mutex); +static DEFINE_MUTEX(info_mutex); struct snd_info_private_data { struct snd_info_buffer *rbuffer; @@ -265,11 +266,11 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) struct proc_dir_entry *p; int mode, err; - down(&info_mutex); + mutex_lock(&info_mutex); p = PDE(inode); entry = p == NULL ? NULL : (struct snd_info_entry *)p->data; if (entry == NULL || entry->disconnected) { - up(&info_mutex); + mutex_unlock(&info_mutex); return -ENODEV; } if (!try_module_get(entry->module)) { @@ -361,13 +362,13 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) break; } file->private_data = data; - up(&info_mutex); + mutex_unlock(&info_mutex); if (entry->content == SNDRV_INFO_CONTENT_TEXT && (mode == O_RDONLY || mode == O_RDWR)) { if (entry->c.text.read) { - down(&entry->access); + mutex_lock(&entry->access); entry->c.text.read(entry, data->rbuffer); - up(&entry->access); + mutex_unlock(&entry->access); } } return 0; @@ -375,7 +376,7 @@ static int snd_info_entry_open(struct inode *inode, struct file *file) __error: module_put(entry->module); __error1: - up(&info_mutex); + mutex_unlock(&info_mutex); return err; } @@ -747,7 +748,7 @@ static struct snd_info_entry *snd_info_create_entry(const char *name) } entry->mode = S_IFREG | S_IRUGO; entry->content = SNDRV_INFO_CONTENT_TEXT; - init_MUTEX(&entry->access); + mutex_init(&entry->access); return entry; } @@ -896,10 +897,10 @@ int snd_info_register(struct snd_info_entry * entry) snd_assert(entry != NULL, return -ENXIO); root = entry->parent == NULL ? snd_proc_root : entry->parent->p; - down(&info_mutex); + mutex_lock(&info_mutex); p = snd_create_proc_entry(entry->name, entry->mode, root); if (!p) { - up(&info_mutex); + mutex_unlock(&info_mutex); return -ENOMEM; } p->owner = entry->module; @@ -908,7 +909,7 @@ int snd_info_register(struct snd_info_entry * entry) p->size = entry->size; p->data = entry; entry->p = p; - up(&info_mutex); + mutex_unlock(&info_mutex); return 0; } @@ -929,9 +930,9 @@ int snd_info_unregister(struct snd_info_entry * entry) snd_assert(entry->p != NULL, return -ENXIO); root = entry->parent == NULL ? snd_proc_root : entry->parent->p; snd_assert(root, return -ENXIO); - down(&info_mutex); + mutex_lock(&info_mutex); snd_remove_proc_entry(root, entry->p); - up(&info_mutex); + mutex_unlock(&info_mutex); snd_info_free_entry(entry); return 0; } diff --git a/sound/core/info_oss.c b/sound/core/info_oss.c index 820f4772e44aee..f9ce854b3d114b 100644 --- a/sound/core/info_oss.c +++ b/sound/core/info_oss.c @@ -28,6 +28,7 @@ #include #include #include +#include #if defined(CONFIG_SND_OSSEMUL) && defined(CONFIG_PROC_FS) @@ -35,7 +36,7 @@ * OSS compatible part */ -static DECLARE_MUTEX(strings); +static DEFINE_MUTEX(strings); static char *snd_sndstat_strings[SNDRV_CARDS][SNDRV_OSS_INFO_DEV_COUNT]; static struct snd_info_entry *snd_sndstat_proc_entry; @@ -45,7 +46,7 @@ int snd_oss_info_register(int dev, int num, char *string) snd_assert(dev >= 0 && dev < SNDRV_OSS_INFO_DEV_COUNT, return -ENXIO); snd_assert(num >= 0 && num < SNDRV_CARDS, return -ENXIO); - down(&strings); + mutex_lock(&strings); if (string == NULL) { if ((x = snd_sndstat_strings[num][dev]) != NULL) { kfree(x); @@ -54,12 +55,12 @@ int snd_oss_info_register(int dev, int num, char *string) } else { x = kstrdup(string, GFP_KERNEL); if (x == NULL) { - up(&strings); + mutex_unlock(&strings); return -ENOMEM; } } snd_sndstat_strings[num][dev] = x; - up(&strings); + mutex_unlock(&strings); return 0; } @@ -71,7 +72,7 @@ static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int d char *str; snd_iprintf(buf, "\n%s:", id); - down(&strings); + mutex_lock(&strings); for (idx = 0; idx < SNDRV_CARDS; idx++) { str = snd_sndstat_strings[idx][dev]; if (str) { @@ -82,7 +83,7 @@ static int snd_sndstat_show_strings(struct snd_info_buffer *buf, char *id, int d snd_iprintf(buf, "%i: %s\n", idx, str); } } - up(&strings); + mutex_unlock(&strings); if (ok < 0) snd_iprintf(buf, " NOT ENABLED IN CONFIG\n"); return ok; diff --git a/sound/core/init.c b/sound/core/init.c index 75816688607c7e..17fbd6c14fb879 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -145,7 +145,7 @@ struct snd_card *snd_card_new(int idx, const char *xid, init_waitqueue_head(&card->shutdown_sleep); INIT_WORK(&card->free_workq, snd_card_free_thread, card); #ifdef CONFIG_PM - init_MUTEX(&card->power_lock); + mutex_init(&card->power_lock); init_waitqueue_head(&card->power_sleep); #endif /* the control interface cannot be accessed from the user space until */ diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 19b3dcbb09c23b..9f2b88cb5877fc 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -31,7 +31,7 @@ #include #include #include -#include +#include #include #ifdef CONFIG_SBUS #include @@ -54,7 +54,7 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab); /* */ -static DECLARE_MUTEX(list_mutex); +static DEFINE_MUTEX(list_mutex); static LIST_HEAD(mem_list_head); /* buffer preservation list */ @@ -440,7 +440,7 @@ size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id) snd_assert(dmab, return 0); - down(&list_mutex); + mutex_lock(&list_mutex); list_for_each(p, &mem_list_head) { mem = list_entry(p, struct snd_mem_list, list); if (mem->id == id && @@ -452,11 +452,11 @@ size_t snd_dma_get_reserved_buf(struct snd_dma_buffer *dmab, unsigned int id) if (dmab->dev.dev == NULL) dmab->dev.dev = dev; kfree(mem); - up(&list_mutex); + mutex_unlock(&list_mutex); return dmab->bytes; } } - up(&list_mutex); + mutex_unlock(&list_mutex); return 0; } @@ -477,11 +477,11 @@ int snd_dma_reserve_buf(struct snd_dma_buffer *dmab, unsigned int id) mem = kmalloc(sizeof(*mem), GFP_KERNEL); if (! mem) return -ENOMEM; - down(&list_mutex); + mutex_lock(&list_mutex); mem->buffer = *dmab; mem->id = id; list_add_tail(&mem->list, &mem_list_head); - up(&list_mutex); + mutex_unlock(&list_mutex); return 0; } @@ -493,7 +493,7 @@ static void free_all_reserved_pages(void) struct list_head *p; struct snd_mem_list *mem; - down(&list_mutex); + mutex_lock(&list_mutex); while (! list_empty(&mem_list_head)) { p = mem_list_head.next; mem = list_entry(p, struct snd_mem_list, list); @@ -501,7 +501,7 @@ static void free_all_reserved_pages(void) snd_dma_free_pages(&mem->buffer); kfree(mem); } - up(&list_mutex); + mutex_unlock(&list_mutex); } @@ -522,7 +522,7 @@ static int snd_mem_proc_read(char *page, char **start, off_t off, int devno; static char *types[] = { "UNKNOWN", "CONT", "DEV", "DEV-SG", "SBUS" }; - down(&list_mutex); + mutex_lock(&list_mutex); len += snprintf(page + len, count - len, "pages : %li bytes (%li pages per %likB)\n", pages * PAGE_SIZE, pages, PAGE_SIZE / 1024); @@ -537,7 +537,7 @@ static int snd_mem_proc_read(char *page, char **start, off_t off, " addr = 0x%lx, size = %d bytes\n", (unsigned long)mem->buffer.addr, (int)mem->buffer.bytes); } - up(&list_mutex); + mutex_unlock(&list_mutex); return len; } diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index f08e65a2bffe7d..9c68bc3f97aabc 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -1095,7 +1095,7 @@ static void snd_mixer_oss_proc_read(struct snd_info_entry *entry, struct snd_mixer_oss *mixer = entry->private_data; int i; - down(&mixer->reg_mutex); + mutex_lock(&mixer->reg_mutex); for (i = 0; i < SNDRV_OSS_MAX_MIXERS; i++) { struct slot *p; @@ -1110,7 +1110,7 @@ static void snd_mixer_oss_proc_read(struct snd_info_entry *entry, else snd_iprintf(buffer, "\"\" 0\n"); } - up(&mixer->reg_mutex); + mutex_unlock(&mixer->reg_mutex); } static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, @@ -1134,9 +1134,9 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, cptr = snd_info_get_str(str, cptr, sizeof(str)); if (! *str) { /* remove the entry */ - down(&mixer->reg_mutex); + mutex_lock(&mixer->reg_mutex); mixer_slot_clear(&mixer->slots[ch]); - up(&mixer->reg_mutex); + mutex_unlock(&mixer->reg_mutex); continue; } snd_info_get_str(idxstr, cptr, sizeof(idxstr)); @@ -1145,7 +1145,7 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, snd_printk(KERN_ERR "mixer_oss: invalid index %d\n", idx); continue; } - down(&mixer->reg_mutex); + mutex_lock(&mixer->reg_mutex); slot = (struct slot *)mixer->slots[ch].private_data; if (slot && slot->assigned && slot->assigned->index == idx && ! strcmp(slot->assigned->name, str)) @@ -1168,7 +1168,7 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry, kfree(tbl); } __unlock: - up(&mixer->reg_mutex); + mutex_unlock(&mixer->reg_mutex); } } @@ -1288,7 +1288,7 @@ static int snd_mixer_oss_notify_handler(struct snd_card *card, int cmd) mixer = kcalloc(2, sizeof(*mixer), GFP_KERNEL); if (mixer == NULL) return -ENOMEM; - init_MUTEX(&mixer->reg_mutex); + mutex_init(&mixer->reg_mutex); sprintf(name, "mixer%i%i", card->number, 0); if ((err = snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_MIXER, card, 0, diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index bc24d028f5181e..f8302b703a30e8 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1643,10 +1643,10 @@ static struct snd_pcm_oss_setup *snd_pcm_oss_look_for_setup(struct snd_pcm *pcm, const char *ptr, *ptrl; struct snd_pcm_oss_setup *setup; - down(&pcm->streams[stream].oss.setup_mutex); + mutex_lock(&pcm->streams[stream].oss.setup_mutex); for (setup = pcm->streams[stream].oss.setup_list; setup; setup = setup->next) { if (!strcmp(setup->task_name, task_name)) { - up(&pcm->streams[stream].oss.setup_mutex); + mutex_unlock(&pcm->streams[stream].oss.setup_mutex); return setup; } } @@ -1662,12 +1662,12 @@ static struct snd_pcm_oss_setup *snd_pcm_oss_look_for_setup(struct snd_pcm *pcm, } for (setup = pcm->streams[stream].oss.setup_list; setup; setup = setup->next) { if (!strcmp(setup->task_name, ptrl)) { - up(&pcm->streams[stream].oss.setup_mutex); + mutex_unlock(&pcm->streams[stream].oss.setup_mutex); return setup; } } __not_found: - up(&pcm->streams[stream].oss.setup_mutex); + mutex_unlock(&pcm->streams[stream].oss.setup_mutex); return NULL; } @@ -1895,7 +1895,7 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) init_waitqueue_entry(&wait, current); add_wait_queue(&pcm->open_wait, &wait); - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); while (1) { err = snd_pcm_oss_open_file(file, pcm, &pcm_oss_file, iminor(inode), psetup, csetup); @@ -1909,16 +1909,16 @@ static int snd_pcm_oss_open(struct inode *inode, struct file *file) } else break; set_current_state(TASK_INTERRUPTIBLE); - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); schedule(); - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); if (signal_pending(current)) { err = -ERESTARTSYS; break; } } remove_wait_queue(&pcm->open_wait, &wait); - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); if (err < 0) goto __error; return err; @@ -1944,9 +1944,9 @@ static int snd_pcm_oss_release(struct inode *inode, struct file *file) snd_assert(substream != NULL, return -ENXIO); pcm = substream->pcm; snd_pcm_oss_sync(pcm_oss_file); - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); snd_pcm_oss_release_file(pcm_oss_file); - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); wake_up(&pcm->open_wait); module_put(pcm->card->module); snd_card_file_remove(pcm->card, file); @@ -2293,7 +2293,7 @@ static void snd_pcm_oss_proc_read(struct snd_info_entry *entry, { struct snd_pcm_str *pstr = entry->private_data; struct snd_pcm_oss_setup *setup = pstr->oss.setup_list; - down(&pstr->oss.setup_mutex); + mutex_lock(&pstr->oss.setup_mutex); while (setup) { snd_iprintf(buffer, "%s %u %u%s%s%s%s%s%s\n", setup->task_name, @@ -2307,7 +2307,7 @@ static void snd_pcm_oss_proc_read(struct snd_info_entry *entry, setup->nosilence ? " no-silence" : ""); setup = setup->next; } - up(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); } static void snd_pcm_oss_proc_free_setup_list(struct snd_pcm_str * pstr) @@ -2337,12 +2337,12 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, struct snd_pcm_oss_setup *setup, *setup1, template; while (!snd_info_get_line(buffer, line, sizeof(line))) { - down(&pstr->oss.setup_mutex); + mutex_lock(&pstr->oss.setup_mutex); memset(&template, 0, sizeof(template)); ptr = snd_info_get_str(task_name, line, sizeof(task_name)); if (!strcmp(task_name, "clear") || !strcmp(task_name, "erase")) { snd_pcm_oss_proc_free_setup_list(pstr); - up(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); continue; } for (setup = pstr->oss.setup_list; setup; setup = setup->next) { @@ -2394,7 +2394,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, } if (setup) *setup = template; - up(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); } } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index d92c3ce4a4c9ca..f903d1bd74d5f7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -35,7 +36,7 @@ MODULE_LICENSE("GPL"); static LIST_HEAD(snd_pcm_devices); static LIST_HEAD(snd_pcm_notify_list); -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); static int snd_pcm_free(struct snd_pcm *pcm); static int snd_pcm_dev_free(struct snd_device *device); @@ -67,7 +68,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)arg)) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_PCM_DEVICES) { if (snd_pcm_search(card, device)) @@ -76,7 +77,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, } if (device == SNDRV_PCM_DEVICES) device = -1; - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (put_user(device, (int __user *)arg)) return -EFAULT; return 0; @@ -100,7 +101,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, return -EINVAL; if (get_user(subdevice, &info->subdevice)) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); pcm = snd_pcm_search(card, device); if (pcm == NULL) { err = -ENXIO; @@ -125,7 +126,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, } err = snd_pcm_info_user(substream, info); _error: - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } case SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE: @@ -262,6 +263,7 @@ static const char *snd_pcm_state_name(snd_pcm_state_t state) #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) #include + static const char *snd_pcm_oss_format_name(int format) { switch (format) { @@ -624,7 +626,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) struct snd_pcm_substream *substream, *prev; #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) - init_MUTEX(&pstr->oss.setup_mutex); + mutex_init(&pstr->oss.setup_mutex); #endif pstr->stream = stream; pstr->pcm = pcm; @@ -718,7 +720,7 @@ int snd_pcm_new(struct snd_card *card, char *id, int device, snd_pcm_free(pcm); return err; } - init_MUTEX(&pcm->open_mutex); + mutex_init(&pcm->open_mutex); init_waitqueue_head(&pcm->open_wait); if ((err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops)) < 0) { snd_pcm_free(pcm); @@ -904,9 +906,9 @@ static int snd_pcm_dev_register(struct snd_device *device) struct snd_pcm *pcm = device->device_data; snd_assert(pcm != NULL && device != NULL, return -ENXIO); - down(®ister_mutex); + mutex_lock(®ister_mutex); if (snd_pcm_search(pcm->card, pcm->device)) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EBUSY; } list_add_tail(&pcm->list, &snd_pcm_devices); @@ -930,7 +932,7 @@ static int snd_pcm_dev_register(struct snd_device *device) pcm, str)) < 0) { list_del(&pcm->list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) @@ -941,7 +943,7 @@ static int snd_pcm_dev_register(struct snd_device *device) notify = list_entry(list, struct snd_pcm_notify, list); notify->n_register(pcm); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -952,7 +954,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) struct snd_pcm_substream *substream; int cidx; - down(®ister_mutex); + mutex_lock(®ister_mutex); list_del_init(&pcm->list); for (cidx = 0; cidx < 2; cidx++) for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) @@ -963,7 +965,7 @@ static int snd_pcm_dev_disconnect(struct snd_device *device) notify = list_entry(list, struct snd_pcm_notify, list); notify->n_disconnect(pcm); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -975,7 +977,7 @@ static int snd_pcm_dev_unregister(struct snd_device *device) struct snd_pcm *pcm = device->device_data; snd_assert(pcm != NULL, return -ENXIO); - down(®ister_mutex); + mutex_lock(®ister_mutex); list_del(&pcm->list); for (cidx = 0; cidx < 2; cidx++) { devtype = -1; @@ -996,7 +998,7 @@ static int snd_pcm_dev_unregister(struct snd_device *device) notify = list_entry(list, struct snd_pcm_notify, list); notify->n_unregister(pcm); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); return snd_pcm_free(pcm); } @@ -1005,7 +1007,7 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) struct list_head *p; snd_assert(notify != NULL && notify->n_register != NULL && notify->n_unregister != NULL, return -EINVAL); - down(®ister_mutex); + mutex_lock(®ister_mutex); if (nfree) { list_del(¬ify->list); list_for_each(p, &snd_pcm_devices) @@ -1016,7 +1018,7 @@ int snd_pcm_notify(struct snd_pcm_notify *notify, int nfree) list_for_each(p, &snd_pcm_devices) notify->n_register(list_entry(p, struct snd_pcm, list)); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -1031,7 +1033,7 @@ static void snd_pcm_proc_read(struct snd_info_entry *entry, struct list_head *p; struct snd_pcm *pcm; - down(®ister_mutex); + mutex_lock(®ister_mutex); list_for_each(p, &snd_pcm_devices) { pcm = list_entry(p, struct snd_pcm, list); snd_iprintf(buffer, "%02i-%02i: %s : %s", @@ -1044,7 +1046,7 @@ static void snd_pcm_proc_read(struct snd_info_entry *entry, pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream_count); snd_iprintf(buffer, "\n"); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); } static struct snd_info_entry *snd_pcm_proc_entry = NULL; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index f3d5de7b55acd3..ce1956a5528d53 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2112,7 +2112,7 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) } init_waitqueue_entry(&wait, current); add_wait_queue(&pcm->open_wait, &wait); - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); while (1) { err = snd_pcm_open_file(file, pcm, stream, &pcm_file); if (err >= 0) @@ -2125,16 +2125,16 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) } else break; set_current_state(TASK_INTERRUPTIBLE); - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); schedule(); - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); if (signal_pending(current)) { err = -ERESTARTSYS; break; } } remove_wait_queue(&pcm->open_wait, &wait); - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); if (err < 0) goto __error; return err; @@ -2160,9 +2160,9 @@ static int snd_pcm_release(struct inode *inode, struct file *file) pcm = substream->pcm; snd_pcm_drop(substream); fasync_helper(-1, file, 0, &substream->runtime->fasync); - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); snd_pcm_release_file(pcm_file); - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); wake_up(&pcm->open_wait); module_put(pcm->card->module); snd_card_file_remove(pcm->card, file); diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index d4d124e2192483..6b7a3677429891 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -57,7 +58,7 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device); static int snd_rawmidi_dev_unregister(struct snd_device *device); static LIST_HEAD(snd_rawmidi_devices); -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); static struct snd_rawmidi *snd_rawmidi_search(struct snd_card *card, int device) { @@ -237,9 +238,9 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, if (rfile) rfile->input = rfile->output = NULL; - down(®ister_mutex); + mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, device); - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (rmidi == NULL) { err = -ENODEV; goto __error1; @@ -249,7 +250,7 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, goto __error1; } if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - down(&rmidi->open_mutex); + mutex_lock(&rmidi->open_mutex); if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT)) { err = -ENXIO; @@ -359,7 +360,7 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, soutput = NULL; } if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - up(&rmidi->open_mutex); + mutex_unlock(&rmidi->open_mutex); if (rfile) { rfile->rmidi = rmidi; rfile->input = sinput; @@ -374,7 +375,7 @@ int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, snd_rawmidi_runtime_free(soutput); module_put(rmidi->card->module); if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - up(&rmidi->open_mutex); + mutex_unlock(&rmidi->open_mutex); __error1: return err; } @@ -422,7 +423,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) } init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); - down(&rmidi->open_mutex); + mutex_lock(&rmidi->open_mutex); while (1) { subdevice = -1; down_read(&card->controls_rwsem); @@ -446,9 +447,9 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) } else break; set_current_state(TASK_INTERRUPTIBLE); - up(&rmidi->open_mutex); + mutex_unlock(&rmidi->open_mutex); schedule(); - down(&rmidi->open_mutex); + mutex_lock(&rmidi->open_mutex); if (signal_pending(current)) { err = -ERESTARTSYS; break; @@ -467,7 +468,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) snd_card_file_remove(card, file); kfree(rawmidi_file); } - up(&rmidi->open_mutex); + mutex_unlock(&rmidi->open_mutex); return err; } @@ -480,7 +481,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile) snd_assert(rfile != NULL, return -ENXIO); snd_assert(rfile->input != NULL || rfile->output != NULL, return -ENXIO); rmidi = rfile->rmidi; - down(&rmidi->open_mutex); + mutex_lock(&rmidi->open_mutex); if (rfile->input != NULL) { substream = rfile->input; rfile->input = NULL; @@ -514,7 +515,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile) } rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened--; } - up(&rmidi->open_mutex); + mutex_unlock(&rmidi->open_mutex); module_put(rmidi->card->module); return 0; } @@ -576,9 +577,9 @@ int snd_rawmidi_info_select(struct snd_card *card, struct snd_rawmidi_info *info struct snd_rawmidi_substream *substream; struct list_head *list; - down(®ister_mutex); + mutex_lock(®ister_mutex); rmidi = snd_rawmidi_search(card, info->device); - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (!rmidi) return -ENXIO; if (info->stream < 0 || info->stream > 1) @@ -818,7 +819,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_RAWMIDI_DEVICES) { if (snd_rawmidi_search(card, device)) @@ -827,7 +828,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, } if (device == SNDRV_RAWMIDI_DEVICES) device = -1; - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (put_user(device, (int __user *)argp)) return -EFAULT; return 0; @@ -1314,7 +1315,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); - down(&rmidi->open_mutex); + mutex_lock(&rmidi->open_mutex); if (rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT) { list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { substream = list_entry(list, struct snd_rawmidi_substream, list); @@ -1355,7 +1356,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, } } } - up(&rmidi->open_mutex); + mutex_unlock(&rmidi->open_mutex); } /* @@ -1436,7 +1437,7 @@ int snd_rawmidi_new(struct snd_card *card, char *id, int device, } rmidi->card = card; rmidi->device = device; - init_MUTEX(&rmidi->open_mutex); + mutex_init(&rmidi->open_mutex); init_waitqueue_head(&rmidi->open_wait); if (id != NULL) strlcpy(rmidi->id, id, sizeof(rmidi->id)); @@ -1507,9 +1508,9 @@ static int snd_rawmidi_dev_register(struct snd_device *device) if (rmidi->device >= SNDRV_RAWMIDI_DEVICES) return -ENOMEM; - down(®ister_mutex); + mutex_lock(®ister_mutex); if (snd_rawmidi_search(rmidi->card, rmidi->device)) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EBUSY; } list_add_tail(&rmidi->list, &snd_rawmidi_devices); @@ -1519,14 +1520,14 @@ static int snd_rawmidi_dev_register(struct snd_device *device) &snd_rawmidi_f_ops, rmidi, name)) < 0) { snd_printk(KERN_ERR "unable to register rawmidi device %i:%i\n", rmidi->card->number, rmidi->device); list_del(&rmidi->list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } if (rmidi->ops && rmidi->ops->dev_register && (err = rmidi->ops->dev_register(rmidi)) < 0) { snd_unregister_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device); list_del(&rmidi->list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } #ifdef CONFIG_SND_OSSEMUL @@ -1553,7 +1554,7 @@ static int snd_rawmidi_dev_register(struct snd_device *device) } } #endif /* CONFIG_SND_OSSEMUL */ - up(®ister_mutex); + mutex_unlock(®ister_mutex); sprintf(name, "midi%d", rmidi->device); entry = snd_info_create_card_entry(rmidi->card, name, rmidi->card->proc_root); if (entry) { @@ -1583,9 +1584,9 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device) { struct snd_rawmidi *rmidi = device->device_data; - down(®ister_mutex); + mutex_lock(®ister_mutex); list_del_init(&rmidi->list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -1594,7 +1595,7 @@ static int snd_rawmidi_dev_unregister(struct snd_device *device) struct snd_rawmidi *rmidi = device->device_data; snd_assert(rmidi != NULL, return -ENXIO); - down(®ister_mutex); + mutex_lock(®ister_mutex); list_del(&rmidi->list); if (rmidi->proc_entry) { snd_info_unregister(rmidi->proc_entry); @@ -1616,7 +1617,7 @@ static int snd_rawmidi_dev_unregister(struct snd_device *device) if (rmidi->ops && rmidi->ops->dev_unregister) rmidi->ops->dev_unregister(rmidi); snd_unregister_device(SNDRV_DEVICE_TYPE_RAWMIDI, rmidi->card, rmidi->device); - up(®ister_mutex); + mutex_unlock(®ister_mutex); #if defined(CONFIG_SND_SEQUENCER) || (defined(MODULE) && defined(CONFIG_SND_SEQUENCER_MODULE)) if (rmidi->seq_dev) { snd_device_free(rmidi->card, rmidi->seq_dev); diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index c98f0ba13810d7..b9919785180b82 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include @@ -124,7 +125,7 @@ module_exit(alsa_seq_oss_exit) * ALSA minor device interface */ -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); static int odev_open(struct inode *inode, struct file *file) @@ -136,9 +137,9 @@ odev_open(struct inode *inode, struct file *file) else level = SNDRV_SEQ_OSS_MODE_SYNTH; - down(®ister_mutex); + mutex_lock(®ister_mutex); rc = snd_seq_oss_open(file, level); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return rc; } @@ -153,9 +154,9 @@ odev_release(struct inode *inode, struct file *file) snd_seq_oss_drain_write(dp); - down(®ister_mutex); + mutex_lock(®ister_mutex); snd_seq_oss_release(dp); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -224,13 +225,13 @@ register_device(void) { int rc; - down(®ister_mutex); + mutex_lock(®ister_mutex); if ((rc = snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_SEQUENCER, NULL, 0, &seq_oss_f_ops, NULL, SNDRV_SEQ_OSS_DEVNAME)) < 0) { snd_printk(KERN_ERR "can't register device seq\n"); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return rc; } if ((rc = snd_register_oss_device(SNDRV_OSS_DEVICE_TYPE_MUSIC, @@ -239,24 +240,24 @@ register_device(void) SNDRV_SEQ_OSS_DEVNAME)) < 0) { snd_printk(KERN_ERR "can't register device music\n"); snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_SEQUENCER, NULL, 0); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return rc; } debug_printk(("device registered\n")); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } static void unregister_device(void) { - down(®ister_mutex); + mutex_lock(®ister_mutex); debug_printk(("device unregistered\n")); if (snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_MUSIC, NULL, 0) < 0) snd_printk(KERN_ERR "error unregister device music\n"); if (snd_unregister_oss_device(SNDRV_OSS_DEVICE_TYPE_SEQUENCER, NULL, 0) < 0) snd_printk(KERN_ERR "error unregister device seq\n"); - up(®ister_mutex); + mutex_unlock(®ister_mutex); } /* @@ -270,12 +271,12 @@ static struct snd_info_entry *info_entry; static void info_read(struct snd_info_entry *entry, struct snd_info_buffer *buf) { - down(®ister_mutex); + mutex_lock(®ister_mutex); snd_iprintf(buf, "OSS sequencer emulation version %s\n", SNDRV_SEQ_OSS_VERSION_STR); snd_seq_oss_system_info_read(buf); snd_seq_oss_synth_info_read(buf); snd_seq_oss_midi_info_read(buf); - up(®ister_mutex); + mutex_unlock(®ister_mutex); } diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index fd2032eae21417..aae6420f594870 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -67,7 +67,7 @@ #define SNDRV_SEQ_LFLG_OPEN (SNDRV_SEQ_LFLG_INPUT|SNDRV_SEQ_LFLG_OUTPUT) static DEFINE_SPINLOCK(clients_lock); -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); /* * client table @@ -237,7 +237,7 @@ static struct snd_seq_client *seq_create_client1(int client_index, int poolsize) client->type = NO_CLIENT; snd_use_lock_init(&client->use_lock); rwlock_init(&client->ports_lock); - init_MUTEX(&client->ports_mutex); + mutex_init(&client->ports_mutex); INIT_LIST_HEAD(&client->ports_list_head); /* find free slot in the client table */ @@ -290,7 +290,7 @@ static int seq_free_client1(struct snd_seq_client *client) static void seq_free_client(struct snd_seq_client * client) { - down(®ister_mutex); + mutex_lock(®ister_mutex); switch (client->type) { case NO_CLIENT: snd_printk(KERN_WARNING "Seq: Trying to free unused client %d\n", @@ -306,7 +306,7 @@ static void seq_free_client(struct snd_seq_client * client) snd_printk(KERN_ERR "Seq: Trying to free client %d with undefined type = %d\n", client->number, client->type); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); snd_seq_system_client_ev_client_exit(client->number); } @@ -322,11 +322,11 @@ static int snd_seq_open(struct inode *inode, struct file *file) struct snd_seq_client *client; struct snd_seq_user_client *user; - if (down_interruptible(®ister_mutex)) + if (mutex_lock_interruptible(®ister_mutex)) return -ERESTARTSYS; client = seq_create_client1(-1, SNDRV_SEQ_DEFAULT_EVENTS); if (client == NULL) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENOMEM; /* failure code */ } @@ -346,14 +346,14 @@ static int snd_seq_open(struct inode *inode, struct file *file) if (user->fifo == NULL) { seq_free_client1(client); kfree(client); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENOMEM; } } usage_alloc(&client_usage, 1); client->type = USER_CLIENT; - up(®ister_mutex); + mutex_unlock(®ister_mutex); c = client->number; file->private_data = client; @@ -1743,7 +1743,7 @@ static int snd_seq_ioctl_get_queue_timer(struct snd_seq_client *client, if (queue == NULL) return -EINVAL; - if (down_interruptible(&queue->timer_mutex)) { + if (mutex_lock_interruptible(&queue->timer_mutex)) { queuefree(queue); return -ERESTARTSYS; } @@ -1756,7 +1756,7 @@ static int snd_seq_ioctl_get_queue_timer(struct snd_seq_client *client, timer.u.alsa.id = tmr->alsa_id; timer.u.alsa.resolution = tmr->preferred_resolution; } - up(&queue->timer_mutex); + mutex_unlock(&queue->timer_mutex); queuefree(queue); if (copy_to_user(arg, &timer, sizeof(timer))) @@ -1785,7 +1785,7 @@ static int snd_seq_ioctl_set_queue_timer(struct snd_seq_client *client, q = queueptr(timer.queue); if (q == NULL) return -ENXIO; - if (down_interruptible(&q->timer_mutex)) { + if (mutex_lock_interruptible(&q->timer_mutex)) { queuefree(q); return -ERESTARTSYS; } @@ -1797,7 +1797,7 @@ static int snd_seq_ioctl_set_queue_timer(struct snd_seq_client *client, tmr->preferred_resolution = timer.u.alsa.resolution; } result = snd_seq_queue_timer_open(timer.queue); - up(&q->timer_mutex); + mutex_unlock(&q->timer_mutex); queuefree(q); } else { return -EPERM; @@ -2230,7 +2230,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, if (card == NULL && client_index >= SNDRV_SEQ_GLOBAL_CLIENTS) return -EINVAL; - if (down_interruptible(®ister_mutex)) + if (mutex_lock_interruptible(®ister_mutex)) return -ERESTARTSYS; if (card) { @@ -2243,7 +2243,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, /* empty write queue as default */ client = seq_create_client1(client_index, 0); if (client == NULL) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EBUSY; /* failure code */ } usage_alloc(&client_usage, 1); @@ -2256,7 +2256,7 @@ int snd_seq_create_kernel_client(struct snd_card *card, int client_index, va_end(args); client->type = KERNEL_CLIENT; - up(®ister_mutex); + mutex_unlock(®ister_mutex); /* make others aware this new client */ snd_seq_system_client_ev_client_start(client->number); @@ -2464,7 +2464,7 @@ static void snd_seq_info_dump_ports(struct snd_info_buffer *buffer, { struct list_head *l; - down(&client->ports_mutex); + mutex_lock(&client->ports_mutex); list_for_each(l, &client->ports_list_head) { struct snd_seq_client_port *p = list_entry(l, struct snd_seq_client_port, list); snd_iprintf(buffer, " Port %3d : \"%s\" (%c%c%c%c)\n", @@ -2476,7 +2476,7 @@ static void snd_seq_info_dump_ports(struct snd_info_buffer *buffer, snd_seq_info_dump_subscribers(buffer, &p->c_src, 1, " Connecting To: "); snd_seq_info_dump_subscribers(buffer, &p->c_dest, 0, " Connected From: "); } - up(&client->ports_mutex); + mutex_unlock(&client->ports_mutex); } @@ -2550,16 +2550,16 @@ int __init snd_sequencer_device_init(void) { int err; - if (down_interruptible(®ister_mutex)) + if (mutex_lock_interruptible(®ister_mutex)) return -ERESTARTSYS; if ((err = snd_register_device(SNDRV_DEVICE_TYPE_SEQUENCER, NULL, 0, &snd_seq_f_ops, NULL, "seq")) < 0) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } diff --git a/sound/core/seq/seq_clientmgr.h b/sound/core/seq/seq_clientmgr.h index 450091ca153d27..5e04e20e239f4f 100644 --- a/sound/core/seq/seq_clientmgr.h +++ b/sound/core/seq/seq_clientmgr.h @@ -58,7 +58,7 @@ struct snd_seq_client { int num_ports; /* number of ports */ struct list_head ports_list_head; rwlock_t ports_lock; - struct semaphore ports_mutex; + struct mutex ports_mutex; int convert32; /* convert 32->64bit */ /* output pool */ diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index 9ece443fba5599..d9a3e5a18d6ad0 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -45,6 +45,7 @@ #include #include #include +#include MODULE_AUTHOR("Takashi Iwai "); MODULE_DESCRIPTION("ALSA sequencer device management"); @@ -69,7 +70,7 @@ struct ops_list { struct list_head dev_list; /* list of devices */ int num_devices; /* number of associated devices */ int num_init_devices; /* number of initialized devices */ - struct semaphore reg_mutex; + struct mutex reg_mutex; struct list_head list; /* next driver */ }; @@ -77,7 +78,7 @@ struct ops_list { static LIST_HEAD(opslist); static int num_ops; -static DECLARE_MUTEX(ops_mutex); +static DEFINE_MUTEX(ops_mutex); #ifdef CONFIG_PROC_FS static struct snd_info_entry *info_entry = NULL; #endif @@ -108,7 +109,7 @@ static void snd_seq_device_info(struct snd_info_entry *entry, { struct list_head *head; - down(&ops_mutex); + mutex_lock(&ops_mutex); list_for_each(head, &opslist) { struct ops_list *ops = list_entry(head, struct ops_list, list); snd_iprintf(buffer, "snd-%s%s%s%s,%d\n", @@ -118,7 +119,7 @@ static void snd_seq_device_info(struct snd_info_entry *entry, ops->driver & DRIVER_LOCKED ? ",locked" : "", ops->num_devices); } - up(&ops_mutex); + mutex_unlock(&ops_mutex); } #endif @@ -154,20 +155,20 @@ void snd_seq_device_load_drivers(void) if (! current->fs->root) return; - down(&ops_mutex); + mutex_lock(&ops_mutex); list_for_each(head, &opslist) { struct ops_list *ops = list_entry(head, struct ops_list, list); if (! (ops->driver & DRIVER_LOADED) && ! (ops->driver & DRIVER_REQUESTED)) { ops->used++; - up(&ops_mutex); + mutex_unlock(&ops_mutex); ops->driver |= DRIVER_REQUESTED; request_module("snd-%s", ops->id); - down(&ops_mutex); + mutex_lock(&ops_mutex); ops->used--; } } - up(&ops_mutex); + mutex_unlock(&ops_mutex); #endif } @@ -214,10 +215,10 @@ int snd_seq_device_new(struct snd_card *card, int device, char *id, int argsize, dev->status = SNDRV_SEQ_DEVICE_FREE; /* add this device to the list */ - down(&ops->reg_mutex); + mutex_lock(&ops->reg_mutex); list_add_tail(&dev->list, &ops->dev_list); ops->num_devices++; - up(&ops->reg_mutex); + mutex_unlock(&ops->reg_mutex); unlock_driver(ops); @@ -246,10 +247,10 @@ static int snd_seq_device_free(struct snd_seq_device *dev) return -ENXIO; /* remove the device from the list */ - down(&ops->reg_mutex); + mutex_lock(&ops->reg_mutex); list_del(&dev->list); ops->num_devices--; - up(&ops->reg_mutex); + mutex_unlock(&ops->reg_mutex); free_device(dev, ops); if (dev->private_free) @@ -344,7 +345,7 @@ int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, return -EBUSY; } - down(&ops->reg_mutex); + mutex_lock(&ops->reg_mutex); /* copy driver operators */ ops->ops = *entry; ops->driver |= DRIVER_LOADED; @@ -355,7 +356,7 @@ int snd_seq_device_register_driver(char *id, struct snd_seq_dev_ops *entry, struct snd_seq_device *dev = list_entry(head, struct snd_seq_device, list); init_device(dev, ops); } - up(&ops->reg_mutex); + mutex_unlock(&ops->reg_mutex); unlock_driver(ops); snd_seq_autoload_unlock(); @@ -378,17 +379,17 @@ static struct ops_list * create_driver(char *id) /* set up driver entry */ strlcpy(ops->id, id, sizeof(ops->id)); - init_MUTEX(&ops->reg_mutex); + mutex_init(&ops->reg_mutex); ops->driver = DRIVER_EMPTY; INIT_LIST_HEAD(&ops->dev_list); /* lock this instance */ ops->used = 1; /* register driver entry */ - down(&ops_mutex); + mutex_lock(&ops_mutex); list_add_tail(&ops->list, &opslist); num_ops++; - up(&ops_mutex); + mutex_unlock(&ops_mutex); return ops; } @@ -414,7 +415,7 @@ int snd_seq_device_unregister_driver(char *id) } /* close and release all devices associated with this driver */ - down(&ops->reg_mutex); + mutex_lock(&ops->reg_mutex); ops->driver |= DRIVER_LOCKED; /* do not remove this driver recursively */ list_for_each(head, &ops->dev_list) { struct snd_seq_device *dev = list_entry(head, struct snd_seq_device, list); @@ -425,7 +426,7 @@ int snd_seq_device_unregister_driver(char *id) if (ops->num_init_devices > 0) snd_printk(KERN_ERR "free_driver: init_devices > 0!! (%d)\n", ops->num_init_devices); - up(&ops->reg_mutex); + mutex_unlock(&ops->reg_mutex); unlock_driver(ops); @@ -443,7 +444,7 @@ static void remove_drivers(void) { struct list_head *head; - down(&ops_mutex); + mutex_lock(&ops_mutex); head = opslist.next; while (head != &opslist) { struct ops_list *ops = list_entry(head, struct ops_list, list); @@ -456,7 +457,7 @@ static void remove_drivers(void) } else head = head->next; } - up(&ops_mutex); + mutex_unlock(&ops_mutex); } /* @@ -519,16 +520,16 @@ static struct ops_list * find_driver(char *id, int create_if_empty) { struct list_head *head; - down(&ops_mutex); + mutex_lock(&ops_mutex); list_for_each(head, &opslist) { struct ops_list *ops = list_entry(head, struct ops_list, list); if (strcmp(ops->id, id) == 0) { ops->used++; - up(&ops_mutex); + mutex_unlock(&ops_mutex); return ops; } } - up(&ops_mutex); + mutex_unlock(&ops_mutex); if (create_if_empty) return create_driver(id); return NULL; @@ -536,9 +537,9 @@ static struct ops_list * find_driver(char *id, int create_if_empty) static void unlock_driver(struct ops_list *ops) { - down(&ops_mutex); + mutex_lock(&ops_mutex); ops->used--; - up(&ops_mutex); + mutex_unlock(&ops_mutex); } diff --git a/sound/core/seq/seq_instr.c b/sound/core/seq/seq_instr.c index 4874520639650c..f30d171b6d964a 100644 --- a/sound/core/seq/seq_instr.c +++ b/sound/core/seq/seq_instr.c @@ -36,7 +36,7 @@ static void snd_instr_lock_ops(struct snd_seq_kinstr_list *list) if (!(list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT)) { spin_lock_irqsave(&list->ops_lock, list->ops_flags); } else { - down(&list->ops_mutex); + mutex_lock(&list->ops_mutex); } } @@ -45,7 +45,7 @@ static void snd_instr_unlock_ops(struct snd_seq_kinstr_list *list) if (!(list->flags & SNDRV_SEQ_INSTR_FLG_DIRECT)) { spin_unlock_irqrestore(&list->ops_lock, list->ops_flags); } else { - up(&list->ops_mutex); + mutex_unlock(&list->ops_mutex); } } @@ -82,7 +82,7 @@ struct snd_seq_kinstr_list *snd_seq_instr_list_new(void) return NULL; spin_lock_init(&list->lock); spin_lock_init(&list->ops_lock); - init_MUTEX(&list->ops_mutex); + mutex_init(&list->ops_mutex); list->owner = -1; return list; } diff --git a/sound/core/seq/seq_midi.c b/sound/core/seq/seq_midi.c index ce0df86157de68..9caa1372bece85 100644 --- a/sound/core/seq/seq_midi.c +++ b/sound/core/seq/seq_midi.c @@ -32,7 +32,7 @@ Possible options for midisynth module: #include #include #include -#include +#include #include #include #include @@ -70,7 +70,7 @@ struct seq_midisynth_client { }; static struct seq_midisynth_client *synths[SNDRV_CARDS]; -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); /* handle rawmidi input event (MIDI v1.0 stream) */ static void snd_midi_input_event(struct snd_rawmidi_substream *substream) @@ -308,13 +308,13 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) if (ports > (256 / SNDRV_RAWMIDI_DEVICES)) ports = 256 / SNDRV_RAWMIDI_DEVICES; - down(®ister_mutex); + mutex_lock(®ister_mutex); client = synths[card->number]; if (client == NULL) { newclient = 1; client = kzalloc(sizeof(*client), GFP_KERNEL); if (client == NULL) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); kfree(info); return -ENOMEM; } @@ -324,7 +324,7 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) (const char *)info->name : "External MIDI"); if (client->seq_client < 0) { kfree(client); - up(®ister_mutex); + mutex_unlock(®ister_mutex); kfree(info); return -ENOMEM; } @@ -397,7 +397,7 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) client->num_ports++; if (newclient) synths[card->number] = client; - up(®ister_mutex); + mutex_unlock(®ister_mutex); kfree(info); kfree(port); return 0; /* success */ @@ -414,7 +414,7 @@ snd_seq_midisynth_register_port(struct snd_seq_device *dev) } kfree(info); kfree(port); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENOMEM; } @@ -427,10 +427,10 @@ snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) struct snd_card *card = dev->card; int device = dev->device, p, ports; - down(®ister_mutex); + mutex_lock(®ister_mutex); client = synths[card->number]; if (client == NULL || client->ports[device] == NULL) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENODEV; } ports = client->ports_per_device[device]; @@ -446,7 +446,7 @@ snd_seq_midisynth_unregister_port(struct snd_seq_device *dev) synths[card->number] = NULL; kfree(client); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 2b384fd7967f5d..41e078c938cdea 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -159,7 +159,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, port_subs_info_init(&new_port->c_dest); num = port >= 0 ? port : 0; - down(&client->ports_mutex); + mutex_lock(&client->ports_mutex); write_lock_irqsave(&client->ports_lock, flags); list_for_each(l, &client->ports_list_head) { struct snd_seq_client_port *p = list_entry(l, struct snd_seq_client_port, list); @@ -173,7 +173,7 @@ struct snd_seq_client_port *snd_seq_create_port(struct snd_seq_client *client, client->num_ports++; new_port->addr.port = num; /* store the port number in the port */ write_unlock_irqrestore(&client->ports_lock, flags); - up(&client->ports_mutex); + mutex_unlock(&client->ports_mutex); sprintf(new_port->name, "port-%d", num); return new_port; @@ -292,7 +292,7 @@ int snd_seq_delete_port(struct snd_seq_client *client, int port) struct list_head *l; struct snd_seq_client_port *found = NULL; - down(&client->ports_mutex); + mutex_lock(&client->ports_mutex); write_lock_irqsave(&client->ports_lock, flags); list_for_each(l, &client->ports_list_head) { struct snd_seq_client_port *p = list_entry(l, struct snd_seq_client_port, list); @@ -305,7 +305,7 @@ int snd_seq_delete_port(struct snd_seq_client *client, int port) } } write_unlock_irqrestore(&client->ports_lock, flags); - up(&client->ports_mutex); + mutex_unlock(&client->ports_mutex); if (found) return port_delete(client, found); else @@ -321,7 +321,7 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client) /* move the port list to deleted_list, and * clear the port list in the client data. */ - down(&client->ports_mutex); + mutex_lock(&client->ports_mutex); write_lock_irqsave(&client->ports_lock, flags); if (! list_empty(&client->ports_list_head)) { __list_add(&deleted_list, @@ -341,7 +341,7 @@ int snd_seq_delete_all_ports(struct snd_seq_client *client) snd_seq_system_client_ev_port_exit(port->addr.client, port->addr.port); port_delete(client, port); } - up(&client->ports_mutex); + mutex_unlock(&client->ports_mutex); return 0; } diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 9cf20f0455426d..9b87bb0c7f33a3 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -119,7 +119,7 @@ static struct snd_seq_queue *queue_new(int owner, int locked) spin_lock_init(&q->owner_lock); spin_lock_init(&q->check_lock); - init_MUTEX(&q->timer_mutex); + mutex_init(&q->timer_mutex); snd_use_lock_init(&q->use_lock); q->queue = -1; @@ -516,7 +516,7 @@ int snd_seq_queue_use(int queueid, int client, int use) queue = queueptr(queueid); if (queue == NULL) return -EINVAL; - down(&queue->timer_mutex); + mutex_lock(&queue->timer_mutex); if (use) { if (!test_and_set_bit(client, queue->clients_bitmap)) queue->clients++; @@ -531,7 +531,7 @@ int snd_seq_queue_use(int queueid, int client, int use) } else { snd_seq_timer_close(queue); } - up(&queue->timer_mutex); + mutex_unlock(&queue->timer_mutex); queuefree(queue); return 0; } diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index 888438599387bf..30c8111477f61e 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -54,7 +54,7 @@ struct snd_seq_queue { /* clients which uses this queue (bitmap) */ DECLARE_BITMAP(clients_bitmap, SNDRV_SEQ_MAX_CLIENTS); unsigned int clients; /* users of this queue */ - struct semaphore timer_mutex; + struct mutex timer_mutex; snd_use_lock_t use_lock; }; diff --git a/sound/core/sound.c b/sound/core/sound.c index a8eda02bcf1c0f..df4ab94d006e0f 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -33,6 +33,7 @@ #include #include #include +#include #define SNDRV_OS_MINORS 256 @@ -61,7 +62,7 @@ MODULE_ALIAS_CHARDEV_MAJOR(CONFIG_SND_MAJOR); int snd_ecards_limit; static struct snd_minor *snd_minors[SNDRV_OS_MINORS]; -static DECLARE_MUTEX(sound_mutex); +static DEFINE_MUTEX(sound_mutex); extern struct class *sound_class; @@ -122,13 +123,13 @@ void *snd_lookup_minor_data(unsigned int minor, int type) if (minor > ARRAY_SIZE(snd_minors)) return NULL; - down(&sound_mutex); + mutex_lock(&sound_mutex); mreg = snd_minors[minor]; if (mreg && mreg->type == type) private_data = mreg->private_data; else private_data = NULL; - up(&sound_mutex); + mutex_unlock(&sound_mutex); return private_data; } @@ -256,7 +257,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, preg->f_ops = f_ops; preg->private_data = private_data; strcpy(preg->name, name); - down(&sound_mutex); + mutex_lock(&sound_mutex); #ifdef CONFIG_SND_DYNAMIC_MINORS minor = snd_find_free_minor(); #else @@ -265,7 +266,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, minor = -EBUSY; #endif if (minor < 0) { - up(&sound_mutex); + mutex_unlock(&sound_mutex); kfree(preg); return minor; } @@ -276,7 +277,7 @@ int snd_register_device(int type, struct snd_card *card, int dev, device = card->dev; class_device_create(sound_class, NULL, MKDEV(major, minor), device, "%s", name); - up(&sound_mutex); + mutex_unlock(&sound_mutex); return 0; } @@ -297,7 +298,7 @@ int snd_unregister_device(int type, struct snd_card *card, int dev) struct snd_minor *mptr; cardnum = card ? card->number : -1; - down(&sound_mutex); + mutex_lock(&sound_mutex); for (minor = 0; minor < ARRAY_SIZE(snd_minors); ++minor) if ((mptr = snd_minors[minor]) != NULL && mptr->type == type && @@ -305,7 +306,7 @@ int snd_unregister_device(int type, struct snd_card *card, int dev) mptr->device == dev) break; if (minor == ARRAY_SIZE(snd_minors)) { - up(&sound_mutex); + mutex_unlock(&sound_mutex); return -EINVAL; } @@ -315,7 +316,7 @@ int snd_unregister_device(int type, struct snd_card *card, int dev) class_device_destroy(sound_class, MKDEV(major, minor)); snd_minors[minor] = NULL; - up(&sound_mutex); + mutex_unlock(&sound_mutex); kfree(mptr); return 0; } @@ -354,7 +355,7 @@ static void snd_minor_info_read(struct snd_info_entry *entry, struct snd_info_bu int minor; struct snd_minor *mptr; - down(&sound_mutex); + mutex_lock(&sound_mutex); for (minor = 0; minor < SNDRV_OS_MINORS; ++minor) { if (!(mptr = snd_minors[minor])) continue; @@ -371,7 +372,7 @@ static void snd_minor_info_read(struct snd_info_entry *entry, struct snd_info_bu snd_iprintf(buffer, "%3i: : %s\n", minor, snd_device_type_name(mptr->type)); } - up(&sound_mutex); + mutex_unlock(&sound_mutex); } int __init snd_minor_info_init(void) diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index d0be32b517c156..6b4a4bccd8fee4 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -34,11 +34,12 @@ #include #include #include +#include #define SNDRV_OSS_MINORS 128 static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS]; -static DECLARE_MUTEX(sound_oss_mutex); +static DEFINE_MUTEX(sound_oss_mutex); void *snd_lookup_oss_minor_data(unsigned int minor, int type) { @@ -47,13 +48,13 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) if (minor > ARRAY_SIZE(snd_oss_minors)) return NULL; - down(&sound_oss_mutex); + mutex_lock(&sound_oss_mutex); mreg = snd_oss_minors[minor]; if (mreg && mreg->type == type) private_data = mreg->private_data; else private_data = NULL; - up(&sound_oss_mutex); + mutex_unlock(&sound_oss_mutex); return private_data; } @@ -117,7 +118,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, preg->device = dev; preg->f_ops = f_ops; preg->private_data = private_data; - down(&sound_oss_mutex); + mutex_lock(&sound_oss_mutex); snd_oss_minors[minor] = preg; minor_unit = SNDRV_MINOR_OSS_DEVICE(minor); switch (minor_unit) { @@ -143,7 +144,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, goto __end; snd_oss_minors[track2] = preg; } - up(&sound_oss_mutex); + mutex_unlock(&sound_oss_mutex); return 0; __end: @@ -152,7 +153,7 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, if (register1 >= 0) unregister_sound_special(register1); snd_oss_minors[minor] = NULL; - up(&sound_oss_mutex); + mutex_unlock(&sound_oss_mutex); kfree(preg); return -EBUSY; } @@ -168,10 +169,10 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) return 0; if (minor < 0) return minor; - down(&sound_oss_mutex); + mutex_lock(&sound_oss_mutex); mptr = snd_oss_minors[minor]; if (mptr == NULL) { - up(&sound_oss_mutex); + mutex_unlock(&sound_oss_mutex); return -ENOENT; } unregister_sound_special(minor); @@ -191,7 +192,7 @@ int snd_unregister_oss_device(int type, struct snd_card *card, int dev) snd_oss_minors[track2] = NULL; } snd_oss_minors[minor] = NULL; - up(&sound_oss_mutex); + mutex_unlock(&sound_oss_mutex); kfree(mptr); return 0; } @@ -229,7 +230,7 @@ static void snd_minor_info_oss_read(struct snd_info_entry *entry, int minor; struct snd_minor *mptr; - down(&sound_oss_mutex); + mutex_lock(&sound_oss_mutex); for (minor = 0; minor < SNDRV_OSS_MINORS; ++minor) { if (!(mptr = snd_oss_minors[minor])) continue; @@ -241,7 +242,7 @@ static void snd_minor_info_oss_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%3i: : %s\n", minor, snd_oss_device_type_name(mptr->type)); } - up(&sound_oss_mutex); + mutex_unlock(&sound_oss_mutex); } diff --git a/sound/core/timer.c b/sound/core/timer.c index 2425b971b240b0..cdeeb639b675a3 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -70,7 +71,7 @@ struct snd_timer_user { struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; struct fasync_struct *fasync; - struct semaphore tread_sem; + struct mutex tread_sem; }; /* list of timers */ @@ -82,7 +83,7 @@ static LIST_HEAD(snd_timer_slave_list); /* lock for slave active lists */ static DEFINE_SPINLOCK(slave_active_lock); -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); static int snd_timer_free(struct snd_timer *timer); static int snd_timer_dev_free(struct snd_device *device); @@ -252,10 +253,10 @@ int snd_timer_open(struct snd_timer_instance **ti, snd_printd("invalid slave class %i\n", tid->dev_sclass); return -EINVAL; } - down(®ister_mutex); + mutex_lock(®ister_mutex); timeri = snd_timer_instance_new(owner, NULL); if (!timeri) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENOMEM; } timeri->slave_class = tid->dev_sclass; @@ -263,37 +264,37 @@ int snd_timer_open(struct snd_timer_instance **ti, timeri->flags |= SNDRV_TIMER_IFLG_SLAVE; list_add_tail(&timeri->open_list, &snd_timer_slave_list); snd_timer_check_slave(timeri); - up(®ister_mutex); + mutex_unlock(®ister_mutex); *ti = timeri; return 0; } /* open a master instance */ - down(®ister_mutex); + mutex_lock(®ister_mutex); timer = snd_timer_find(tid); #ifdef CONFIG_KMOD if (timer == NULL) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); snd_timer_request(tid); - down(®ister_mutex); + mutex_lock(®ister_mutex); timer = snd_timer_find(tid); } #endif if (!timer) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENODEV; } if (!list_empty(&timer->open_list_head)) { timeri = list_entry(timer->open_list_head.next, struct snd_timer_instance, open_list); if (timeri->flags & SNDRV_TIMER_IFLG_EXCLUSIVE) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EBUSY; } } timeri = snd_timer_instance_new(owner, timer); if (!timeri) { - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -ENOMEM; } timeri->slave_class = tid->dev_sclass; @@ -302,7 +303,7 @@ int snd_timer_open(struct snd_timer_instance **ti, timer->hw.open(timer); list_add_tail(&timeri->open_list, &timer->open_list_head); snd_timer_check_master(timeri); - up(®ister_mutex); + mutex_unlock(®ister_mutex); *ti = timeri; return 0; } @@ -333,9 +334,9 @@ int snd_timer_close(struct snd_timer_instance *timeri) spin_lock_irq(&slave_active_lock); } spin_unlock_irq(&slave_active_lock); - down(®ister_mutex); + mutex_lock(®ister_mutex); list_del(&timeri->open_list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); } else { timer = timeri->timer; /* wait, until the active callback is finished */ @@ -346,7 +347,7 @@ int snd_timer_close(struct snd_timer_instance *timeri) spin_lock_irq(&timer->lock); } spin_unlock_irq(&timer->lock); - down(®ister_mutex); + mutex_lock(®ister_mutex); list_del(&timeri->open_list); if (timer && list_empty(&timer->open_list_head) && timer->hw.close) @@ -362,7 +363,7 @@ int snd_timer_close(struct snd_timer_instance *timeri) slave->timer = NULL; spin_unlock_irq(&slave_active_lock); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); } if (timeri->private_free) timeri->private_free(timeri); @@ -835,7 +836,7 @@ static int snd_timer_dev_register(struct snd_device *dev) !timer->hw.resolution && timer->hw.c_resolution == NULL) return -EINVAL; - down(®ister_mutex); + mutex_lock(®ister_mutex); list_for_each(p, &snd_timer_list) { timer1 = list_entry(p, struct snd_timer, device_list); if (timer1->tmr_class > timer->tmr_class) @@ -857,11 +858,11 @@ static int snd_timer_dev_register(struct snd_device *dev) if (timer1->tmr_subdevice < timer->tmr_subdevice) continue; /* conflicts.. */ - up(®ister_mutex); + mutex_unlock(®ister_mutex); return -EBUSY; } list_add_tail(&timer->device_list, p); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return 0; } @@ -871,7 +872,7 @@ static int snd_timer_unregister(struct snd_timer *timer) struct snd_timer_instance *ti; snd_assert(timer != NULL, return -ENXIO); - down(®ister_mutex); + mutex_lock(®ister_mutex); if (! list_empty(&timer->open_list_head)) { snd_printk(KERN_WARNING "timer 0x%lx is busy?\n", (long)timer); list_for_each_safe(p, n, &timer->open_list_head) { @@ -881,7 +882,7 @@ static int snd_timer_unregister(struct snd_timer *timer) } } list_del(&timer->device_list); - up(®ister_mutex); + mutex_unlock(®ister_mutex); return snd_timer_free(timer); } @@ -1065,7 +1066,7 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, struct snd_timer_instance *ti; struct list_head *p, *q; - down(®ister_mutex); + mutex_lock(®ister_mutex); list_for_each(p, &snd_timer_list) { timer = list_entry(p, struct snd_timer, device_list); switch (timer->tmr_class) { @@ -1105,7 +1106,7 @@ static void snd_timer_proc_read(struct snd_info_entry *entry, } spin_unlock_irqrestore(&timer->lock, flags); } - up(®ister_mutex); + mutex_unlock(®ister_mutex); } static struct snd_info_entry *snd_timer_proc_entry = NULL; @@ -1269,7 +1270,7 @@ static int snd_timer_user_open(struct inode *inode, struct file *file) return -ENOMEM; spin_lock_init(&tu->qlock); init_waitqueue_head(&tu->qchange_sleep); - init_MUTEX(&tu->tread_sem); + mutex_init(&tu->tread_sem); tu->ticks = 1; tu->queue_size = 128; tu->queue = kmalloc(tu->queue_size * sizeof(struct snd_timer_read), @@ -1325,7 +1326,7 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) if (copy_from_user(&id, _tid, sizeof(id))) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); if (id.dev_class < 0) { /* first item */ if (list_empty(&snd_timer_list)) snd_timer_user_zero_id(&id); @@ -1407,7 +1408,7 @@ static int snd_timer_user_next_device(struct snd_timer_id __user *_tid) snd_timer_user_zero_id(&id); } } - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (copy_to_user(_tid, &id, sizeof(*_tid))) return -EFAULT; return 0; @@ -1432,7 +1433,7 @@ static int snd_timer_user_ginfo(struct file *file, tid = ginfo->tid; memset(ginfo, 0, sizeof(*ginfo)); ginfo->tid = tid; - down(®ister_mutex); + mutex_lock(®ister_mutex); t = snd_timer_find(&tid); if (t != NULL) { ginfo->card = t->card ? t->card->number : -1; @@ -1451,7 +1452,7 @@ static int snd_timer_user_ginfo(struct file *file, } else { err = -ENODEV; } - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (err >= 0 && copy_to_user(_ginfo, ginfo, sizeof(*ginfo))) err = -EFAULT; kfree(ginfo); @@ -1467,7 +1468,7 @@ static int snd_timer_user_gparams(struct file *file, if (copy_from_user(&gparams, _gparams, sizeof(gparams))) return -EFAULT; - down(®ister_mutex); + mutex_lock(®ister_mutex); t = snd_timer_find(&gparams.tid); if (!t) { err = -ENODEV; @@ -1483,7 +1484,7 @@ static int snd_timer_user_gparams(struct file *file, } err = t->hw.set_period(t, gparams.period_num, gparams.period_den); _error: - up(®ister_mutex); + mutex_unlock(®ister_mutex); return err; } @@ -1500,7 +1501,7 @@ static int snd_timer_user_gstatus(struct file *file, tid = gstatus.tid; memset(&gstatus, 0, sizeof(gstatus)); gstatus.tid = tid; - down(®ister_mutex); + mutex_lock(®ister_mutex); t = snd_timer_find(&tid); if (t != NULL) { if (t->hw.c_resolution) @@ -1517,7 +1518,7 @@ static int snd_timer_user_gstatus(struct file *file, } else { err = -ENODEV; } - up(®ister_mutex); + mutex_unlock(®ister_mutex); if (err >= 0 && copy_to_user(_gstatus, &gstatus, sizeof(gstatus))) err = -EFAULT; return err; @@ -1532,7 +1533,7 @@ static int snd_timer_user_tselect(struct file *file, int err = 0; tu = file->private_data; - down(&tu->tread_sem); + mutex_lock(&tu->tread_sem); if (tu->timeri) { snd_timer_close(tu->timeri); tu->timeri = NULL; @@ -1576,7 +1577,7 @@ static int snd_timer_user_tselect(struct file *file, } __err: - up(&tu->tread_sem); + mutex_unlock(&tu->tread_sem); return err; } @@ -1797,17 +1798,17 @@ static long snd_timer_user_ioctl(struct file *file, unsigned int cmd, { int xarg; - down(&tu->tread_sem); + mutex_lock(&tu->tread_sem); if (tu->timeri) { /* too late */ - up(&tu->tread_sem); + mutex_unlock(&tu->tread_sem); return -EBUSY; } if (get_user(xarg, p)) { - up(&tu->tread_sem); + mutex_unlock(&tu->tread_sem); return -EFAULT; } tu->tread = xarg ? 1 : 0; - up(&tu->tread_sem); + mutex_unlock(&tu->tread_sem); return 0; } case SNDRV_TIMER_IOCTL_GINFO: -- cgit 1.2.3-korg From ef9f0a42db987e7e2df72289fb4522d24027786b Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Mon, 16 Jan 2006 16:31:42 +0100 Subject: [ALSA] semaphore -> mutex (driver part) Semaphore to mutex conversion. The conversion was generated via scripts, and the result was validated automatically via a script as well. Signed-off-by: Ingo Molnar Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- include/sound/emux_synth.h | 2 +- include/sound/i2c.h | 10 +++--- include/sound/opl3.h | 3 +- include/sound/soundfont.h | 2 +- include/sound/util_mem.h | 4 ++- include/sound/vx_core.h | 2 +- sound/drivers/opl3/opl3_lib.c | 2 +- sound/drivers/opl3/opl3_seq.c | 10 +++--- sound/drivers/opl3/opl3_synth.c | 10 +++--- sound/drivers/opl4/opl4_lib.c | 2 +- sound/drivers/opl4/opl4_local.h | 2 +- sound/drivers/opl4/opl4_proc.c | 10 +++--- sound/drivers/opl4/opl4_seq.c | 12 +++---- sound/drivers/vx/vx_core.c | 2 +- sound/drivers/vx/vx_mixer.c | 72 ++++++++++++++++++++--------------------- sound/i2c/i2c.c | 2 +- sound/synth/emux/emux.c | 2 +- sound/synth/emux/emux_oss.c | 12 +++---- sound/synth/emux/emux_proc.c | 8 ++--- sound/synth/emux/emux_seq.c | 12 +++---- sound/synth/emux/soundfont.c | 6 ++-- sound/synth/util_mem.c | 15 +++++---- 22 files changed, 103 insertions(+), 99 deletions(-) diff --git a/include/sound/emux_synth.h b/include/sound/emux_synth.h index b2d6b2acc7c7dd..d8cb51b86c20c1 100644 --- a/include/sound/emux_synth.h +++ b/include/sound/emux_synth.h @@ -113,7 +113,7 @@ struct snd_emux { struct snd_emux_voice *voices; /* Voices (EMU 'channel') */ int use_time; /* allocation counter */ spinlock_t voice_lock; /* Lock for voice access */ - struct semaphore register_mutex; + struct mutex register_mutex; int client; /* For the sequencer client */ int ports[SNDRV_EMUX_MAX_PORTS]; /* The ports for this device */ struct snd_emux_port *portptrs[SNDRV_EMUX_MAX_PORTS]; diff --git a/include/sound/i2c.h b/include/sound/i2c.h index 81eb23ed761fa1..d125ff8c85e851 100644 --- a/include/sound/i2c.h +++ b/include/sound/i2c.h @@ -55,7 +55,7 @@ struct snd_i2c_bus { struct snd_card *card; /* card which I2C belongs to */ char name[32]; /* some useful label */ - struct semaphore lock_mutex; + struct mutex lock_mutex; struct snd_i2c_bus *master; /* master bus when SCK/SCL is shared */ struct list_head buses; /* master: slave buses sharing SCK/SCL, slave: link list */ @@ -84,17 +84,17 @@ int snd_i2c_device_free(struct snd_i2c_device *device); static inline void snd_i2c_lock(struct snd_i2c_bus *bus) { if (bus->master) - down(&bus->master->lock_mutex); + mutex_lock(&bus->master->lock_mutex); else - down(&bus->lock_mutex); + mutex_lock(&bus->lock_mutex); } static inline void snd_i2c_unlock(struct snd_i2c_bus *bus) { if (bus->master) - up(&bus->master->lock_mutex); + mutex_unlock(&bus->master->lock_mutex); else - up(&bus->lock_mutex); + mutex_unlock(&bus->lock_mutex); } int snd_i2c_sendbytes(struct snd_i2c_device *device, unsigned char *bytes, int count); diff --git a/include/sound/opl3.h b/include/sound/opl3.h index 83392641cb4745..444907166f9796 100644 --- a/include/sound/opl3.h +++ b/include/sound/opl3.h @@ -53,6 +53,7 @@ #include "driver.h" #include +#include #include "core.h" #include "hwdep.h" #include "timer.h" @@ -312,7 +313,7 @@ struct snd_opl3 { int sys_timer_status; /* system timer run status */ spinlock_t sys_timer_lock; /* Lock for system timer access */ #endif - struct semaphore access_mutex; /* locking */ + struct mutex access_mutex; /* locking */ }; /* opl3.c */ diff --git a/include/sound/soundfont.h b/include/sound/soundfont.h index 61a010c65d0278..f95d99ba7f7473 100644 --- a/include/sound/soundfont.h +++ b/include/sound/soundfont.h @@ -93,7 +93,7 @@ struct snd_sf_list { int sample_locked; /* locked time for sample */ struct snd_sf_callback callback; /* callback functions */ int presets_locked; - struct semaphore presets_mutex; + struct mutex presets_mutex; spinlock_t lock; struct snd_util_memhdr *memhdr; }; diff --git a/include/sound/util_mem.h b/include/sound/util_mem.h index 69944bbb5445fb..a1fb706b59a6ec 100644 --- a/include/sound/util_mem.h +++ b/include/sound/util_mem.h @@ -1,5 +1,7 @@ #ifndef __SOUND_UTIL_MEM_H #define __SOUND_UTIL_MEM_H + +#include /* * Copyright (C) 2000 Takashi Iwai * @@ -40,7 +42,7 @@ struct snd_util_memhdr { int nblocks; /* # of allocated blocks */ unsigned int used; /* used memory size */ int block_extra_size; /* extra data size of chunk */ - struct semaphore block_mutex; /* lock */ + struct mutex block_mutex; /* lock */ }; /* diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 5fd6f3305e0d38..9821a6194caae3 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -206,7 +206,7 @@ struct vx_core { int audio_monitor[4]; /* playback hw-monitor level */ unsigned char audio_monitor_active[4]; /* playback hw-monitor mute/unmute */ - struct semaphore mixer_mutex; + struct mutex mixer_mutex; const struct firmware *firmware[4]; /* loaded firmware data */ }; diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 1e0c76b9acfc93..4f85569767742b 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -358,7 +358,7 @@ int snd_opl3_new(struct snd_card *card, opl3->hardware = hardware; spin_lock_init(&opl3->reg_lock); spin_lock_init(&opl3->timer_lock); - init_MUTEX(&opl3->access_mutex); + mutex_init(&opl3->access_mutex); if ((err = snd_device_new(card, SNDRV_DEV_CODEC, opl3, &ops)) < 0) { snd_opl3_free(opl3); diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index c4ead790008a32..e26556d500e599 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -52,13 +52,13 @@ int snd_opl3_synth_setup(struct snd_opl3 * opl3) { int idx; - down(&opl3->access_mutex); + mutex_lock(&opl3->access_mutex); if (opl3->used) { - up(&opl3->access_mutex); + mutex_unlock(&opl3->access_mutex); return -EBUSY; } opl3->used++; - up(&opl3->access_mutex); + mutex_unlock(&opl3->access_mutex); snd_opl3_reset(opl3); @@ -91,9 +91,9 @@ void snd_opl3_synth_cleanup(struct snd_opl3 * opl3) spin_unlock_irqrestore(&opl3->sys_timer_lock, flags); snd_opl3_reset(opl3); - down(&opl3->access_mutex); + mutex_lock(&opl3->access_mutex); opl3->used--; - up(&opl3->access_mutex); + mutex_unlock(&opl3->access_mutex); } static int snd_opl3_synth_use(void *private_data, struct snd_seq_port_subscribe * info) diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 3534a0e3342a91..6db503f025b3a7 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -76,13 +76,13 @@ int snd_opl3_open(struct snd_hwdep * hw, struct file *file) { struct snd_opl3 *opl3 = hw->private_data; - down(&opl3->access_mutex); + mutex_lock(&opl3->access_mutex); if (opl3->used) { - up(&opl3->access_mutex); + mutex_unlock(&opl3->access_mutex); return -EAGAIN; } opl3->used++; - up(&opl3->access_mutex); + mutex_unlock(&opl3->access_mutex); return 0; } @@ -179,9 +179,9 @@ int snd_opl3_release(struct snd_hwdep * hw, struct file *file) struct snd_opl3 *opl3 = hw->private_data; snd_opl3_reset(opl3); - down(&opl3->access_mutex); + mutex_lock(&opl3->access_mutex); opl3->used--; - up(&opl3->access_mutex); + mutex_unlock(&opl3->access_mutex); return 0; } diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index ddfc10d04be2ee..4bc860ae02deec 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -214,7 +214,7 @@ int snd_opl4_create(struct snd_card *card, opl4->fm_port = fm_port; opl4->pcm_port = pcm_port; spin_lock_init(&opl4->reg_lock); - init_MUTEX(&opl4->access_mutex); + mutex_init(&opl4->access_mutex); err = snd_opl4_detect(opl4); if (err < 0) { diff --git a/sound/drivers/opl4/opl4_local.h b/sound/drivers/opl4/opl4_local.h index 7e088a4a2f4c56..470e5a758a0236 100644 --- a/sound/drivers/opl4/opl4_local.h +++ b/sound/drivers/opl4/opl4_local.h @@ -182,7 +182,7 @@ struct snd_opl4 { struct snd_info_entry *proc_entry; int memory_access; #endif - struct semaphore access_mutex; + struct mutex access_mutex; #if defined(CONFIG_SND_SEQUENCER) || defined(CONFIG_SND_SEQUENCER_MODULE) int used; diff --git a/sound/drivers/opl4/opl4_proc.c b/sound/drivers/opl4/opl4_proc.c index f4b4e74fcc1856..e552ec34166fed 100644 --- a/sound/drivers/opl4/opl4_proc.c +++ b/sound/drivers/opl4/opl4_proc.c @@ -28,13 +28,13 @@ static int snd_opl4_mem_proc_open(struct snd_info_entry *entry, { struct snd_opl4 *opl4 = entry->private_data; - down(&opl4->access_mutex); + mutex_lock(&opl4->access_mutex); if (opl4->memory_access) { - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); return -EBUSY; } opl4->memory_access++; - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); return 0; } @@ -43,9 +43,9 @@ static int snd_opl4_mem_proc_release(struct snd_info_entry *entry, { struct snd_opl4 *opl4 = entry->private_data; - down(&opl4->access_mutex); + mutex_lock(&opl4->access_mutex); opl4->memory_access--; - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); return 0; } diff --git a/sound/drivers/opl4/opl4_seq.c b/sound/drivers/opl4/opl4_seq.c index e3480326e735a1..dc0dcdc6c3139e 100644 --- a/sound/drivers/opl4/opl4_seq.c +++ b/sound/drivers/opl4/opl4_seq.c @@ -62,10 +62,10 @@ static int snd_opl4_seq_use(void *private_data, struct snd_seq_port_subscribe *i struct snd_opl4 *opl4 = private_data; int err; - down(&opl4->access_mutex); + mutex_lock(&opl4->access_mutex); if (opl4->used) { - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); return -EBUSY; } opl4->used++; @@ -73,12 +73,12 @@ static int snd_opl4_seq_use(void *private_data, struct snd_seq_port_subscribe *i if (info->sender.client != SNDRV_SEQ_CLIENT_SYSTEM) { err = snd_opl4_seq_use_inc(opl4); if (err < 0) { - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); return err; } } - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); snd_opl4_synth_reset(opl4); return 0; @@ -90,9 +90,9 @@ static int snd_opl4_seq_unuse(void *private_data, struct snd_seq_port_subscribe snd_opl4_synth_shutdown(opl4); - down(&opl4->access_mutex); + mutex_lock(&opl4->access_mutex); opl4->used--; - up(&opl4->access_mutex); + mutex_unlock(&opl4->access_mutex); if (info->sender.client != SNDRV_SEQ_CLIENT_SYSTEM) snd_opl4_seq_use_dec(opl4); diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 43f615d7a545ca..fa4a2b5c2d8d19 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -778,7 +778,7 @@ struct vx_core *snd_vx_create(struct snd_card *card, struct snd_vx_hardware *hw, chip->type = hw->type; chip->ops = ops; tasklet_init(&chip->tq, vx_interrupt, (unsigned long)chip); - init_MUTEX(&chip->mixer_mutex); + mutex_init(&chip->mixer_mutex); chip->card = card; card->private_data = chip; diff --git a/sound/drivers/vx/vx_mixer.c b/sound/drivers/vx/vx_mixer.c index 8ec2c605d2f05c..c1d7fcdd1973ec 100644 --- a/sound/drivers/vx/vx_mixer.c +++ b/sound/drivers/vx/vx_mixer.c @@ -427,10 +427,10 @@ static int vx_output_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ele { struct vx_core *chip = snd_kcontrol_chip(kcontrol); int codec = kcontrol->id.index; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->output_level[codec][0]; ucontrol->value.integer.value[1] = chip->output_level[codec][1]; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -438,7 +438,7 @@ static int vx_output_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele { struct vx_core *chip = snd_kcontrol_chip(kcontrol); int codec = kcontrol->id.index; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (ucontrol->value.integer.value[0] != chip->output_level[codec][0] || ucontrol->value.integer.value[1] != chip->output_level[codec][1]) { vx_set_analog_output_level(chip, codec, @@ -446,10 +446,10 @@ static int vx_output_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele ucontrol->value.integer.value[1]); chip->output_level[codec][0] = ucontrol->value.integer.value[0]; chip->output_level[codec][1] = ucontrol->value.integer.value[1]; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -502,14 +502,14 @@ static int vx_audio_src_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v static int vx_audio_src_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (chip->audio_source_target != ucontrol->value.enumerated.item[0]) { chip->audio_source_target = ucontrol->value.enumerated.item[0]; vx_sync_audio_source(chip); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -550,14 +550,14 @@ static int vx_clock_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ static int vx_clock_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct vx_core *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (chip->clock_mode != ucontrol->value.enumerated.item[0]) { chip->clock_mode = ucontrol->value.enumerated.item[0]; vx_set_clock(chip, chip->freq); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -587,10 +587,10 @@ static int vx_audio_gain_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ int audio = kcontrol->private_value & 0xff; int capture = (kcontrol->private_value >> 8) & 1; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->audio_gain[capture][audio]; ucontrol->value.integer.value[1] = chip->audio_gain[capture][audio+1]; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -600,15 +600,15 @@ static int vx_audio_gain_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ int audio = kcontrol->private_value & 0xff; int capture = (kcontrol->private_value >> 8) & 1; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (ucontrol->value.integer.value[0] != chip->audio_gain[capture][audio] || ucontrol->value.integer.value[1] != chip->audio_gain[capture][audio+1]) { vx_set_audio_gain(chip, audio, capture, ucontrol->value.integer.value[0]); vx_set_audio_gain(chip, audio+1, capture, ucontrol->value.integer.value[1]); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -617,10 +617,10 @@ static int vx_audio_monitor_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->audio_monitor[audio]; ucontrol->value.integer.value[1] = chip->audio_monitor[audio+1]; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -629,17 +629,17 @@ static int vx_audio_monitor_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (ucontrol->value.integer.value[0] != chip->audio_monitor[audio] || ucontrol->value.integer.value[1] != chip->audio_monitor[audio+1]) { vx_set_monitor_level(chip, audio, ucontrol->value.integer.value[0], chip->audio_monitor_active[audio]); vx_set_monitor_level(chip, audio+1, ucontrol->value.integer.value[1], chip->audio_monitor_active[audio+1]); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -657,10 +657,10 @@ static int vx_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->audio_active[audio]; ucontrol->value.integer.value[1] = chip->audio_active[audio+1]; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -669,15 +669,15 @@ static int vx_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (ucontrol->value.integer.value[0] != chip->audio_active[audio] || ucontrol->value.integer.value[1] != chip->audio_active[audio+1]) { vx_set_audio_switch(chip, audio, ucontrol->value.integer.value[0]); vx_set_audio_switch(chip, audio+1, ucontrol->value.integer.value[1]); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -686,10 +686,10 @@ static int vx_monitor_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->audio_monitor_active[audio]; ucontrol->value.integer.value[1] = chip->audio_monitor_active[audio+1]; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -698,17 +698,17 @@ static int vx_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ struct vx_core *chip = snd_kcontrol_chip(kcontrol); int audio = kcontrol->private_value & 0xff; - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (ucontrol->value.integer.value[0] != chip->audio_monitor_active[audio] || ucontrol->value.integer.value[1] != chip->audio_monitor_active[audio+1]) { vx_set_monitor_level(chip, audio, chip->audio_monitor[audio], ucontrol->value.integer.value[0]); vx_set_monitor_level(chip, audio+1, chip->audio_monitor[audio+1], ucontrol->value.integer.value[1]); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -756,12 +756,12 @@ static int vx_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu { struct vx_core *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); ucontrol->value.iec958.status[0] = (chip->uer_bits >> 0) & 0xff; ucontrol->value.iec958.status[1] = (chip->uer_bits >> 8) & 0xff; ucontrol->value.iec958.status[2] = (chip->uer_bits >> 16) & 0xff; ucontrol->value.iec958.status[3] = (chip->uer_bits >> 24) & 0xff; - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } @@ -783,14 +783,14 @@ static int vx_iec958_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu (ucontrol->value.iec958.status[1] << 8) | (ucontrol->value.iec958.status[2] << 16) | (ucontrol->value.iec958.status[3] << 24); - down(&chip->mixer_mutex); + mutex_lock(&chip->mixer_mutex); if (chip->uer_bits != val) { chip->uer_bits = val; vx_set_iec958_status(chip, val); - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 1; } - up(&chip->mixer_mutex); + mutex_unlock(&chip->mixer_mutex); return 0; } diff --git a/sound/i2c/i2c.c b/sound/i2c/i2c.c index c4e1f2c23ced40..edfe76fb007461 100644 --- a/sound/i2c/i2c.c +++ b/sound/i2c/i2c.c @@ -88,7 +88,7 @@ int snd_i2c_bus_create(struct snd_card *card, const char *name, bus = kzalloc(sizeof(*bus), GFP_KERNEL); if (bus == NULL) return -ENOMEM; - init_MUTEX(&bus->lock_mutex); + mutex_init(&bus->lock_mutex); INIT_LIST_HEAD(&bus->devices); INIT_LIST_HEAD(&bus->buses); bus->card = card; diff --git a/sound/synth/emux/emux.c b/sound/synth/emux/emux.c index 7c8e328fae6274..fc733bbf448731 100644 --- a/sound/synth/emux/emux.c +++ b/sound/synth/emux/emux.c @@ -45,7 +45,7 @@ int snd_emux_new(struct snd_emux **remu) return -ENOMEM; spin_lock_init(&emu->voice_lock); - init_MUTEX(&emu->register_mutex); + mutex_init(&emu->register_mutex); emu->client = -1; #ifdef CONFIG_SND_SEQUENCER_OSS diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index dfbfcfbe5dd26c..3436816727c819 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -117,10 +117,10 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) emu = closure; snd_assert(arg != NULL && emu != NULL, return -ENXIO); - down(&emu->register_mutex); + mutex_lock(&emu->register_mutex); if (!snd_emux_inc_count(emu)) { - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); return -EFAULT; } @@ -134,7 +134,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) if (p == NULL) { snd_printk("can't create port\n"); snd_emux_dec_count(emu); - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); return -ENOMEM; } @@ -148,7 +148,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) snd_emux_reset_port(p); - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); return 0; } @@ -191,13 +191,13 @@ snd_emux_close_seq_oss(struct snd_seq_oss_arg *arg) emu = p->emu; snd_assert(emu != NULL, return -ENXIO); - down(&emu->register_mutex); + mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); snd_soundfont_close_check(emu->sflist, SF_CLIENT_NO(p->chset.port)); snd_seq_event_port_detach(p->chset.client, p->chset.port); snd_emux_dec_count(emu); - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); return 0; } diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index a70a179f694739..1ba68ce302793c 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -37,7 +37,7 @@ snd_emux_proc_info_read(struct snd_info_entry *entry, int i; emu = entry->private_data; - down(&emu->register_mutex); + mutex_lock(&emu->register_mutex); if (emu->name) snd_iprintf(buf, "Device: %s\n", emu->name); snd_iprintf(buf, "Ports: %d\n", emu->num_ports); @@ -56,13 +56,13 @@ snd_emux_proc_info_read(struct snd_info_entry *entry, snd_iprintf(buf, "Memory Size: 0\n"); } if (emu->sflist) { - down(&emu->sflist->presets_mutex); + mutex_lock(&emu->sflist->presets_mutex); snd_iprintf(buf, "SoundFonts: %d\n", emu->sflist->fonts_size); snd_iprintf(buf, "Instruments: %d\n", emu->sflist->zone_counter); snd_iprintf(buf, "Samples: %d\n", emu->sflist->sample_counter); snd_iprintf(buf, "Locked Instruments: %d\n", emu->sflist->zone_locked); snd_iprintf(buf, "Locked Samples: %d\n", emu->sflist->sample_locked); - up(&emu->sflist->presets_mutex); + mutex_unlock(&emu->sflist->presets_mutex); } #if 0 /* debug */ if (emu->voices[0].state != SNDRV_EMUX_ST_OFF && emu->voices[0].ch >= 0) { @@ -103,7 +103,7 @@ snd_emux_proc_info_read(struct snd_info_entry *entry, snd_iprintf(buf, "sample_mode=%x, rate=%x\n", vp->reg.sample_mode, vp->reg.rate_offset); } #endif - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); } diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 1a973d7a90f8a3..8f00f07701c460 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -123,12 +123,12 @@ snd_emux_detach_seq(struct snd_emux *emu) if (emu->voices) snd_emux_terminate_all(emu); - down(&emu->register_mutex); + mutex_lock(&emu->register_mutex); if (emu->client >= 0) { snd_seq_delete_kernel_client(emu->client); emu->client = -1; } - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); } @@ -311,10 +311,10 @@ snd_emux_use(void *private_data, struct snd_seq_port_subscribe *info) emu = p->emu; snd_assert(emu != NULL, return -EINVAL); - down(&emu->register_mutex); + mutex_lock(&emu->register_mutex); snd_emux_init_port(p); snd_emux_inc_count(emu); - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); return 0; } @@ -332,10 +332,10 @@ snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *info) emu = p->emu; snd_assert(emu != NULL, return -EINVAL); - down(&emu->register_mutex); + mutex_lock(&emu->register_mutex); snd_emux_sounds_off_all(p); snd_emux_dec_count(emu); - up(&emu->register_mutex); + mutex_unlock(&emu->register_mutex); return 0; } diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 4c5754d4a2e800..32c27162dfb60a 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -79,7 +79,7 @@ static void lock_preset(struct snd_sf_list *sflist) { unsigned long flags; - down(&sflist->presets_mutex); + mutex_lock(&sflist->presets_mutex); spin_lock_irqsave(&sflist->lock, flags); sflist->presets_locked = 1; spin_unlock_irqrestore(&sflist->lock, flags); @@ -96,7 +96,7 @@ unlock_preset(struct snd_sf_list *sflist) spin_lock_irqsave(&sflist->lock, flags); sflist->presets_locked = 0; spin_unlock_irqrestore(&sflist->lock, flags); - up(&sflist->presets_mutex); + mutex_unlock(&sflist->presets_mutex); } @@ -1390,7 +1390,7 @@ snd_sf_new(struct snd_sf_callback *callback, struct snd_util_memhdr *hdr) if ((sflist = kzalloc(sizeof(*sflist), GFP_KERNEL)) == NULL) return NULL; - init_MUTEX(&sflist->presets_mutex); + mutex_init(&sflist->presets_mutex); spin_lock_init(&sflist->lock); sflist->memhdr = hdr; diff --git a/sound/synth/util_mem.c b/sound/synth/util_mem.c index 217e8e552a429b..1d9b11f345f891 100644 --- a/sound/synth/util_mem.c +++ b/sound/synth/util_mem.c @@ -18,6 +18,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include #include #include #include @@ -42,7 +43,7 @@ snd_util_memhdr_new(int memsize) if (hdr == NULL) return NULL; hdr->size = memsize; - init_MUTEX(&hdr->block_mutex); + mutex_init(&hdr->block_mutex); INIT_LIST_HEAD(&hdr->block); return hdr; @@ -136,9 +137,9 @@ struct snd_util_memblk * snd_util_mem_alloc(struct snd_util_memhdr *hdr, int size) { struct snd_util_memblk *blk; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); blk = __snd_util_mem_alloc(hdr, size); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return blk; } @@ -163,9 +164,9 @@ int snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk) { snd_assert(hdr && blk, return -EINVAL); - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return 0; } @@ -175,9 +176,9 @@ int snd_util_mem_free(struct snd_util_memhdr *hdr, struct snd_util_memblk *blk) int snd_util_mem_avail(struct snd_util_memhdr *hdr) { unsigned int size; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); size = hdr->size - hdr->used; - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return size; } -- cgit 1.2.3-korg From 8b7547f95cbe8a5940df62ed730646fdfcba5fda Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Mon, 16 Jan 2006 16:33:08 +0100 Subject: [ALSA] semaphore -> mutex (ISA part) Semaphore to mutex conversion. The conversion was generated via scripts, and the result was validated automatically via a script as well. Signed-off-by: Ingo Molnar Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- include/sound/ad1848.h | 2 +- include/sound/cs4231.h | 4 ++-- include/sound/gus.h | 6 +++--- include/sound/sb16_csp.h | 2 +- sound/isa/ad1848/ad1848_lib.c | 14 +++++++------- sound/isa/cs423x/cs4231_lib.c | 30 +++++++++++++++--------------- sound/isa/cs423x/cs4236_lib.c | 4 ++-- sound/isa/gus/gus_dma.c | 10 +++++----- sound/isa/gus/gus_main.c | 2 +- sound/isa/gus/gus_mem.c | 14 +++++++------- sound/isa/gus/gus_synth.c | 14 +++++++------- sound/isa/sb/sb16_csp.c | 12 ++++++------ 12 files changed, 57 insertions(+), 57 deletions(-) diff --git a/include/sound/ad1848.h b/include/sound/ad1848.h index 1a2759f3a292b6..57af1fe7b30998 100644 --- a/include/sound/ad1848.h +++ b/include/sound/ad1848.h @@ -154,7 +154,7 @@ struct snd_ad1848 { #endif spinlock_t reg_lock; - struct semaphore open_mutex; + struct mutex open_mutex; }; /* exported functions */ diff --git a/include/sound/cs4231.h b/include/sound/cs4231.h index ac6a5d8820888c..60b5b92a131985 100644 --- a/include/sound/cs4231.h +++ b/include/sound/cs4231.h @@ -248,8 +248,8 @@ struct snd_cs4231 { unsigned int c_dma_size; spinlock_t reg_lock; - struct semaphore mce_mutex; - struct semaphore open_mutex; + struct mutex mce_mutex; + struct mutex open_mutex; int (*rate_constraint) (struct snd_pcm_runtime *runtime); void (*set_playback_format) (struct snd_cs4231 *chip, struct snd_pcm_hw_params *hw_params, unsigned char pdfr); diff --git a/include/sound/gus.h b/include/sound/gus.h index 63da50fae7733e..68a664ab97f391 100644 --- a/include/sound/gus.h +++ b/include/sound/gus.h @@ -209,7 +209,7 @@ struct snd_gf1_mem { struct snd_gf1_bank_info banks_16[4]; struct snd_gf1_mem_block *first; struct snd_gf1_mem_block *last; - struct semaphore memory_mutex; + struct mutex memory_mutex; }; struct snd_gf1_dma_block { @@ -467,8 +467,8 @@ struct snd_gus_card { spinlock_t dma_lock; spinlock_t pcm_volume_level_lock; spinlock_t uart_cmd_lock; - struct semaphore dma_mutex; - struct semaphore register_mutex; + struct mutex dma_mutex; + struct mutex register_mutex; }; /* I/O functions for GF1/InterWave chip - gus_io.c */ diff --git a/include/sound/sb16_csp.h b/include/sound/sb16_csp.h index 3b44d4b370f5fe..caf6fe21514dca 100644 --- a/include/sound/sb16_csp.h +++ b/include/sound/sb16_csp.h @@ -158,7 +158,7 @@ struct snd_sb_csp { struct snd_kcontrol *qsound_switch; struct snd_kcontrol *qsound_space; - struct semaphore access_mutex; /* locking */ + struct mutex access_mutex; /* locking */ }; int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep); diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index b78530d7ea90c9..d4b0e580557ede 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -387,9 +387,9 @@ static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode) { unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); if (chip->mode & AD1848_MODE_OPEN) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return -EAGAIN; } snd_ad1848_mce_down(chip); @@ -432,7 +432,7 @@ static int snd_ad1848_open(struct snd_ad1848 *chip, unsigned int mode) spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } @@ -441,9 +441,9 @@ static void snd_ad1848_close(struct snd_ad1848 *chip) { unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); if (!chip->mode) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return; } /* disable IRQ */ @@ -471,7 +471,7 @@ static void snd_ad1848_close(struct snd_ad1848 *chip) spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = 0; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); } /* @@ -889,7 +889,7 @@ int snd_ad1848_create(struct snd_card *card, if (chip == NULL) return -ENOMEM; spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->port = port; chip->irq = -1; diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index eab7eb59b5f73e..823db824670193 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -531,7 +531,7 @@ static void snd_cs4231_playback_format(struct snd_cs4231 *chip, unsigned long flags; int full_calib = 1; - down(&chip->mce_mutex); + mutex_lock(&chip->mce_mutex); snd_cs4231_calibrate_mute(chip, 1); if (chip->hardware == CS4231_HW_CS4231A || (chip->hardware & CS4231_HW_CS4232_MASK)) { @@ -560,7 +560,7 @@ static void snd_cs4231_playback_format(struct snd_cs4231 *chip, snd_cs4231_mce_down(chip); } snd_cs4231_calibrate_mute(chip, 0); - up(&chip->mce_mutex); + mutex_unlock(&chip->mce_mutex); } static void snd_cs4231_capture_format(struct snd_cs4231 *chip, @@ -570,7 +570,7 @@ static void snd_cs4231_capture_format(struct snd_cs4231 *chip, unsigned long flags; int full_calib = 1; - down(&chip->mce_mutex); + mutex_lock(&chip->mce_mutex); snd_cs4231_calibrate_mute(chip, 1); if (chip->hardware == CS4231_HW_CS4231A || (chip->hardware & CS4231_HW_CS4232_MASK)) { @@ -603,7 +603,7 @@ static void snd_cs4231_capture_format(struct snd_cs4231 *chip, snd_cs4231_mce_down(chip); } snd_cs4231_calibrate_mute(chip, 0); - up(&chip->mce_mutex); + mutex_unlock(&chip->mce_mutex); } /* @@ -709,15 +709,15 @@ static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) { unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); if ((chip->mode & mode) || ((chip->mode & CS4231_MODE_OPEN) && chip->single_dma)) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return -EAGAIN; } if (chip->mode & CS4231_MODE_OPEN) { chip->mode |= mode; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } /* ok. now enable and ack CODEC IRQ */ @@ -737,7 +737,7 @@ static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } @@ -745,10 +745,10 @@ static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) { unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); chip->mode &= ~mode; if (chip->mode & CS4231_MODE_OPEN) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return; } snd_cs4231_calibrate_mute(chip, 1); @@ -785,7 +785,7 @@ static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) snd_cs4231_calibrate_mute(chip, 0); chip->mode = 0; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); } /* @@ -1408,8 +1408,8 @@ static int snd_cs4231_new(struct snd_card *card, chip->hwshare = hwshare; spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->mce_mutex); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->mce_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; @@ -1538,8 +1538,8 @@ int snd_cs4231_pcm(struct snd_cs4231 *chip, int device, struct snd_pcm **rpcm) return err; spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->mce_mutex); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->mce_mutex); + mutex_init(&chip->open_mutex); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs4231_capture_ops); diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index e36981d64ec5d2..1125ddb2b1aa82 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -841,7 +841,7 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn enable = ucontrol->value.integer.value[0] & 1; - down(&chip->mce_mutex); + mutex_lock(&chip->mce_mutex); snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); val = (chip->image[CS4231_ALT_FEATURE_1] & ~0x0e) | (0<<2) | (enable << 1); @@ -854,7 +854,7 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn snd_cs4236_ctrl_out(chip, 4, val); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_cs4231_mce_down(chip); - up(&chip->mce_mutex); + mutex_unlock(&chip->mce_mutex); #if 0 printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 930f4bc56f34c4..44ee5d3674a169 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -149,10 +149,10 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) int snd_gf1_dma_init(struct snd_gus_card * gus) { - down(&gus->dma_mutex); + mutex_lock(&gus->dma_mutex); gus->gf1.dma_shared++; if (gus->gf1.dma_shared > 1) { - up(&gus->dma_mutex); + mutex_unlock(&gus->dma_mutex); return 0; } gus->gf1.interrupt_handler_dma_write = snd_gf1_dma_interrupt; @@ -160,7 +160,7 @@ int snd_gf1_dma_init(struct snd_gus_card * gus) gus->gf1.dma_data_pcm_last = gus->gf1.dma_data_synth = gus->gf1.dma_data_synth_last = NULL; - up(&gus->dma_mutex); + mutex_unlock(&gus->dma_mutex); return 0; } @@ -168,7 +168,7 @@ int snd_gf1_dma_done(struct snd_gus_card * gus) { struct snd_gf1_dma_block *block; - down(&gus->dma_mutex); + mutex_lock(&gus->dma_mutex); gus->gf1.dma_shared--; if (!gus->gf1.dma_shared) { snd_dma_disable(gus->gf1.dma1); @@ -185,7 +185,7 @@ int snd_gf1_dma_done(struct snd_gus_card * gus) gus->gf1.dma_data_pcm_last = gus->gf1.dma_data_synth_last = NULL; } - up(&gus->dma_mutex); + mutex_unlock(&gus->dma_mutex); return 0; } diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 6d15b3d18a8752..53eeaf37007d15 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -225,7 +225,7 @@ int snd_gus_create(struct snd_card *card, spin_lock_init(&gus->dma_lock); spin_lock_init(&gus->pcm_volume_level_lock); spin_lock_init(&gus->uart_cmd_lock); - init_MUTEX(&gus->dma_mutex); + mutex_init(&gus->dma_mutex); if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, gus, &ops)) < 0) { snd_gus_free(gus); return err; diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index e8bdb860a19fc0..3c0d27aa08b306 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -34,9 +34,9 @@ static void snd_gf1_mem_info_read(struct snd_info_entry *entry, void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup) { if (!xup) { - down(&alloc->memory_mutex); + mutex_lock(&alloc->memory_mutex); } else { - up(&alloc->memory_mutex); + mutex_unlock(&alloc->memory_mutex); } } @@ -59,7 +59,7 @@ static struct snd_gf1_mem_block *snd_gf1_mem_xalloc(struct snd_gf1_mem * alloc, alloc->first = nblock; else nblock->prev->next = nblock; - up(&alloc->memory_mutex); + mutex_unlock(&alloc->memory_mutex); return NULL; } pblock = pblock->next; @@ -80,7 +80,7 @@ int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * blo { if (block->share) { /* ok.. shared block */ block->share--; - up(&alloc->memory_mutex); + mutex_unlock(&alloc->memory_mutex); return 0; } if (alloc->first == block) { @@ -244,7 +244,7 @@ int snd_gf1_mem_init(struct snd_gus_card * gus) #endif alloc = &gus->gf1.mem_alloc; - init_MUTEX(&alloc->memory_mutex); + mutex_init(&alloc->memory_mutex); alloc->first = alloc->last = NULL; if (!gus->gf1.memory) return 0; @@ -299,7 +299,7 @@ static void snd_gf1_mem_info_read(struct snd_info_entry *entry, gus = entry->private_data; alloc = &gus->gf1.mem_alloc; - down(&alloc->memory_mutex); + mutex_lock(&alloc->memory_mutex); snd_iprintf(buffer, "8-bit banks : \n "); for (i = 0; i < 4; i++) snd_iprintf(buffer, "0x%06x (%04ik)%s", alloc->banks_8[i].address, alloc->banks_8[i].size >> 10, i + 1 < 4 ? "," : ""); @@ -343,7 +343,7 @@ static void snd_gf1_mem_info_read(struct snd_info_entry *entry, } snd_iprintf(buffer, " Total: memory = %i, used = %i, free = %i\n", total, used, total - used); - up(&alloc->memory_mutex); + mutex_unlock(&alloc->memory_mutex); #if 0 ultra_iprintf(buffer, " Verify: free = %i, max 8-bit block = %i, max 16-bit block = %i\n", ultra_memory_free_size(card, &card->gf1.mem_alloc), diff --git a/sound/isa/gus/gus_synth.c b/sound/isa/gus/gus_synth.c index 85a1b051f09a4f..2767cc187ae39e 100644 --- a/sound/isa/gus/gus_synth.c +++ b/sound/isa/gus/gus_synth.c @@ -55,9 +55,9 @@ static int snd_gus_synth_use(void *private_data, struct snd_seq_port_subscribe * if (info->voices > 32) return -EINVAL; - down(&gus->register_mutex); + mutex_lock(&gus->register_mutex); if (!snd_gus_use_inc(gus)) { - up(&gus->register_mutex); + mutex_unlock(&gus->register_mutex); return -EFAULT; } for (idx = 0; idx < info->voices; idx++) { @@ -65,12 +65,12 @@ static int snd_gus_synth_use(void *private_data, struct snd_seq_port_subscribe * if (voice == NULL) { snd_gus_synth_free_voices(gus, info->sender.client, info->sender.port); snd_gus_use_dec(gus); - up(&gus->register_mutex); + mutex_unlock(&gus->register_mutex); return -EBUSY; } voice->index = idx; } - up(&gus->register_mutex); + mutex_unlock(&gus->register_mutex); return 0; } @@ -79,10 +79,10 @@ static int snd_gus_synth_unuse(void *private_data, struct snd_seq_port_subscribe struct snd_gus_port * port = private_data; struct snd_gus_card * gus = port->gus; - down(&gus->register_mutex); + mutex_lock(&gus->register_mutex); snd_gus_synth_free_voices(gus, info->sender.client, info->sender.port); snd_gus_use_dec(gus); - up(&gus->register_mutex); + mutex_unlock(&gus->register_mutex); return 0; } @@ -223,7 +223,7 @@ static int snd_gus_synth_new_device(struct snd_seq_device *dev) if (gus == NULL) return -EINVAL; - init_MUTEX(&gus->register_mutex); + mutex_init(&gus->register_mutex); gus->gf1.seq_client = -1; /* allocate new client */ diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 9c2b5efbacbf51..9703c68e4e0803 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -138,7 +138,7 @@ int snd_sb_csp_new(struct snd_sb *chip, int device, struct snd_hwdep ** rhwdep) p->ops.csp_stop = snd_sb_csp_stop; p->ops.csp_qsound_transfer = snd_sb_csp_qsound_transfer; - init_MUTEX(&p->access_mutex); + mutex_init(&p->access_mutex); sprintf(hw->name, "CSP v%d.%d", (version >> 4), (version & 0x0f)); hw->iface = SNDRV_HWDEP_IFACE_SB16CSP; hw->private_data = p; @@ -265,13 +265,13 @@ static int snd_sb_csp_release(struct snd_hwdep * hw, struct file *file) */ static int snd_sb_csp_use(struct snd_sb_csp * p) { - down(&p->access_mutex); + mutex_lock(&p->access_mutex); if (p->used) { - up(&p->access_mutex); + mutex_unlock(&p->access_mutex); return -EAGAIN; } p->used++; - up(&p->access_mutex); + mutex_unlock(&p->access_mutex); return 0; @@ -282,9 +282,9 @@ static int snd_sb_csp_use(struct snd_sb_csp * p) */ static int snd_sb_csp_unuse(struct snd_sb_csp * p) { - down(&p->access_mutex); + mutex_lock(&p->access_mutex); p->used--; - up(&p->access_mutex); + mutex_unlock(&p->access_mutex); return 0; } -- cgit 1.2.3-korg From 62932df8fb20ba2fb53a95fa52445eba22e821fe Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Mon, 16 Jan 2006 16:34:20 +0100 Subject: [ALSA] semaphore -> mutex (PCI part) Semaphore to mutex conversion. The conversion was generated via scripts, and the result was validated automatically via a script as well. Signed-off-by: Ingo Molnar Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 4 +- include/sound/ak4531_codec.h | 2 +- include/sound/cs46xx.h | 2 +- include/sound/emu10k1.h | 4 +- sound/pci/ac97/ac97_codec.c | 45 +++++++++---------- sound/pci/ac97/ac97_patch.c | 10 +++-- sound/pci/ac97/ac97_pcm.c | 6 ++- sound/pci/ac97/ac97_proc.c | 14 +++--- sound/pci/ac97/ak4531_codec.c | 28 ++++++------ sound/pci/atiixp.c | 21 ++++----- sound/pci/atiixp_modem.c | 13 +++--- sound/pci/cmipci.c | 25 +++++------ sound/pci/cs46xx/cs46xx_lib.c | 52 +++++++++++----------- sound/pci/cs46xx/dsp_spos.c | 58 +++++++++++++------------ sound/pci/cs46xx/dsp_spos_scb_lib.c | 6 ++- sound/pci/emu10k1/emu10k1_main.c | 5 ++- sound/pci/emu10k1/emufx.c | 22 +++++----- sound/pci/emu10k1/memory.c | 26 +++++------ sound/pci/ens1370.c | 32 +++++++------- sound/pci/es1968.c | 22 +++++----- sound/pci/hda/hda_codec.c | 51 +++++++++++----------- sound/pci/hda/hda_codec.h | 4 +- sound/pci/hda/hda_intel.c | 17 ++++---- sound/pci/hda/patch_analog.c | 28 ++++++------ sound/pci/ice1712/aureon.c | 38 ++++++++-------- sound/pci/ice1712/delta.c | 26 +++++------ sound/pci/ice1712/hoontech.c | 26 +++++------ sound/pci/ice1712/ice1712.c | 7 +-- sound/pci/ice1712/ice1712.h | 10 ++--- sound/pci/ice1712/ice1724.c | 37 ++++++++-------- sound/pci/ice1712/phase.c | 10 +++-- sound/pci/ice1712/pontis.c | 86 +++++++++++++++++++------------------ sound/pci/korg1212/korg1212.c | 17 ++++---- sound/pci/mixart/mixart.c | 21 ++++----- sound/pci/mixart/mixart.h | 7 +-- sound/pci/mixart/mixart_core.c | 18 ++++---- sound/pci/mixart/mixart_mixer.c | 52 +++++++++++----------- sound/pci/nm256/nm256.c | 16 ++++--- sound/pci/pcxhr/pcxhr.c | 36 ++++++++-------- sound/pci/pcxhr/pcxhr.h | 5 ++- sound/pci/pcxhr/pcxhr_mixer.c | 75 ++++++++++++++++---------------- sound/pci/trident/trident_memory.c | 36 ++++++++-------- sound/pci/vx222/vx222_ops.c | 18 ++++---- 43 files changed, 549 insertions(+), 489 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index b0b3ea7b365e81..ad3fe046f6cf81 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -456,8 +456,8 @@ struct snd_ac97 { struct snd_info_entry *proc_regs; unsigned short subsystem_vendor; unsigned short subsystem_device; - struct semaphore reg_mutex; - struct semaphore page_mutex; /* mutex for AD18xx multi-codecs and paging (2.3) */ + struct mutex reg_mutex; + struct mutex page_mutex; /* mutex for AD18xx multi-codecs and paging (2.3) */ unsigned short num; /* number of codec: 0 = primary, 1 = secondary */ unsigned short addr; /* physical address of codec [0-3] */ unsigned int id; /* identification of codec */ diff --git a/include/sound/ak4531_codec.h b/include/sound/ak4531_codec.h index edf04070ce7c44..fb30faab43a831 100644 --- a/include/sound/ak4531_codec.h +++ b/include/sound/ak4531_codec.h @@ -71,7 +71,7 @@ struct snd_ak4531 { void (*private_free) (struct snd_ak4531 *ak4531); /* --- */ unsigned char regs[0x20]; - struct semaphore reg_mutex; + struct mutex reg_mutex; }; int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index 199b5098ff7e8e..80b2979c0cba94 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1711,7 +1711,7 @@ struct snd_cs46xx { int current_gpio; #endif #ifdef CONFIG_SND_CS46XX_NEW_DSP - struct semaphore spos_mutex; + struct mutex spos_mutex; struct dsp_spos_instance * dsp_spos_instance; diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h index 951e40d720d9b1..186e00ad9e79fe 100644 --- a/include/sound/emu10k1.h +++ b/include/sound/emu10k1.h @@ -33,6 +33,7 @@ #include #include #include +#include #include /* ------------------- DEFINES -------------------- */ @@ -1022,7 +1023,7 @@ struct snd_emu10k1_fx8010 { int gpr_size; /* size of allocated GPR controls */ int gpr_count; /* count of used kcontrols */ struct list_head gpr_ctl; /* GPR controls */ - struct semaphore lock; + struct mutex lock; struct snd_emu10k1_fx8010_pcm pcm[8]; spinlock_t irq_lock; struct snd_emu10k1_fx8010_irq *irq_handlers; @@ -1122,7 +1123,6 @@ struct snd_emu10k1 { spinlock_t reg_lock; spinlock_t emu_lock; spinlock_t voice_lock; - struct semaphore ptb_lock; struct snd_emu10k1_voice voices[NUM_G]; struct snd_emu10k1_voice p16v_voices[4]; diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 3020ca2b602bd1..6108cdc5efb655 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include @@ -296,11 +297,11 @@ void snd_ac97_write_cache(struct snd_ac97 *ac97, unsigned short reg, unsigned sh { if (!snd_ac97_valid_reg(ac97, reg)) return; - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); ac97->regs[reg] = value; ac97->bus->ops->write(ac97, reg, value); set_bit(reg, ac97->reg_accessed); - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); } /** @@ -321,14 +322,14 @@ int snd_ac97_update(struct snd_ac97 *ac97, unsigned short reg, unsigned short va if (!snd_ac97_valid_reg(ac97, reg)) return -EINVAL; - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); change = ac97->regs[reg] != value; if (change) { ac97->regs[reg] = value; ac97->bus->ops->write(ac97, reg, value); } set_bit(reg, ac97->reg_accessed); - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); return change; } @@ -351,9 +352,9 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho if (!snd_ac97_valid_reg(ac97, reg)) return -EINVAL; - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); change = snd_ac97_update_bits_nolock(ac97, reg, mask, value); - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); return change; } @@ -380,12 +381,12 @@ static int snd_ac97_ad18xx_update_pcm_bits(struct snd_ac97 *ac97, int codec, uns int change; unsigned short old, new, cfg; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); old = ac97->spec.ad18xx.pcmreg[codec]; new = (old & ~mask) | value; change = old != new; if (change) { - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); cfg = snd_ac97_read_cache(ac97, AC97_AD_SERIAL_CFG); ac97->spec.ad18xx.pcmreg[codec] = new; /* select single codec */ @@ -397,9 +398,9 @@ static int snd_ac97_ad18xx_update_pcm_bits(struct snd_ac97 *ac97, int codec, uns /* select all codecs */ ac97->bus->ops->write(ac97, AC97_AD_SERIAL_CFG, cfg | 0x7000); - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); } - up(&ac97->page_mutex); + mutex_unlock(&ac97->page_mutex); return change; } @@ -467,7 +468,7 @@ static int snd_ac97_page_save(struct snd_ac97 *ac97, int reg, struct snd_kcontro (ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23 && (reg >= 0x60 && reg < 0x70)) { unsigned short page = (kcontrol->private_value >> 26) & 0x0f; - down(&ac97->page_mutex); /* lock paging */ + mutex_lock(&ac97->page_mutex); /* lock paging */ page_save = snd_ac97_read(ac97, AC97_INT_PAGING) & AC97_PAGE_MASK; snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page); } @@ -478,7 +479,7 @@ static void snd_ac97_page_restore(struct snd_ac97 *ac97, int page_save) { if (page_save >= 0) { snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page_save); - up(&ac97->page_mutex); /* unlock paging */ + mutex_unlock(&ac97->page_mutex); /* unlock paging */ } } @@ -674,12 +675,12 @@ static int snd_ac97_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_ { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); ucontrol->value.iec958.status[0] = ac97->spdif_status & 0xff; ucontrol->value.iec958.status[1] = (ac97->spdif_status >> 8) & 0xff; ucontrol->value.iec958.status[2] = (ac97->spdif_status >> 16) & 0xff; ucontrol->value.iec958.status[3] = (ac97->spdif_status >> 24) & 0xff; - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); return 0; } @@ -718,7 +719,7 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ } } - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); change = ac97->spdif_status != new; ac97->spdif_status = new; @@ -746,7 +747,7 @@ static int snd_ac97_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF); /* turn on again */ } } - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); return change; } @@ -763,7 +764,7 @@ static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ value = (ucontrol->value.integer.value[0] & mask); - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); mask <<= shift; value <<= shift; old = snd_ac97_read_cache(ac97, reg); @@ -777,7 +778,7 @@ static int snd_ac97_put_spsa(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ if (extst & AC97_EA_SPDIF) snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF); /* turn on again */ } - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); return change; } @@ -888,10 +889,10 @@ static int snd_ac97_ad18xx_pcm_get_volume(struct snd_kcontrol *kcontrol, struct struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); int codec = kcontrol->private_value & 3; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); ucontrol->value.integer.value[0] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 0) & 31); ucontrol->value.integer.value[1] = 31 - ((ac97->spec.ad18xx.pcmreg[codec] >> 8) & 31); - up(&ac97->page_mutex); + mutex_unlock(&ac97->page_mutex); return 0; } @@ -1856,8 +1857,8 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, ac97->limited_regs = template->limited_regs; memcpy(ac97->reg_accessed, template->reg_accessed, sizeof(ac97->reg_accessed)); bus->codec[ac97->num] = ac97; - init_MUTEX(&ac97->reg_mutex); - init_MUTEX(&ac97->page_mutex); + mutex_init(&ac97->reg_mutex); + mutex_init(&ac97->page_mutex); #ifdef CONFIG_PCI if (ac97->pci) { diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a444a78c7c9444..8bc79cbe321523 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -27,6 +27,8 @@ #include #include #include +#include + #include #include #include @@ -55,12 +57,12 @@ static int ac97_update_bits_page(struct snd_ac97 *ac97, unsigned short reg, unsi unsigned short page_save; int ret; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); page_save = snd_ac97_read(ac97, AC97_INT_PAGING) & AC97_PAGE_MASK; snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page); ret = snd_ac97_update_bits(ac97, reg, mask, value); snd_ac97_update_bits(ac97, AC97_INT_PAGING, AC97_PAGE_MASK, page_save); - up(&ac97->page_mutex); /* unlock paging */ + mutex_unlock(&ac97->page_mutex); /* unlock paging */ return ret; } @@ -897,12 +899,12 @@ static int snd_ac97_stac9708_put_bias(struct snd_kcontrol *kcontrol, struct snd_ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); int err; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); snd_ac97_write(ac97, AC97_SIGMATEL_BIAS1, 0xabba); err = snd_ac97_update_bits(ac97, AC97_SIGMATEL_BIAS2, 0x0010, (ucontrol->value.integer.value[0] & 1) << 4); snd_ac97_write(ac97, AC97_SIGMATEL_BIAS1, 0); - up(&ac97->page_mutex); + mutex_unlock(&ac97->page_mutex); return err; } diff --git a/sound/pci/ac97/ac97_pcm.c b/sound/pci/ac97/ac97_pcm.c index c3e590bf7a02b4..512a3583b0ce91 100644 --- a/sound/pci/ac97/ac97_pcm.c +++ b/sound/pci/ac97/ac97_pcm.c @@ -27,6 +27,8 @@ #include #include #include +#include + #include #include #include @@ -206,7 +208,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate) mask = AC97_SC_SPSR_MASK; } - down(&ac97->reg_mutex); + mutex_lock(&ac97->reg_mutex); old = snd_ac97_read(ac97, reg) & mask; if (old != bits) { snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); @@ -231,7 +233,7 @@ static int set_spdif_rate(struct snd_ac97 *ac97, unsigned short rate) ac97->spdif_status = sbits; } snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, AC97_EA_SPDIF); - up(&ac97->reg_mutex); + mutex_unlock(&ac97->reg_mutex); return 0; } diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 7134b3f55fb526..4d523df79cc71e 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -24,6 +24,8 @@ #include #include +#include + #include #include #include @@ -338,7 +340,7 @@ static void snd_ac97_proc_read(struct snd_info_entry *entry, struct snd_info_buf { struct snd_ac97 *ac97 = entry->private_data; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { // Analog Devices AD1881/85/86 int idx; for (idx = 0; idx < 3; idx++) @@ -364,7 +366,7 @@ static void snd_ac97_proc_read(struct snd_info_entry *entry, struct snd_info_buf } else { snd_ac97_proc_read_main(ac97, buffer, 0); } - up(&ac97->page_mutex); + mutex_unlock(&ac97->page_mutex); } #ifdef CONFIG_SND_DEBUG @@ -374,7 +376,7 @@ static void snd_ac97_proc_regs_write(struct snd_info_entry *entry, struct snd_in struct snd_ac97 *ac97 = entry->private_data; char line[64]; unsigned int reg, val; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; @@ -382,7 +384,7 @@ static void snd_ac97_proc_regs_write(struct snd_info_entry *entry, struct snd_in if (reg < 0x80 && (reg & 1) == 0 && val <= 0xffff) snd_ac97_write_cache(ac97, reg, val); } - up(&ac97->page_mutex); + mutex_unlock(&ac97->page_mutex); } #endif @@ -401,7 +403,7 @@ static void snd_ac97_proc_regs_read(struct snd_info_entry *entry, { struct snd_ac97 *ac97 = entry->private_data; - down(&ac97->page_mutex); + mutex_lock(&ac97->page_mutex); if ((ac97->id & 0xffffff40) == AC97_ID_AD1881) { // Analog Devices AD1881/85/86 int idx; @@ -417,7 +419,7 @@ static void snd_ac97_proc_regs_read(struct snd_info_entry *entry, } else { snd_ac97_proc_regs_read_main(ac97, buffer, 0); } - up(&ac97->page_mutex); + mutex_unlock(&ac97->page_mutex); } void snd_ac97_proc_init(struct snd_ac97 * ac97) diff --git a/sound/pci/ac97/ak4531_codec.c b/sound/pci/ac97/ak4531_codec.c index dcfb5036ff8bdb..0fb7b34073129a 100644 --- a/sound/pci/ac97/ak4531_codec.c +++ b/sound/pci/ac97/ak4531_codec.c @@ -23,6 +23,8 @@ #include #include #include +#include + #include #include @@ -82,9 +84,9 @@ static int snd_ak4531_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e int invert = (kcontrol->private_value >> 22) & 1; int val; - down(&ak4531->reg_mutex); + mutex_lock(&ak4531->reg_mutex); val = (ak4531->regs[reg] >> shift) & mask; - up(&ak4531->reg_mutex); + mutex_unlock(&ak4531->reg_mutex); if (invert) { val = mask - val; } @@ -107,11 +109,11 @@ static int snd_ak4531_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_e val = mask - val; } val <<= shift; - down(&ak4531->reg_mutex); + mutex_lock(&ak4531->reg_mutex); val = (ak4531->regs[reg] & ~(mask << shift)) | val; change = val != ak4531->regs[reg]; ak4531->write(ak4531, reg, ak4531->regs[reg] = val); - up(&ak4531->reg_mutex); + mutex_unlock(&ak4531->reg_mutex); return change; } @@ -143,10 +145,10 @@ static int snd_ak4531_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e int invert = (kcontrol->private_value >> 22) & 1; int left, right; - down(&ak4531->reg_mutex); + mutex_lock(&ak4531->reg_mutex); left = (ak4531->regs[left_reg] >> left_shift) & mask; right = (ak4531->regs[right_reg] >> right_shift) & mask; - up(&ak4531->reg_mutex); + mutex_unlock(&ak4531->reg_mutex); if (invert) { left = mask - left; right = mask - right; @@ -176,7 +178,7 @@ static int snd_ak4531_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e } left <<= left_shift; right <<= right_shift; - down(&ak4531->reg_mutex); + mutex_lock(&ak4531->reg_mutex); if (left_reg == right_reg) { left = (ak4531->regs[left_reg] & ~((mask << left_shift) | (mask << right_shift))) | left | right; change = left != ak4531->regs[left_reg]; @@ -188,7 +190,7 @@ static int snd_ak4531_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_e ak4531->write(ak4531, left_reg, ak4531->regs[left_reg] = left); ak4531->write(ak4531, right_reg, ak4531->regs[right_reg] = right); } - up(&ak4531->reg_mutex); + mutex_unlock(&ak4531->reg_mutex); return change; } @@ -215,12 +217,12 @@ static int snd_ak4531_get_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl int left_shift = (kcontrol->private_value >> 16) & 0x0f; int right_shift = (kcontrol->private_value >> 24) & 0x0f; - down(&ak4531->reg_mutex); + mutex_lock(&ak4531->reg_mutex); ucontrol->value.integer.value[0] = (ak4531->regs[reg1] >> left_shift) & 1; ucontrol->value.integer.value[1] = (ak4531->regs[reg2] >> left_shift) & 1; ucontrol->value.integer.value[2] = (ak4531->regs[reg1] >> right_shift) & 1; ucontrol->value.integer.value[3] = (ak4531->regs[reg2] >> right_shift) & 1; - up(&ak4531->reg_mutex); + mutex_unlock(&ak4531->reg_mutex); return 0; } @@ -234,7 +236,7 @@ static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl int change; int val1, val2; - down(&ak4531->reg_mutex); + mutex_lock(&ak4531->reg_mutex); val1 = ak4531->regs[reg1] & ~((1 << left_shift) | (1 << right_shift)); val2 = ak4531->regs[reg2] & ~((1 << left_shift) | (1 << right_shift)); val1 |= (ucontrol->value.integer.value[0] & 1) << left_shift; @@ -244,7 +246,7 @@ static int snd_ak4531_put_input_sw(struct snd_kcontrol *kcontrol, struct snd_ctl change = val1 != ak4531->regs[reg1] || val2 != ak4531->regs[reg2]; ak4531->write(ak4531, reg1, ak4531->regs[reg1] = val1); ak4531->write(ak4531, reg2, ak4531->regs[reg2] = val2); - up(&ak4531->reg_mutex); + mutex_unlock(&ak4531->reg_mutex); return change; } @@ -366,7 +368,7 @@ int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, if (ak4531 == NULL) return -ENOMEM; *ak4531 = *_ak4531; - init_MUTEX(&ak4531->reg_mutex); + mutex_init(&ak4531->reg_mutex); if ((err = snd_component_add(card, "AK4531")) < 0) { snd_ak4531_free(ak4531); return err; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index b7217adaf1d76b..12e618851262a3 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -277,7 +278,7 @@ struct atiixp { unsigned int codec_not_ready_bits; /* for codec detection */ int spdif_over_aclink; /* passed from the module option */ - struct semaphore open_mutex; /* playback open mutex */ + struct mutex open_mutex; /* playback open mutex */ }; @@ -1051,9 +1052,9 @@ static int snd_atiixp_playback_open(struct snd_pcm_substream *substream) struct atiixp *chip = snd_pcm_substream_chip(substream); int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_PLAYBACK], 0); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); if (err < 0) return err; substream->runtime->hw.channels_max = chip->max_channels; @@ -1068,9 +1069,9 @@ static int snd_atiixp_playback_close(struct snd_pcm_substream *substream) { struct atiixp *chip = snd_pcm_substream_chip(substream); int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_PLAYBACK]); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return err; } @@ -1090,12 +1091,12 @@ static int snd_atiixp_spdif_open(struct snd_pcm_substream *substream) { struct atiixp *chip = snd_pcm_substream_chip(substream); int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); if (chip->spdif_over_aclink) /* share DMA_PLAYBACK */ err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_PLAYBACK], 2); else err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_SPDIF], -1); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return err; } @@ -1103,12 +1104,12 @@ static int snd_atiixp_spdif_close(struct snd_pcm_substream *substream) { struct atiixp *chip = snd_pcm_substream_chip(substream); int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); if (chip->spdif_over_aclink) err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_PLAYBACK]); else err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_SPDIF]); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return err; } @@ -1560,7 +1561,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, } spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->pci = pci; chip->irq = -1; diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 8d8fd5a4ed35ae..1d376604464365 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -255,7 +256,7 @@ struct atiixp_modem { unsigned int codec_not_ready_bits; /* for codec detection */ int spdif_over_aclink; /* passed from the module option */ - struct semaphore open_mutex; /* playback open mutex */ + struct mutex open_mutex; /* playback open mutex */ }; @@ -911,9 +912,9 @@ static int snd_atiixp_playback_open(struct snd_pcm_substream *substream) struct atiixp_modem *chip = snd_pcm_substream_chip(substream); int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); err = snd_atiixp_pcm_open(substream, &chip->dmas[ATI_DMA_PLAYBACK], 0); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); if (err < 0) return err; return 0; @@ -923,9 +924,9 @@ static int snd_atiixp_playback_close(struct snd_pcm_substream *substream) { struct atiixp_modem *chip = snd_pcm_substream_chip(substream); int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); err = snd_atiixp_pcm_close(substream, &chip->dmas[ATI_DMA_PLAYBACK]); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return err; } @@ -1233,7 +1234,7 @@ static int __devinit snd_atiixp_create(struct snd_card *card, } spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->pci = pci; chip->irq = -1; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index c03b0a0a3b27a0..2ecbddbbdcf073 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include @@ -439,7 +440,7 @@ struct cmipci { struct snd_pcm_hardware *hw_info[3]; /* for playbacks */ int opened[2]; /* open mode */ - struct semaphore open_mutex; + struct mutex open_mutex; unsigned int mixer_insensitive: 1; struct snd_kcontrol *mixer_res_ctl[CM_SAVED_MIXERS]; @@ -641,14 +642,14 @@ static int snd_cmipci_playback2_hw_params(struct snd_pcm_substream *substream, { struct cmipci *cm = snd_pcm_substream_chip(substream); if (params_channels(hw_params) > 2) { - down(&cm->open_mutex); + mutex_lock(&cm->open_mutex); if (cm->opened[CM_CH_PLAY]) { - up(&cm->open_mutex); + mutex_unlock(&cm->open_mutex); return -EBUSY; } /* reserve the channel A */ cm->opened[CM_CH_PLAY] = CM_OPEN_PLAYBACK_MULTI; - up(&cm->open_mutex); + mutex_unlock(&cm->open_mutex); } return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -1461,9 +1462,9 @@ static int open_device_check(struct cmipci *cm, int mode, struct snd_pcm_substre * pcm framework doesn't pass file pointer before actually opened, * we can't know whether blocking mode or not in open callback.. */ - down(&cm->open_mutex); + mutex_lock(&cm->open_mutex); if (cm->opened[ch]) { - up(&cm->open_mutex); + mutex_unlock(&cm->open_mutex); return -EBUSY; } cm->opened[ch] = mode; @@ -1475,7 +1476,7 @@ static int open_device_check(struct cmipci *cm, int mode, struct snd_pcm_substre snd_cmipci_clear_bit(cm, CM_REG_MISC_CTRL, CM_ENDBDAC); spin_unlock_irq(&cm->reg_lock); } - up(&cm->open_mutex); + mutex_unlock(&cm->open_mutex); return 0; } @@ -1483,7 +1484,7 @@ static void close_device_check(struct cmipci *cm, int mode) { int ch = mode & CM_OPEN_CH_MASK; - down(&cm->open_mutex); + mutex_lock(&cm->open_mutex); if (cm->opened[ch] == mode) { if (cm->channel[ch].substream) { snd_cmipci_ch_reset(cm, ch); @@ -1499,7 +1500,7 @@ static void close_device_check(struct cmipci *cm, int mode) spin_unlock_irq(&cm->reg_lock); } } - up(&cm->open_mutex); + mutex_unlock(&cm->open_mutex); } /* @@ -1546,7 +1547,7 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) if ((err = open_device_check(cm, CM_OPEN_PLAYBACK2, substream)) < 0) /* use channel B */ return err; runtime->hw = snd_cmipci_playback2; - down(&cm->open_mutex); + mutex_lock(&cm->open_mutex); if (! cm->opened[CM_CH_PLAY]) { if (cm->can_multi_ch) { runtime->hw.channels_max = cm->max_channels; @@ -1559,7 +1560,7 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); } - up(&cm->open_mutex); + mutex_unlock(&cm->open_mutex); return 0; } @@ -2844,7 +2845,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc } spin_lock_init(&cm->reg_lock); - init_MUTEX(&cm->open_mutex); + mutex_init(&cm->open_mutex); cm->device = pci->device; cm->card = card; cm->pci = pci; diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 8fb275d6eb77c6..69dbf542a6de20 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -53,6 +53,8 @@ #include #include #include +#include + #include #include @@ -909,22 +911,22 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, #ifdef CONFIG_SND_CS46XX_NEW_DSP snd_assert (sample_rate != 0, return -ENXIO); - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); if (_cs46xx_adjust_sample_rate (chip,cpcm,sample_rate)) { - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return -ENXIO; } snd_assert (cpcm->pcm_channel != NULL); if (!cpcm->pcm_channel) { - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return -ENXIO; } if (cs46xx_dsp_pcm_channel_set_period (chip,cpcm->pcm_channel,period_size)) { - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return -EINVAL; } @@ -965,7 +967,7 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, } if ((err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params))) < 0) { #ifdef CONFIG_SND_CS46XX_NEW_DSP - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); #endif return err; } @@ -989,7 +991,7 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, } #ifdef CONFIG_SND_CS46XX_NEW_DSP - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); #endif return 0; @@ -1319,7 +1321,7 @@ static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,in cpcm->substream = substream; #ifdef CONFIG_SND_CS46XX_NEW_DSP - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); cpcm->pcm_channel = NULL; cpcm->pcm_channel_id = pcm_channel_id; @@ -1328,7 +1330,7 @@ static int _cs46xx_playback_open_channel (struct snd_pcm_substream *substream,in SNDRV_PCM_HW_PARAM_PERIOD_BYTES, &hw_constraints_period_sizes); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); #else chip->playback_pcm = cpcm; /* HACK */ #endif @@ -1367,9 +1369,9 @@ static int snd_cs46xx_playback_open_iec958(struct snd_pcm_substream *substream) snd_printdd("open raw iec958 channel\n"); - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); cs46xx_iec958_pre_open (chip); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return _cs46xx_playback_open_channel(substream,DSP_IEC958_CHANNEL); } @@ -1385,9 +1387,9 @@ static int snd_cs46xx_playback_close_iec958(struct snd_pcm_substream *substream) err = snd_cs46xx_playback_close(substream); - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); cs46xx_iec958_post_close (chip); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return err; } @@ -1428,12 +1430,12 @@ static int snd_cs46xx_playback_close(struct snd_pcm_substream *substream) if (!cpcm) return -ENXIO; #ifdef CONFIG_SND_CS46XX_NEW_DSP - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); if (cpcm->pcm_channel) { cs46xx_dsp_destroy_pcm_channel(chip,cpcm->pcm_channel); cpcm->pcm_channel = NULL; } - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); #else chip->playback_pcm = NULL; #endif @@ -1848,7 +1850,7 @@ static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, switch (kcontrol->private_value) { case CS46XX_MIXER_SPDIF_OUTPUT_ELEMENT: - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); change = (chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED); if (ucontrol->value.integer.value[0] && !change) cs46xx_dsp_enable_spdif_out(chip); @@ -1856,7 +1858,7 @@ static int snd_cs46xx_iec958_put(struct snd_kcontrol *kcontrol, cs46xx_dsp_disable_spdif_out(chip); res = (change != (chip->dsp_spos_instance->spdif_status_out & DSP_SPDIF_STATUS_OUTPUT_ENABLED)); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); break; case CS46XX_MIXER_SPDIF_INPUT_ELEMENT: change = chip->dsp_spos_instance->spdif_status_in; @@ -1997,12 +1999,12 @@ static int snd_cs46xx_spdif_default_get(struct snd_kcontrol *kcontrol, struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); struct dsp_spos_instance * ins = chip->dsp_spos_instance; - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_default >> 24) & 0xff); ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_default >> 16) & 0xff); ucontrol->value.iec958.status[2] = 0; ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_default) & 0xff); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -2015,7 +2017,7 @@ static int snd_cs46xx_spdif_default_put(struct snd_kcontrol *kcontrol, unsigned int val; int change; - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); val = ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[0]) << 24) | ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[2]) << 16) | ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[3])) | @@ -2029,7 +2031,7 @@ static int snd_cs46xx_spdif_default_put(struct snd_kcontrol *kcontrol, if ( !(ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) ) cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return change; } @@ -2050,12 +2052,12 @@ static int snd_cs46xx_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); struct dsp_spos_instance * ins = chip->dsp_spos_instance; - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); ucontrol->value.iec958.status[0] = _wrap_all_bits((ins->spdif_csuv_stream >> 24) & 0xff); ucontrol->value.iec958.status[1] = _wrap_all_bits((ins->spdif_csuv_stream >> 16) & 0xff); ucontrol->value.iec958.status[2] = 0; ucontrol->value.iec958.status[3] = _wrap_all_bits((ins->spdif_csuv_stream) & 0xff); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -2068,7 +2070,7 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, unsigned int val; int change; - down (&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); val = ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[0]) << 24) | ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[1]) << 16) | ((unsigned int)_wrap_all_bits(ucontrol->value.iec958.status[3])) | @@ -2082,7 +2084,7 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, if ( ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN ) cs46xx_poke_via_dsp (chip,SP_SPDOUT_CSUV,val); - up (&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return change; } @@ -3755,7 +3757,7 @@ int __devinit snd_cs46xx_create(struct snd_card *card, } spin_lock_init(&chip->reg_lock); #ifdef CONFIG_SND_CS46XX_NEW_DSP - init_MUTEX(&chip->spos_mutex); + mutex_init(&chip->spos_mutex); #endif chip->card = card; chip->pci = pci; diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 445a448949e738..8726a68051e793 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -28,6 +28,8 @@ #include #include #include +#include + #include #include #include @@ -287,7 +289,7 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) snd_assert(ins != NULL, return); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); for (i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) continue; @@ -298,7 +300,7 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) vfree(ins->symbol_table.symbols); kfree(ins->modules); kfree(ins); - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); } int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module) @@ -497,7 +499,7 @@ static void cs46xx_dsp_proc_modules_read (struct snd_info_entry *entry, struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i,j; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); snd_iprintf(buffer, "MODULES:\n"); for ( i = 0; i < ins->nmodules; ++i ) { snd_iprintf(buffer, "\n%s:\n", ins->modules[i].module_name); @@ -510,7 +512,7 @@ static void cs46xx_dsp_proc_modules_read (struct snd_info_entry *entry, desc->segment_type,desc->offset, desc->size); } } - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); } static void cs46xx_dsp_proc_task_tree_read (struct snd_info_entry *entry, @@ -521,7 +523,7 @@ static void cs46xx_dsp_proc_task_tree_read (struct snd_info_entry *entry, int i, j, col; void __iomem *dst = chip->region.idx[1].remap_addr + DSP_PARAMETER_BYTE_OFFSET; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); snd_iprintf(buffer, "TASK TREES:\n"); for ( i = 0; i < ins->ntask; ++i) { snd_iprintf(buffer,"\n%04x %s:\n",ins->tasks[i].address,ins->tasks[i].task_name); @@ -538,7 +540,7 @@ static void cs46xx_dsp_proc_task_tree_read (struct snd_info_entry *entry, } snd_iprintf(buffer,"\n"); - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); } static void cs46xx_dsp_proc_scb_read (struct snd_info_entry *entry, @@ -548,7 +550,7 @@ static void cs46xx_dsp_proc_scb_read (struct snd_info_entry *entry, struct dsp_spos_instance * ins = chip->dsp_spos_instance; int i; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); snd_iprintf(buffer, "SCB's:\n"); for ( i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) @@ -571,7 +573,7 @@ static void cs46xx_dsp_proc_scb_read (struct snd_info_entry *entry, } snd_iprintf(buffer,"\n"); - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); } static void cs46xx_dsp_proc_parameter_dump_read (struct snd_info_entry *entry, @@ -852,14 +854,14 @@ int cs46xx_dsp_proc_init (struct snd_card *card, struct snd_cs46xx *chip) } ins->proc_scb_info_entry = entry; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); /* register/update SCB's entries on proc */ for (i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_register_scb_desc (chip, (ins->scbs + i)); } - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -899,12 +901,12 @@ int cs46xx_dsp_proc_done (struct snd_cs46xx *chip) ins->proc_task_info_entry = NULL; } - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); for (i = 0; i < ins->nscb; ++i) { if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); } - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); if (ins->proc_dsp_dir) { snd_info_unregister (ins->proc_dsp_dir); @@ -1694,7 +1696,7 @@ int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip) snd_assert (ins->asynch_rx_scb == NULL,return -EINVAL); snd_assert (ins->spdif_in_src != NULL,return -EINVAL); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); if ( ! (ins->spdif_status_out & DSP_SPDIF_STATUS_INPUT_CTRL_ENABLED) ) { /* time countdown enable */ @@ -1738,7 +1740,7 @@ int cs46xx_dsp_enable_spdif_in (struct snd_cs46xx *chip) /* monitor state */ ins->spdif_status_in = 1; - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -1750,7 +1752,7 @@ int cs46xx_dsp_disable_spdif_in (struct snd_cs46xx *chip) snd_assert (ins->asynch_rx_scb != NULL, return -EINVAL); snd_assert (ins->spdif_in_src != NULL,return -EINVAL); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); /* Remove the asynchronous receiver SCB */ cs46xx_dsp_remove_scb (chip,ins->asynch_rx_scb); @@ -1760,7 +1762,7 @@ int cs46xx_dsp_disable_spdif_in (struct snd_cs46xx *chip) /* monitor state */ ins->spdif_status_in = 0; - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); /* restore amplifier */ chip->active_ctrl(chip, -1); @@ -1776,10 +1778,10 @@ int cs46xx_dsp_enable_pcm_capture (struct snd_cs46xx *chip) snd_assert (ins->pcm_input == NULL,return -EINVAL); snd_assert (ins->ref_snoop_scb != NULL,return -EINVAL); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); ins->pcm_input = cs46xx_add_record_source(chip,ins->ref_snoop_scb,PCMSERIALIN_PCM_SCB_ADDR, "PCMSerialInput_Wave"); - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -1790,10 +1792,10 @@ int cs46xx_dsp_disable_pcm_capture (struct snd_cs46xx *chip) snd_assert (ins->pcm_input != NULL,return -EINVAL); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); cs46xx_dsp_remove_scb (chip,ins->pcm_input); ins->pcm_input = NULL; - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -1805,10 +1807,10 @@ int cs46xx_dsp_enable_adc_capture (struct snd_cs46xx *chip) snd_assert (ins->adc_input == NULL,return -EINVAL); snd_assert (ins->codec_in_scb != NULL,return -EINVAL); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); ins->adc_input = cs46xx_add_record_source(chip,ins->codec_in_scb,PCMSERIALIN_SCB_ADDR, "PCMSerialInput_ADC"); - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -1819,10 +1821,10 @@ int cs46xx_dsp_disable_adc_capture (struct snd_cs46xx *chip) snd_assert (ins->adc_input != NULL,return -EINVAL); - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); cs46xx_dsp_remove_scb (chip,ins->adc_input); ins->adc_input = NULL; - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -1869,7 +1871,7 @@ int cs46xx_dsp_set_dac_volume (struct snd_cs46xx * chip, u16 left, u16 right) struct dsp_spos_instance * ins = chip->dsp_spos_instance; struct dsp_scb_descriptor * scb; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); /* main output */ scb = ins->master_mix_scb->sub_list_ptr; @@ -1888,7 +1890,7 @@ int cs46xx_dsp_set_dac_volume (struct snd_cs46xx * chip, u16 left, u16 right) ins->dac_volume_left = left; ins->dac_volume_right = right; - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } @@ -1897,7 +1899,7 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); if (ins->asynch_rx_scb != NULL) cs46xx_dsp_scb_set_volume (chip,ins->asynch_rx_scb, @@ -1906,7 +1908,7 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) ins->spdif_input_volume_left = left; ins->spdif_input_volume_right = right; - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); return 0; } diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index d4e0fb39bd069f..2c4ee45fe10c3f 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -28,6 +28,8 @@ #include #include #include +#include + #include #include #include @@ -77,7 +79,7 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, ins = chip->dsp_spos_instance; - down(&chip->spos_mutex); + mutex_lock(&chip->spos_mutex); snd_iprintf(buffer,"%04x %s:\n",scb->address,scb->scb_name); for (col = 0,j = 0;j < 0x10; j++,col++) { @@ -105,7 +107,7 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, scb->task_entry->address); snd_iprintf(buffer,"index [%d] ref_count [%d]\n",scb->index,scb->ref_count); - up(&chip->spos_mutex); + mutex_unlock(&chip->spos_mutex); } #endif diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 3c7043b7d4c92a..103a3f7708b786 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -36,6 +36,8 @@ #include #include #include +#include + #include #include @@ -1097,8 +1099,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, spin_lock_init(&emu->voice_lock); spin_lock_init(&emu->synth_lock); spin_lock_init(&emu->memblk_lock); - init_MUTEX(&emu->ptb_lock); - init_MUTEX(&emu->fx8010.lock); + mutex_init(&emu->fx8010.lock); INIT_LIST_HEAD(&emu->mapped_link_head); INIT_LIST_HEAD(&emu->mapped_order_link_head); emu->pci = pci; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 50983725273516..dfba00230d4dac 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -32,6 +32,8 @@ #include #include #include +#include + #include #include @@ -874,7 +876,7 @@ static int snd_emu10k1_icode_poke(struct snd_emu10k1 *emu, { int err = 0; - down(&emu->fx8010.lock); + mutex_lock(&emu->fx8010.lock); if ((err = snd_emu10k1_verify_controls(emu, icode)) < 0) goto __error; strlcpy(emu->fx8010.name, icode->name, sizeof(emu->fx8010.name)); @@ -897,7 +899,7 @@ static int snd_emu10k1_icode_poke(struct snd_emu10k1 *emu, else snd_emu10k1_ptr_write(emu, DBG, 0, emu->fx8010.dbg); __error: - up(&emu->fx8010.lock); + mutex_unlock(&emu->fx8010.lock); return err; } @@ -906,7 +908,7 @@ static int snd_emu10k1_icode_peek(struct snd_emu10k1 *emu, { int err; - down(&emu->fx8010.lock); + mutex_lock(&emu->fx8010.lock); strlcpy(icode->name, emu->fx8010.name, sizeof(icode->name)); /* ok, do the main job */ err = snd_emu10k1_gpr_peek(emu, icode); @@ -916,7 +918,7 @@ static int snd_emu10k1_icode_peek(struct snd_emu10k1 *emu, err = snd_emu10k1_code_peek(emu, icode); if (err >= 0) err = snd_emu10k1_list_controls(emu, icode); - up(&emu->fx8010.lock); + mutex_unlock(&emu->fx8010.lock); return err; } @@ -932,7 +934,7 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, if (ipcm->channels > 32) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; - down(&emu->fx8010.lock); + mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); if (pcm->opened) { err = -EBUSY; @@ -962,7 +964,7 @@ static int snd_emu10k1_ipcm_poke(struct snd_emu10k1 *emu, } __error: spin_unlock_irq(&emu->reg_lock); - up(&emu->fx8010.lock); + mutex_unlock(&emu->fx8010.lock); return err; } @@ -976,7 +978,7 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, if (ipcm->substream >= EMU10K1_FX8010_PCM_COUNT) return -EINVAL; pcm = &emu->fx8010.pcm[ipcm->substream]; - down(&emu->fx8010.lock); + mutex_lock(&emu->fx8010.lock); spin_lock_irq(&emu->reg_lock); ipcm->channels = pcm->channels; ipcm->tram_start = pcm->tram_start; @@ -992,7 +994,7 @@ static int snd_emu10k1_ipcm_peek(struct snd_emu10k1 *emu, ipcm->res1 = ipcm->res2 = 0; ipcm->pad = 0; spin_unlock_irq(&emu->reg_lock); - up(&emu->fx8010.lock); + mutex_unlock(&emu->fx8010.lock); return err; } @@ -2308,9 +2310,9 @@ static int snd_emu10k1_fx8010_ioctl(struct snd_hwdep * hw, struct file *file, un return -EPERM; if (get_user(addr, (unsigned int __user *)argp)) return -EFAULT; - down(&emu->fx8010.lock); + mutex_lock(&emu->fx8010.lock); res = snd_emu10k1_fx8010_tram_setup(emu, addr); - up(&emu->fx8010.lock); + mutex_unlock(&emu->fx8010.lock); return res; case SNDRV_EMU10K1_IOCTL_STOP: if (!capable(CAP_SYS_ADMIN)) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 68c795c03109cb..e7ec98649f044f 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -24,6 +24,8 @@ #include #include #include +#include + #include #include @@ -302,10 +304,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst hdr = emu->memhdr; snd_assert(hdr, return NULL); - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); blk = search_empty(emu, runtime->dma_bytes); if (blk == NULL) { - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } /* fill buffer addresses but pointers are not stored so that @@ -318,14 +320,14 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst if (idx >= sgbuf->pages) { printk(KERN_ERR "emu: pages overflow! (%d-%d) for %d\n", blk->first_page, blk->last_page, sgbuf->pages); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } #endif addr = sgbuf->table[idx].addr; if (! is_valid_page(emu, addr)) { printk(KERN_ERR "emu: failure page = %d\n", idx); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } emu->page_addr_table[page] = addr; @@ -337,10 +339,10 @@ snd_emu10k1_alloc_pages(struct snd_emu10k1 *emu, struct snd_pcm_substream *subst err = snd_emu10k1_memblk_map(emu, blk); if (err < 0) { __snd_util_mem_free(hdr, (struct snd_util_memblk *)blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return (struct snd_util_memblk *)blk; } @@ -369,19 +371,19 @@ snd_emu10k1_synth_alloc(struct snd_emu10k1 *hw, unsigned int size) struct snd_emu10k1_memblk *blk; struct snd_util_memhdr *hdr = hw->memhdr; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); blk = (struct snd_emu10k1_memblk *)__snd_util_mem_alloc(hdr, size); if (blk == NULL) { - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } if (synth_alloc_pages(hw, blk)) { __snd_util_mem_free(hdr, (struct snd_util_memblk *)blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } snd_emu10k1_memblk_map(hw, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return (struct snd_util_memblk *)blk; } @@ -396,14 +398,14 @@ snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk) struct snd_emu10k1_memblk *blk = (struct snd_emu10k1_memblk *)memblk; unsigned long flags; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); spin_lock_irqsave(&emu->memblk_lock, flags); if (blk->mapped_page >= 0) unmap_memblk(emu, blk); spin_unlock_irqrestore(&emu->memblk_lock, flags); synth_free_pages(emu, blk); __snd_util_mem_free(hdr, memblk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return 0; } diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index bee382995fe995..a5533c86b0b69e 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -35,6 +35,8 @@ #include #include #include +#include + #include #include #include @@ -379,7 +381,7 @@ MODULE_PARM_DESC(lineio, "Line In to Rear Out (0 = auto, 1 = force)."); struct ensoniq { spinlock_t reg_lock; - struct semaphore src_mutex; + struct mutex src_mutex; int irq; @@ -609,7 +611,7 @@ static void snd_es1371_codec_write(struct snd_ac97 *ac97, struct ensoniq *ensoniq = ac97->private_data; unsigned int t, x; - down(&ensoniq->src_mutex); + mutex_lock(&ensoniq->src_mutex); for (t = 0; t < POLL_COUNT; t++) { if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) { /* save the current state for latter */ @@ -634,11 +636,11 @@ static void snd_es1371_codec_write(struct snd_ac97 *ac97, /* restore SRC reg */ snd_es1371_wait_src_ready(ensoniq); outl(x, ES_REG(ensoniq, 1371_SMPRATE)); - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); return; } } - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); snd_printk(KERN_ERR "codec write timeout at 0x%lx [0x%x]\n", ES_REG(ensoniq, 1371_CODEC), inl(ES_REG(ensoniq, 1371_CODEC))); } @@ -650,7 +652,7 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, unsigned int t, x, fail = 0; __again: - down(&ensoniq->src_mutex); + mutex_lock(&ensoniq->src_mutex); for (t = 0; t < POLL_COUNT; t++) { if (!(inl(ES_REG(ensoniq, 1371_CODEC)) & ES_1371_CODEC_WIP)) { /* save the current state for latter */ @@ -683,11 +685,11 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, /* now wait for the stinkin' data (RDY) */ for (t = 0; t < POLL_COUNT; t++) { if ((x = inl(ES_REG(ensoniq, 1371_CODEC))) & ES_1371_CODEC_RDY) { - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); return ES_1371_CODEC_READ(x); } } - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); if (++fail > 10) { snd_printk(KERN_ERR "codec read timeout (final) " "at 0x%lx, reg = 0x%x [0x%x]\n", @@ -698,7 +700,7 @@ static unsigned short snd_es1371_codec_read(struct snd_ac97 *ac97, goto __again; } } - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); snd_printk(KERN_ERR "es1371: codec read timeout at 0x%lx [0x%x]\n", ES_REG(ensoniq, 1371_CODEC), inl(ES_REG(ensoniq, 1371_CODEC))); return 0; @@ -717,7 +719,7 @@ static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) { unsigned int n, truncm, freq, result; - down(&ensoniq->src_mutex); + mutex_lock(&ensoniq->src_mutex); n = rate / 3000; if ((1 << n) & ((1 << 15) | (1 << 13) | (1 << 11) | (1 << 9))) n--; @@ -742,14 +744,14 @@ static void snd_es1371_adc_rate(struct ensoniq * ensoniq, unsigned int rate) snd_es1371_src_write(ensoniq, ES_SMPREG_ADC + ES_SMPREG_VFREQ_FRAC, freq & 0x7fff); snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC, n << 8); snd_es1371_src_write(ensoniq, ES_SMPREG_VOL_ADC + 1, n << 8); - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); } static void snd_es1371_dac1_rate(struct ensoniq * ensoniq, unsigned int rate) { unsigned int freq, r; - down(&ensoniq->src_mutex); + mutex_lock(&ensoniq->src_mutex); freq = ((rate << 15) + 1500) / 3000; r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | ES_1371_DIS_P2 | ES_1371_DIS_R1)) | @@ -763,14 +765,14 @@ static void snd_es1371_dac1_rate(struct ensoniq * ensoniq, unsigned int rate) r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | ES_1371_DIS_P2 | ES_1371_DIS_R1)); outl(r, ES_REG(ensoniq, 1371_SMPRATE)); - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); } static void snd_es1371_dac2_rate(struct ensoniq * ensoniq, unsigned int rate) { unsigned int freq, r; - down(&ensoniq->src_mutex); + mutex_lock(&ensoniq->src_mutex); freq = ((rate << 15) + 1500) / 3000; r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | ES_1371_DIS_R1)) | @@ -785,7 +787,7 @@ static void snd_es1371_dac2_rate(struct ensoniq * ensoniq, unsigned int rate) r = (snd_es1371_wait_src_ready(ensoniq) & (ES_1371_SRC_DISABLE | ES_1371_DIS_P1 | ES_1371_DIS_R1)); outl(r, ES_REG(ensoniq, 1371_SMPRATE)); - up(&ensoniq->src_mutex); + mutex_unlock(&ensoniq->src_mutex); } #endif /* CHIP1371 */ @@ -2123,7 +2125,7 @@ static int __devinit snd_ensoniq_create(struct snd_card *card, return -ENOMEM; } spin_lock_init(&ensoniq->reg_lock); - init_MUTEX(&ensoniq->src_mutex); + mutex_init(&ensoniq->src_mutex); ensoniq->card = card; ensoniq->pci = pci; ensoniq->irq = -1; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 3747a436f0cd20..6a265ab3894ee9 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -103,6 +103,8 @@ #include #include #include +#include + #include #include #include @@ -569,7 +571,7 @@ struct es1968 { u16 maestro_map[32]; int bobclient; /* active timer instancs */ int bob_freq; /* timer frequency */ - struct semaphore memory_mutex; /* memory lock */ + struct mutex memory_mutex; /* memory lock */ /* APU states */ unsigned char apu[NR_APUS]; @@ -1356,13 +1358,13 @@ static int calc_available_memory_size(struct es1968 *chip) struct list_head *p; int max_size = 0; - down(&chip->memory_mutex); + mutex_lock(&chip->memory_mutex); list_for_each(p, &chip->buf_list) { struct esm_memory *buf = list_entry(p, struct esm_memory, list); if (buf->empty && buf->buf.bytes > max_size) max_size = buf->buf.bytes; } - up(&chip->memory_mutex); + mutex_unlock(&chip->memory_mutex); if (max_size >= 128*1024) max_size = 127*1024; return max_size; @@ -1375,20 +1377,20 @@ static struct esm_memory *snd_es1968_new_memory(struct es1968 *chip, int size) struct list_head *p; size = ((size + ESM_MEM_ALIGN - 1) / ESM_MEM_ALIGN) * ESM_MEM_ALIGN; - down(&chip->memory_mutex); + mutex_lock(&chip->memory_mutex); list_for_each(p, &chip->buf_list) { buf = list_entry(p, struct esm_memory, list); if (buf->empty && buf->buf.bytes >= size) goto __found; } - up(&chip->memory_mutex); + mutex_unlock(&chip->memory_mutex); return NULL; __found: if (buf->buf.bytes > size) { struct esm_memory *chunk = kmalloc(sizeof(*chunk), GFP_KERNEL); if (chunk == NULL) { - up(&chip->memory_mutex); + mutex_unlock(&chip->memory_mutex); return NULL; } chunk->buf = buf->buf; @@ -1400,7 +1402,7 @@ __found: list_add(&chunk->list, &buf->list); } buf->empty = 0; - up(&chip->memory_mutex); + mutex_unlock(&chip->memory_mutex); return buf; } @@ -1409,7 +1411,7 @@ static void snd_es1968_free_memory(struct es1968 *chip, struct esm_memory *buf) { struct esm_memory *chunk; - down(&chip->memory_mutex); + mutex_lock(&chip->memory_mutex); buf->empty = 1; if (buf->list.prev != &chip->buf_list) { chunk = list_entry(buf->list.prev, struct esm_memory, list); @@ -1428,7 +1430,7 @@ static void snd_es1968_free_memory(struct es1968 *chip, struct esm_memory *buf) kfree(chunk); } } - up(&chip->memory_mutex); + mutex_unlock(&chip->memory_mutex); } static void snd_es1968_free_dmabuf(struct es1968 *chip) @@ -2579,7 +2581,7 @@ static int __devinit snd_es1968_create(struct snd_card *card, INIT_LIST_HEAD(&chip->buf_list); INIT_LIST_HEAD(&chip->substream_list); spin_lock_init(&chip->ac97_lock); - init_MUTEX(&chip->memory_mutex); + mutex_init(&chip->memory_mutex); tasklet_init(&chip->hwvol_tq, es1968_update_hw_volume, (unsigned long)chip); chip->card = card; chip->pci = pci; diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 4a6dd97deba61e..208a3341ec201b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include "hda_codec.h" #include @@ -76,12 +77,12 @@ unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int dire unsigned int verb, unsigned int parm) { unsigned int res; - down(&codec->bus->cmd_mutex); + mutex_lock(&codec->bus->cmd_mutex); if (! codec->bus->ops.command(codec, nid, direct, verb, parm)) res = codec->bus->ops.get_response(codec); else res = (unsigned int)-1; - up(&codec->bus->cmd_mutex); + mutex_unlock(&codec->bus->cmd_mutex); return res; } @@ -101,9 +102,9 @@ int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct, unsigned int verb, unsigned int parm) { int err; - down(&codec->bus->cmd_mutex); + mutex_lock(&codec->bus->cmd_mutex); err = codec->bus->ops.command(codec, nid, direct, verb, parm); - up(&codec->bus->cmd_mutex); + mutex_unlock(&codec->bus->cmd_mutex); return err; } @@ -371,7 +372,7 @@ int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp, bus->modelname = temp->modelname; bus->ops = temp->ops; - init_MUTEX(&bus->cmd_mutex); + mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops)) < 0) { @@ -523,7 +524,7 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, codec->bus = bus; codec->addr = codec_addr; - init_MUTEX(&codec->spdif_mutex); + mutex_init(&codec->spdif_mutex); init_amp_hash(codec); list_add_tail(&codec->list, &bus->codec_list); @@ -881,12 +882,12 @@ int snd_hda_mixer_bind_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ unsigned long pval; int err; - down(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ pval = kcontrol->private_value; kcontrol->private_value = pval & ~AMP_VAL_IDX_MASK; /* index 0 */ err = snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); kcontrol->private_value = pval; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return err; } @@ -896,7 +897,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ unsigned long pval; int i, indices, err = 0, change = 0; - down(&codec->spdif_mutex); /* reuse spdif_mutex */ + mutex_lock(&codec->spdif_mutex); /* reuse spdif_mutex */ pval = kcontrol->private_value; indices = (pval & AMP_VAL_IDX_MASK) >> AMP_VAL_IDX_SHIFT; for (i = 0; i < indices; i++) { @@ -907,7 +908,7 @@ int snd_hda_mixer_bind_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ change |= err; } kcontrol->private_value = pval; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return err < 0 ? err : change; } @@ -1011,7 +1012,7 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_c unsigned short val; int change; - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); codec->spdif_status = ucontrol->value.iec958.status[0] | ((unsigned int)ucontrol->value.iec958.status[1] << 8) | ((unsigned int)ucontrol->value.iec958.status[2] << 16) | @@ -1026,7 +1027,7 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_c snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8); } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return change; } @@ -1054,7 +1055,7 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, struct sn unsigned short val; int change; - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); val = codec->spdif_ctls & ~1; if (ucontrol->value.integer.value[0]) val |= 1; @@ -1066,7 +1067,7 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, struct sn AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return change; } @@ -1150,13 +1151,13 @@ static int snd_hda_spdif_in_switch_put(struct snd_kcontrol *kcontrol, struct snd unsigned int val = !!ucontrol->value.integer.value[0]; int change; - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); change = codec->spdif_in_enable != val; if (change || codec->in_resume) { codec->spdif_in_enable = val; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val); } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return change; } @@ -1824,13 +1825,13 @@ int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *i */ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout) { - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); if (mout->dig_out_used) { - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return -EBUSY; /* already being used */ } mout->dig_out_used = HDA_DIG_EXCLUSIVE; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -1839,9 +1840,9 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mo */ int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout) { - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); mout->dig_out_used = 0; - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return 0; } @@ -1869,7 +1870,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o int chs = substream->runtime->channels; int i; - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { if (chs == 2 && snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && @@ -1883,7 +1884,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); } } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); /* front */ snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format); @@ -1914,12 +1915,12 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_o snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0); if (mout->hp_nid) snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); - down(&codec->spdif_mutex); + mutex_lock(&codec->spdif_mutex); if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) { snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); mout->dig_out_used = 0; } - up(&codec->spdif_mutex); + mutex_unlock(&codec->spdif_mutex); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 63e26c7a2b7a07..40520e9d5a4b2b 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -438,7 +438,7 @@ struct hda_bus { struct list_head codec_list; struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS + 1]; /* caddr -> codec */ - struct semaphore cmd_mutex; + struct mutex cmd_mutex; /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; @@ -559,7 +559,7 @@ struct hda_codec { int amp_info_size; struct hda_amp_info *amp_info; - struct semaphore spdif_mutex; + struct mutex spdif_mutex; unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b3f37e7b33c047..dbed2644a1920b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -43,6 +43,7 @@ #include #include #include +#include #include #include #include "hda_codec.h" @@ -297,7 +298,7 @@ struct azx { /* locks */ spinlock_t reg_lock; - struct semaphore open_mutex; + struct mutex open_mutex; /* streams (x num_streams) */ struct azx_dev *azx_dev; @@ -993,10 +994,10 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) unsigned long flags; int err; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); azx_dev = azx_assign_device(chip, substream->stream); if (azx_dev == NULL) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return -EBUSY; } runtime->hw = azx_pcm_hw; @@ -1008,7 +1009,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if ((err = hinfo->ops.open(hinfo, apcm->codec, substream)) < 0) { azx_release_device(azx_dev); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return err; } spin_lock_irqsave(&chip->reg_lock, flags); @@ -1017,7 +1018,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) spin_unlock_irqrestore(&chip->reg_lock, flags); runtime->private_data = azx_dev; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } @@ -1029,14 +1030,14 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = NULL; azx_dev->running = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); azx_release_device(azx_dev); hinfo->ops.close(hinfo, apcm->codec, substream); - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } @@ -1408,7 +1409,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, } spin_lock_init(&chip->reg_lock); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->pci = pci; chip->irq = -1; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1ada1b075c9a9c..5a3821ae93a8b7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -23,6 +23,8 @@ #include #include #include +#include + #include #include "hda_codec.h" #include "hda_local.h" @@ -60,7 +62,7 @@ struct ad198x_spec { /* PCM information */ struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ - struct semaphore amp_mutex; /* PCM volume/mute control mutex */ + struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; /* dynamic controls, init_verbs and input_mux */ @@ -371,9 +373,9 @@ static int ad1986a_pcm_amp_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; - down(&ad->amp_mutex); + mutex_lock(&ad->amp_mutex); snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); - up(&ad->amp_mutex); + mutex_unlock(&ad->amp_mutex); return 0; } @@ -383,13 +385,13 @@ static int ad1986a_pcm_amp_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl struct ad198x_spec *ad = codec->spec; int i, change = 0; - down(&ad->amp_mutex); + mutex_lock(&ad->amp_mutex); for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) { kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT); change |= snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); } kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT); - up(&ad->amp_mutex); + mutex_unlock(&ad->amp_mutex); return change; } @@ -400,9 +402,9 @@ static int ad1986a_pcm_amp_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct ad198x_spec *ad = codec->spec; - down(&ad->amp_mutex); + mutex_lock(&ad->amp_mutex); snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); - up(&ad->amp_mutex); + mutex_unlock(&ad->amp_mutex); return 0; } @@ -412,13 +414,13 @@ static int ad1986a_pcm_amp_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct ad198x_spec *ad = codec->spec; int i, change = 0; - down(&ad->amp_mutex); + mutex_lock(&ad->amp_mutex); for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) { kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT); change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT); - up(&ad->amp_mutex); + mutex_unlock(&ad->amp_mutex); return change; } @@ -544,7 +546,7 @@ static int patch_ad1986a(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 6; @@ -708,7 +710,7 @@ static int patch_ad1983(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -854,7 +856,7 @@ static int patch_ad1981(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; spec->multiout.max_channels = 2; @@ -2032,7 +2034,7 @@ static int patch_ad1988(struct hda_codec *codec) if (spec == NULL) return -ENOMEM; - init_MUTEX(&spec->amp_mutex); + mutex_init(&spec->amp_mutex); codec->spec = spec; if (codec->revision_id == AD1988A_REV2) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 2175f6721347bb..0f7f4d8263c0eb 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -53,6 +53,8 @@ #include #include #include +#include + #include #include "ice1712.h" @@ -210,14 +212,14 @@ static int aureon_ac97_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ele struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short vol; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); vol = aureon_ac97_read(ice, kcontrol->private_value & 0x7F); ucontrol->value.integer.value[0] = 0x1F - (vol & 0x1F); if (kcontrol->private_value & AUREON_AC97_STEREO) ucontrol->value.integer.value[1] = 0x1F - ((vol >> 8) & 0x1F); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -252,11 +254,11 @@ static int aureon_ac97_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_el { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = aureon_ac97_read(ice, kcontrol->private_value & 0x7F) & 0x8000 ? 0 : 1; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -288,11 +290,11 @@ static int aureon_ac97_micboost_get(struct snd_kcontrol *kcontrol, struct snd_ct { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = aureon_ac97_read(ice, AC97_MIC) & 0x0020 ? 0 : 1; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -488,11 +490,11 @@ static int aureon_ac97_mmute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = (wm_get(ice, WM_OUT_MUX1) >> 1) & 0x01; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -557,9 +559,9 @@ static int wm_pcm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = (wm_get(ice, WM_MUTE) & 0x10) ? 0 : 1; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -782,11 +784,11 @@ static int wm_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short val; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); val = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff; val = val > PCM_MIN ? (val - PCM_MIN) : 0; ucontrol->value.integer.value[0] = val; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -827,12 +829,12 @@ static int wm_adc_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va unsigned short val; int i; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { val = wm_get(ice, WM_ADC_GAIN + i); ucontrol->value.integer.value[i] = ~val>>5 & 0x1; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -874,13 +876,13 @@ static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val int i, idx; unsigned short vol; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { idx = WM_ADC_GAIN + i; vol = wm_get(ice, idx) & 0x1f; ucontrol->value.integer.value[i] = vol; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -951,11 +953,11 @@ static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short val; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); val = wm_get(ice, WM_ADC_MUX); ucontrol->value.integer.value[0] = val & 7; ucontrol->value.integer.value[1] = (val >> 4) & 7; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 9a51d34e6817af..4bbf1e2ae65372 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -28,6 +28,8 @@ #include #include #include +#include + #include #include #include @@ -130,13 +132,13 @@ static int ap_cs8427_sendbytes(struct snd_i2c_device *device, unsigned char *byt int res = count; unsigned char tmp; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); tmp = ap_cs8427_codec_select(ice); ap_cs8427_write_byte(ice, (device->addr << 1) | 0, tmp); /* address + write mode */ while (count-- > 0) ap_cs8427_write_byte(ice, *bytes++, tmp); ap_cs8427_codec_deassert(ice, tmp); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return res; } @@ -147,13 +149,13 @@ static int ap_cs8427_readbytes(struct snd_i2c_device *device, unsigned char *byt int res = count; unsigned char tmp; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); tmp = ap_cs8427_codec_select(ice); ap_cs8427_write_byte(ice, (device->addr << 1) | 1, tmp); /* address + read mode */ while (count-- > 0) *bytes++ = ap_cs8427_read_byte(ice, tmp); ap_cs8427_codec_deassert(ice, tmp); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return res; } @@ -180,7 +182,7 @@ static void snd_ice1712_delta_cs8403_spdif_write(struct snd_ice1712 *ice, unsign /* send byte to transmitter */ mask1 = ICE1712_DELTA_SPDIF_OUT_STAT_CLOCK; mask2 = ICE1712_DELTA_SPDIF_OUT_STAT_DATA; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); for (idx = 7; idx >= 0; idx--) { tmp &= ~(mask1 | mask2); @@ -194,7 +196,7 @@ static void snd_ice1712_delta_cs8403_spdif_write(struct snd_ice1712 *ice, unsign } tmp &= ~mask1; snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } @@ -296,14 +298,14 @@ static void delta_1010_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) if (rate == 0) /* no hint - S/PDIF input is master, simply return */ return; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); tmp2 = tmp & ~ICE1712_DELTA_DFS; if (rate > 48000) tmp2 |= ICE1712_DELTA_DFS; if (tmp != tmp2) snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp2); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } /* @@ -318,9 +320,9 @@ static void delta_ak4524_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) return; /* check before reset ak4524 to avoid unnecessary clicks */ - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); tmp2 = tmp & ~ICE1712_DELTA_DFS; if (rate > 48000) tmp2 |= ICE1712_DELTA_DFS; @@ -329,12 +331,12 @@ static void delta_ak4524_set_rate_val(struct snd_akm4xxx *ak, unsigned int rate) /* do it again */ snd_akm4xxx_reset(ak, 1); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); tmp = snd_ice1712_read(ice, ICE1712_IREG_GPIO_DATA) & ~ICE1712_DELTA_DFS; if (rate > 48000) tmp |= ICE1712_DELTA_DFS; snd_ice1712_write(ice, ICE1712_IREG_GPIO_DATA, tmp); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); snd_akm4xxx_reset(ak, 0); } diff --git a/sound/pci/ice1712/hoontech.c b/sound/pci/ice1712/hoontech.c index 3f2f918536f53c..3f27d04e7d3c07 100644 --- a/sound/pci/ice1712/hoontech.c +++ b/sound/pci/ice1712/hoontech.c @@ -27,6 +27,8 @@ #include #include #include +#include + #include #include "ice1712.h" @@ -48,31 +50,31 @@ static void __devinit snd_ice1712_stdsp24_gpio_write(struct snd_ice1712 *ice, un static void __devinit snd_ice1712_stdsp24_darear(struct snd_ice1712 *ice, int activate) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ICE1712_STDSP24_0_DAREAR(ice->spec.hoontech.boxbits, activate); snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[0]); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_mute(struct snd_ice1712 *ice, int activate) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ICE1712_STDSP24_3_MUTE(ice->spec.hoontech.boxbits, activate); snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_insel(struct snd_ice1712 *ice, int activate) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ICE1712_STDSP24_3_INSEL(ice->spec.hoontech.boxbits, activate); snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_box_channel(struct snd_ice1712 *ice, int box, int chn, int activate) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); /* select box */ ICE1712_STDSP24_0_BOX(ice->spec.hoontech.boxbits, box); @@ -115,12 +117,12 @@ static void __devinit snd_ice1712_stdsp24_box_channel(struct snd_ice1712 *ice, i ICE1712_STDSP24_2_MIDI1(ice->spec.hoontech.boxbits, 0); snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_box_midi(struct snd_ice1712 *ice, int box, int master) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); /* select box */ ICE1712_STDSP24_0_BOX(ice->spec.hoontech.boxbits, box); @@ -141,15 +143,15 @@ static void __devinit snd_ice1712_stdsp24_box_midi(struct snd_ice1712 *ice, int ICE1712_STDSP24_2_MIDIIN(ice->spec.hoontech.boxbits, 1); snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[2]); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void __devinit snd_ice1712_stdsp24_midi2(struct snd_ice1712 *ice, int activate) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ICE1712_STDSP24_3_MIDI2(ice->spec.hoontech.boxbits, activate); snd_ice1712_stdsp24_gpio_write(ice, ice->spec.hoontech.boxbits[3]); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static int __devinit snd_ice1712_hoontech_init(struct snd_ice1712 *ice) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index ef6f18558c95e9..3156a31329903f 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -55,6 +55,7 @@ #include #include #include +#include #include #include #include @@ -2557,9 +2558,9 @@ static int __devinit snd_ice1712_create(struct snd_card *card, cs8427_timeout = 1000; ice->cs8427_timeout = cs8427_timeout; spin_lock_init(&ice->reg_lock); - init_MUTEX(&ice->gpio_mutex); - init_MUTEX(&ice->i2c_mutex); - init_MUTEX(&ice->open_mutex); + mutex_init(&ice->gpio_mutex); + mutex_init(&ice->i2c_mutex); + mutex_init(&ice->open_mutex); ice->gpio.set_mask = snd_ice1712_set_gpio_mask; ice->gpio.set_dir = snd_ice1712_set_gpio_dir; ice->gpio.set_data = snd_ice1712_set_gpio_data; diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index ce96b3bb653165..d7416a83fcacdf 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -334,7 +334,7 @@ struct snd_ice1712 { unsigned int num_total_adcs; /* total ADCs */ unsigned int cur_rate; /* current rate */ - struct semaphore open_mutex; + struct mutex open_mutex; struct snd_pcm_substream *pcm_reserved[4]; struct snd_pcm_hw_constraint_list *hw_rates; /* card-specific rate constraints */ @@ -342,7 +342,7 @@ struct snd_ice1712 { struct snd_akm4xxx *akm; struct snd_ice1712_spdif spdif; - struct semaphore i2c_mutex; /* I2C mutex for ICE1724 registers */ + struct mutex i2c_mutex; /* I2C mutex for ICE1724 registers */ struct snd_i2c_bus *i2c; /* I2C bus */ struct snd_i2c_device *cs8427; /* CS8427 I2C device */ unsigned int cs8427_timeout; /* CS8427 reset timeout in HZ/100 */ @@ -360,7 +360,7 @@ struct snd_ice1712 { void (*set_pro_rate)(struct snd_ice1712 *ice, unsigned int rate); void (*i2s_mclk_changed)(struct snd_ice1712 *ice); } gpio; - struct semaphore gpio_mutex; + struct mutex gpio_mutex; /* other board-specific data */ union { @@ -423,7 +423,7 @@ static inline unsigned int snd_ice1712_gpio_read(struct snd_ice1712 *ice) */ static inline void snd_ice1712_save_gpio_status(struct snd_ice1712 *ice) { - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ice->gpio.saved[0] = ice->gpio.direction; ice->gpio.saved[1] = ice->gpio.write_mask; } @@ -434,7 +434,7 @@ static inline void snd_ice1712_restore_gpio_status(struct snd_ice1712 *ice) ice->gpio.set_mask(ice, ice->gpio.saved[1]); ice->gpio.direction = ice->gpio.saved[0]; ice->gpio.write_mask = ice->gpio.saved[1]; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } /* for bit controls */ diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 71f08c03601985..fce616c2761f2a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include @@ -487,7 +488,7 @@ static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream, int i, chs; chs = params_channels(hw_params); - down(&ice->open_mutex); + mutex_lock(&ice->open_mutex); /* mark surround channels */ if (substream == ice->playback_pro_substream) { /* PDMA0 can be multi-channel up to 8 */ @@ -495,7 +496,7 @@ static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream, for (i = 0; i < chs; i++) { if (ice->pcm_reserved[i] && ice->pcm_reserved[i] != substream) { - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); return -EBUSY; } ice->pcm_reserved[i] = substream; @@ -510,7 +511,7 @@ static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream, if (ice->playback_con_substream_ds[i] == substream) { if (ice->pcm_reserved[i] && ice->pcm_reserved[i] != substream) { - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); return -EBUSY; } ice->pcm_reserved[i] = substream; @@ -518,7 +519,7 @@ static int snd_vt1724_pcm_hw_params(struct snd_pcm_substream *substream, } } } - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); snd_vt1724_set_pro_rate(ice, params_rate(hw_params), 0); return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -528,12 +529,12 @@ static int snd_vt1724_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); int i; - down(&ice->open_mutex); + mutex_lock(&ice->open_mutex); /* unmark surround channels */ for (i = 0; i < 3; i++) if (ice->pcm_reserved[i] == substream) ice->pcm_reserved[i] = NULL; - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); return snd_pcm_lib_free_pages(substream); } @@ -778,7 +779,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); set_rate_constraints(ice, substream); - down(&ice->open_mutex); + mutex_lock(&ice->open_mutex); /* calculate the currently available channels */ for (chs = 0; chs < 3; chs++) { if (ice->pcm_reserved[chs]) @@ -788,7 +789,7 @@ static int snd_vt1724_playback_pro_open(struct snd_pcm_substream *substream) runtime->hw.channels_max = chs; if (chs > 2) /* channels must be even */ snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, 2); - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, VT1724_BUFFER_ALIGN); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, @@ -1128,13 +1129,13 @@ static int snd_vt1724_playback_indep_open(struct snd_pcm_substream *substream) struct snd_ice1712 *ice = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - down(&ice->open_mutex); + mutex_lock(&ice->open_mutex); /* already used by PDMA0? */ if (ice->pcm_reserved[substream->number]) { - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); return -EBUSY; /* FIXME: should handle blocking mode properly */ } - up(&ice->open_mutex); + mutex_unlock(&ice->open_mutex); runtime->private_data = &vt1724_playback_dma_regs[substream->number]; ice->playback_con_substream_ds[substream->number] = substream; runtime->hw = snd_vt1724_2ch_stereo; @@ -1978,12 +1979,12 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, { unsigned char val; - down(&ice->i2c_mutex); + mutex_lock(&ice->i2c_mutex); outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(dev & ~VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); wait_i2c_busy(ice); val = inb(ICEREG1724(ice, I2C_DATA)); - up(&ice->i2c_mutex); + mutex_unlock(&ice->i2c_mutex); //printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); return val; } @@ -1991,14 +1992,14 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, void snd_vt1724_write_i2c(struct snd_ice1712 *ice, unsigned char dev, unsigned char addr, unsigned char data) { - down(&ice->i2c_mutex); + mutex_lock(&ice->i2c_mutex); wait_i2c_busy(ice); //printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(data, ICEREG1724(ice, I2C_DATA)); outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); wait_i2c_busy(ice); - up(&ice->i2c_mutex); + mutex_unlock(&ice->i2c_mutex); } static int __devinit snd_vt1724_read_eeprom(struct snd_ice1712 *ice, @@ -2229,9 +2230,9 @@ static int __devinit snd_vt1724_create(struct snd_card *card, } ice->vt1724 = 1; spin_lock_init(&ice->reg_lock); - init_MUTEX(&ice->gpio_mutex); - init_MUTEX(&ice->open_mutex); - init_MUTEX(&ice->i2c_mutex); + mutex_init(&ice->gpio_mutex); + mutex_init(&ice->open_mutex); + mutex_init(&ice->i2c_mutex); ice->gpio.set_mask = snd_vt1724_set_gpio_mask; ice->gpio.set_dir = snd_vt1724_set_gpio_dir; ice->gpio.set_data = snd_vt1724_set_gpio_data; diff --git a/sound/pci/ice1712/phase.c b/sound/pci/ice1712/phase.c index ec3757834b93f2..502da1c8b5f719 100644 --- a/sound/pci/ice1712/phase.c +++ b/sound/pci/ice1712/phase.c @@ -39,6 +39,8 @@ #include #include #include +#include + #include #include "ice1712.h" @@ -273,9 +275,9 @@ static int wm_pcm_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = (wm_get(ice, WM_MUTE) & 0x10) ? 0 : 1; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -584,11 +586,11 @@ static int wm_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned short val; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); val = wm_get(ice, WM_DAC_DIG_MASTER_ATTEN) & 0xff; val = val > PCM_MIN ? (val - PCM_MIN) : 0; ucontrol->value.integer.value[0] = val; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index 0dccd7707a4b28..d23fb3fc21330f 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -27,6 +27,8 @@ #include #include #include +#include + #include #include @@ -124,13 +126,13 @@ static int wm_dac_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short val; int i; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { val = wm_get(ice, WM_DAC_ATTEN_L + i) & 0xff; val = val > DAC_MIN ? (val - DAC_MIN) : 0; ucontrol->value.integer.value[i] = val; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -140,7 +142,7 @@ static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short oval, nval; int i, idx, change = 0; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { nval = ucontrol->value.integer.value[i]; nval = (nval ? (nval + DAC_MIN) : 0) & 0xff; @@ -152,7 +154,7 @@ static int wm_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val change = 1; } } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return change; } @@ -179,13 +181,13 @@ static int wm_adc_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short val; int i; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { val = wm_get(ice, WM_ADC_ATTEN_L + i) & 0xff; val = val > ADC_MIN ? (val - ADC_MIN) : 0; ucontrol->value.integer.value[i] = val; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -195,7 +197,7 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short ovol, nvol; int i, idx, change = 0; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (i = 0; i < 2; i++) { nvol = ucontrol->value.integer.value[i]; nvol = nvol ? (nvol + ADC_MIN) : 0; @@ -206,7 +208,7 @@ static int wm_adc_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val change = 1; } } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return change; } @@ -227,9 +229,9 @@ static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); int bit = kcontrol->private_value; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = (wm_get(ice, WM_ADC_MUX) & (1 << bit)) ? 1 : 0; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -240,7 +242,7 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val unsigned short oval, nval; int change; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); nval = oval = wm_get(ice, WM_ADC_MUX); if (ucontrol->value.integer.value[0]) nval |= (1 << bit); @@ -250,7 +252,7 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val if (change) { wm_put(ice, WM_ADC_MUX, nval); } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -270,9 +272,9 @@ static int wm_bypass_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = (wm_get(ice, WM_OUT_MUX) & 0x04) ? 1 : 0; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -282,7 +284,7 @@ static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu unsigned short val, oval; int change = 0; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); val = oval = wm_get(ice, WM_OUT_MUX); if (ucontrol->value.integer.value[0]) val |= 0x04; @@ -292,7 +294,7 @@ static int wm_bypass_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu wm_put(ice, WM_OUT_MUX, val); change = 1; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return change; } @@ -312,9 +314,9 @@ static int wm_chswap_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.integer.value[0] = (wm_get(ice, WM_DAC_CTRL1) & 0xf0) != 0x90; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -324,7 +326,7 @@ static int wm_chswap_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu unsigned short val, oval; int change = 0; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); oval = wm_get(ice, WM_DAC_CTRL1); val = oval & 0x0f; if (ucontrol->value.integer.value[0]) @@ -336,7 +338,7 @@ static int wm_chswap_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu wm_put_nocache(ice, WM_DAC_CTRL1, val); change = 1; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return change; } @@ -449,9 +451,9 @@ static int cs_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); ucontrol->value.enumerated.item[0] = ice->gpio.saved[0]; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -461,14 +463,14 @@ static int cs_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_valu unsigned char val; int change = 0; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); if (ucontrol->value.enumerated.item[0] != ice->gpio.saved[0]) { ice->gpio.saved[0] = ucontrol->value.enumerated.item[0] & 3; val = 0x80 | (ice->gpio.saved[0] << 3); spi_write(ice, CS_DEV, 0x04, val); change = 1; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -488,10 +490,10 @@ static int pontis_gpio_mask_info(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int pontis_gpio_mask_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); /* 4-7 reserved */ ucontrol->value.integer.value[0] = (~ice->gpio.write_mask & 0xffff) | 0x00f0; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -500,22 +502,22 @@ static int pontis_gpio_mask_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int val; int changed; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); /* 4-7 reserved */ val = (~ucontrol->value.integer.value[0] & 0xffff) | 0x00f0; changed = val != ice->gpio.write_mask; ice->gpio.write_mask = val; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return changed; } static int pontis_gpio_dir_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); /* 4-7 reserved */ ucontrol->value.integer.value[0] = ice->gpio.direction & 0xff0f; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -524,23 +526,23 @@ static int pontis_gpio_dir_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int val; int changed; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); /* 4-7 reserved */ val = ucontrol->value.integer.value[0] & 0xff0f; changed = (val != ice->gpio.direction); ice->gpio.direction = val; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return changed; } static int pontis_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask); ucontrol->value.integer.value[0] = snd_ice1712_gpio_read(ice) & 0xffff; - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return 0; } @@ -549,7 +551,7 @@ static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int val, nval; int changed = 0; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); snd_ice1712_gpio_set_dir(ice, ice->gpio.direction); snd_ice1712_gpio_set_mask(ice, ice->gpio.write_mask); val = snd_ice1712_gpio_read(ice) & 0xffff; @@ -558,7 +560,7 @@ static int pontis_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el snd_ice1712_gpio_write(ice, nval); changed = 1; } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); return changed; } @@ -651,14 +653,14 @@ static void wm_proc_regs_write(struct snd_info_entry *entry, struct snd_info_buf struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; char line[64]; unsigned int reg, val; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); while (!snd_info_get_line(buffer, line, sizeof(line))) { if (sscanf(line, "%x %x", ®, &val) != 2) continue; if (reg <= 0x17 && val <= 0xffff) wm_put(ice, reg, val); } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) @@ -666,12 +668,12 @@ static void wm_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; int reg, val; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (reg = 0; reg <= 0x17; reg++) { val = wm_get(ice, reg); snd_iprintf(buffer, "%02x = %04x\n", reg, val); } - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void wm_proc_init(struct snd_ice1712 *ice) @@ -690,14 +692,14 @@ static void cs_proc_regs_read(struct snd_info_entry *entry, struct snd_info_buff struct snd_ice1712 *ice = (struct snd_ice1712 *)entry->private_data; int reg, val; - down(&ice->gpio_mutex); + mutex_lock(&ice->gpio_mutex); for (reg = 0; reg <= 0x26; reg++) { val = spi_read(ice, CS_DEV, reg); snd_iprintf(buffer, "%02x = %02x\n", reg, val); } val = spi_read(ice, CS_DEV, 0x7f); snd_iprintf(buffer, "%02x = %02x\n", 0x7f, val); - up(&ice->gpio_mutex); + mutex_unlock(&ice->gpio_mutex); } static void cs_proc_init(struct snd_ice1712 *ice) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 4eddb512c12fdb..4721c096335ec7 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include @@ -325,7 +326,7 @@ struct snd_korg1212 { int irq; spinlock_t lock; - struct semaphore open_mutex; + struct mutex open_mutex; struct timer_list timer; /* timer callback for checking ack of stop request */ int stop_pending_cnt; /* counter for stop pending check */ @@ -667,13 +668,13 @@ static int snd_korg1212_OpenCard(struct snd_korg1212 * korg1212) { K1212_DEBUG_PRINTK("K1212_DEBUG: OpenCard [%s] %d\n", stateName[korg1212->cardState], korg1212->opencnt); - down(&korg1212->open_mutex); + mutex_lock(&korg1212->open_mutex); if (korg1212->opencnt++ == 0) { snd_korg1212_TurnOffIdleMonitor(korg1212); snd_korg1212_setCardState(korg1212, K1212_STATE_OPEN); } - up(&korg1212->open_mutex); + mutex_unlock(&korg1212->open_mutex); return 1; } @@ -682,9 +683,9 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard [%s] %d\n", stateName[korg1212->cardState], korg1212->opencnt); - down(&korg1212->open_mutex); + mutex_lock(&korg1212->open_mutex); if (--(korg1212->opencnt)) { - up(&korg1212->open_mutex); + mutex_unlock(&korg1212->open_mutex); return 0; } @@ -695,7 +696,7 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) K1212_DEBUG_PRINTK("K1212_DEBUG: CloseCard - RC = %d [%s]\n", rc, stateName[korg1212->cardState]); if (rc != K1212_CMDRET_Success) { - up(&korg1212->open_mutex); + mutex_unlock(&korg1212->open_mutex); return 0; } } else if (korg1212->cardState > K1212_STATE_SETUP) { @@ -707,7 +708,7 @@ static int snd_korg1212_CloseCard(struct snd_korg1212 * korg1212) snd_korg1212_setCardState(korg1212, K1212_STATE_READY); } - up(&korg1212->open_mutex); + mutex_unlock(&korg1212->open_mutex); return 0; } @@ -2179,7 +2180,7 @@ static int __devinit snd_korg1212_create(struct snd_card *card, struct pci_dev * init_waitqueue_head(&korg1212->wait); spin_lock_init(&korg1212->lock); - init_MUTEX(&korg1212->open_mutex); + mutex_init(&korg1212->open_mutex); init_timer(&korg1212->timer); korg1212->timer.function = snd_korg1212_timer_func; korg1212->timer.data = (unsigned long)korg1212; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index b218e1d20c789b..e79fb264532b60 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -589,7 +590,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, /* set up format for the stream */ format = params_format(hw); - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); /* update the stream levels */ if( stream->pcm_number <= MIXART_PCM_DIGITAL ) { @@ -628,7 +629,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, bufferinfo[i].available_length, subs->number); } - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return err; } @@ -700,7 +701,7 @@ static int snd_mixart_playback_open(struct snd_pcm_substream *subs) int err = 0; int pcm_number; - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); if ( pcm == chip->pcm ) { pcm_number = MIXART_PCM_ANALOG; @@ -758,7 +759,7 @@ static int snd_mixart_playback_open(struct snd_pcm_substream *subs) } _exit_open: - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return err; } @@ -775,7 +776,7 @@ static int snd_mixart_capture_open(struct snd_pcm_substream *subs) int err = 0; int pcm_number; - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); if ( pcm == chip->pcm ) { pcm_number = MIXART_PCM_ANALOG; @@ -836,7 +837,7 @@ static int snd_mixart_capture_open(struct snd_pcm_substream *subs) } _exit_open: - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return err; } @@ -849,7 +850,7 @@ static int snd_mixart_close(struct snd_pcm_substream *subs) struct mixart_mgr *mgr = chip->mgr; struct mixart_stream *stream = subs->runtime->private_data; - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); snd_printdd("snd_mixart_close C%d/P%d/Sub%d\n", chip->chip_idx, stream->pcm_number, subs->number); @@ -868,7 +869,7 @@ static int snd_mixart_close(struct snd_pcm_substream *subs) stream->status = MIXART_STREAM_STATUS_FREE; stream->substream = NULL; - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return 0; } @@ -1335,12 +1336,12 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, mgr->msg_fifo_writeptr = 0; spin_lock_init(&mgr->msg_lock); - init_MUTEX(&mgr->msg_mutex); + mutex_init(&mgr->msg_mutex); init_waitqueue_head(&mgr->msg_sleep); atomic_set(&mgr->msg_processed, 0); /* init setup mutex*/ - init_MUTEX(&mgr->setup_mutex); + mutex_init(&mgr->setup_mutex); /* init message taslket */ tasklet_init(&mgr->msg_taskq, snd_mixart_msg_tasklet, (unsigned long) mgr); diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h index 3e84863ca02ccc..561634d5c0076a 100644 --- a/sound/pci/mixart/mixart.h +++ b/sound/pci/mixart/mixart.h @@ -24,6 +24,7 @@ #define __SOUND_MIXART_H #include +#include #include #define MIXART_DRIVER_VERSION 0x000100 /* 0.1.0 */ @@ -92,9 +93,9 @@ struct mixart_mgr { spinlock_t lock; /* interrupt spinlock */ spinlock_t msg_lock; /* mailbox spinlock */ - struct semaphore msg_mutex; /* mutex for blocking_requests */ + struct mutex msg_mutex; /* mutex for blocking_requests */ - struct semaphore setup_mutex; /* mutex used in hw_params, open and close */ + struct mutex setup_mutex; /* mutex used in hw_params, open and close */ /* hardware interface */ unsigned int dsp_loaded; /* bit flags of loaded dsp indices */ @@ -107,7 +108,7 @@ struct mixart_mgr { int sample_rate; int ref_count_rate; - struct semaphore mixer_mutex; /* mutex for mixer */ + struct mutex mixer_mutex; /* mutex for mixer */ }; diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index 07c707d7ebbfee..406ac3a9d42ade 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -22,6 +22,8 @@ #include #include +#include + #include #include #include "mixart.h" @@ -239,7 +241,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int wait_queue_t wait; long timeout; - down(&mgr->msg_mutex); + mutex_lock(&mgr->msg_mutex); init_waitqueue_entry(&wait, current); @@ -248,7 +250,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int err = send_msg(mgr, request, max_resp_size, 1, &msg_frame); /* send and mark the answer pending */ if (err) { spin_unlock_irq(&mgr->msg_lock); - up(&mgr->msg_mutex); + mutex_unlock(&mgr->msg_mutex); return err; } @@ -260,7 +262,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int if (! timeout) { /* error - no ack */ - up(&mgr->msg_mutex); + mutex_unlock(&mgr->msg_mutex); snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame); return -EIO; } @@ -276,7 +278,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int if( request->message_id != resp.message_id ) snd_printk(KERN_ERR "REPONSE ERROR!\n"); - up(&mgr->msg_mutex); + mutex_unlock(&mgr->msg_mutex); return err; } @@ -292,7 +294,7 @@ int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr, snd_assert((notif_event & MSG_TYPE_MASK) == MSG_TYPE_NOTIFY, return -EINVAL); snd_assert((notif_event & MSG_CANCEL_NOTIFY_MASK) == 0, return -EINVAL); - down(&mgr->msg_mutex); + mutex_lock(&mgr->msg_mutex); init_waitqueue_entry(&wait, current); @@ -301,7 +303,7 @@ int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr, err = send_msg(mgr, request, MSG_DEFAULT_SIZE, 1, ¬if_event); /* send and mark the notification event pending */ if(err) { spin_unlock_irq(&mgr->msg_lock); - up(&mgr->msg_mutex); + mutex_unlock(&mgr->msg_mutex); return err; } @@ -313,12 +315,12 @@ int snd_mixart_send_msg_wait_notif(struct mixart_mgr *mgr, if (! timeout) { /* error - no ack */ - up(&mgr->msg_mutex); + mutex_unlock(&mgr->msg_mutex); snd_printk(KERN_ERR "error: notification %x not received\n", notif_event); return -EIO; } - up(&mgr->msg_mutex); + mutex_unlock(&mgr->msg_mutex); return 0; } diff --git a/sound/pci/mixart/mixart_mixer.c b/sound/pci/mixart/mixart_mixer.c index 36a7e9ddfb1562..ed47b732c10345 100644 --- a/sound/pci/mixart/mixart_mixer.c +++ b/sound/pci/mixart/mixart_mixer.c @@ -24,6 +24,8 @@ #include #include #include +#include + #include #include "mixart.h" #include "mixart_core.h" @@ -353,7 +355,7 @@ static int mixart_analog_vol_info(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int mixart_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if(kcontrol->private_value == 0) { /* playback */ ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; @@ -361,7 +363,7 @@ static int mixart_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e ucontrol->value.integer.value[0] = chip->analog_capture_volume[0]; ucontrol->value.integer.value[1] = chip->analog_capture_volume[1]; } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -371,7 +373,7 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e int changed = 0; int is_capture, i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); is_capture = (kcontrol->private_value != 0); for(i=0; i<2; i++) { int new_volume = ucontrol->value.integer.value[i]; @@ -382,7 +384,7 @@ static int mixart_analog_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e } } if(changed) mixart_update_analog_audio_level(chip, is_capture); - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -408,10 +410,10 @@ static int mixart_audio_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_ele { struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->analog_playback_active[0]; ucontrol->value.integer.value[1] = chip->analog_playback_active[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -419,7 +421,7 @@ static int mixart_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele { struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); int i, changed = 0; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for(i=0; i<2; i++) { if(chip->analog_playback_active[i] != ucontrol->value.integer.value[i]) { chip->analog_playback_active[i] = ucontrol->value.integer.value[i]; @@ -427,7 +429,7 @@ static int mixart_audio_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ele } } if(changed) mixart_update_analog_audio_level(chip, 0); /* update playback levels */ - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -817,7 +819,7 @@ static int mixart_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem int *stored_volume; int is_capture = kcontrol->private_value & MIXART_VOL_REC_MASK; int is_aes = kcontrol->private_value & MIXART_VOL_AES_MASK; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if(is_capture) { if(is_aes) stored_volume = chip->digital_capture_volume[1]; /* AES capture */ else stored_volume = chip->digital_capture_volume[0]; /* analog capture */ @@ -828,7 +830,7 @@ static int mixart_pcm_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem } ucontrol->value.integer.value[0] = stored_volume[0]; ucontrol->value.integer.value[1] = stored_volume[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -841,7 +843,7 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem int is_aes = kcontrol->private_value & MIXART_VOL_AES_MASK; int* stored_volume; int i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if(is_capture) { if(is_aes) stored_volume = chip->digital_capture_volume[1]; /* AES capture */ else stored_volume = chip->digital_capture_volume[0]; /* analog capture */ @@ -860,7 +862,7 @@ static int mixart_pcm_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem if(is_capture) mixart_update_capture_stream_level(chip, is_aes); else mixart_update_playback_stream_level(chip, is_aes, idx); } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -880,12 +882,12 @@ static int mixart_pcm_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ snd_assert ( idx < MIXART_PLAYBACK_STREAMS ); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if(kcontrol->private_value & MIXART_VOL_AES_MASK) /* AES playback */ idx += MIXART_PLAYBACK_STREAMS; ucontrol->value.integer.value[0] = chip->digital_playback_active[idx][0]; ucontrol->value.integer.value[1] = chip->digital_playback_active[idx][1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -897,7 +899,7 @@ static int mixart_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ int i, j; snd_assert ( idx < MIXART_PLAYBACK_STREAMS ); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); j = idx; if(is_aes) j += MIXART_PLAYBACK_STREAMS; for(i=0; i<2; i++) { @@ -907,7 +909,7 @@ static int mixart_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ } } if(changed) mixart_update_playback_stream_level(chip, is_aes, idx); - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -956,10 +958,10 @@ static int mixart_update_monitoring(struct snd_mixart* chip, int channel) static int mixart_monitor_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->monitoring_volume[0]; ucontrol->value.integer.value[1] = chip->monitoring_volume[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -968,7 +970,7 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); int changed = 0; int i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for(i=0; i<2; i++) { if(chip->monitoring_volume[i] != ucontrol->value.integer.value[i]) { chip->monitoring_volume[i] = ucontrol->value.integer.value[i]; @@ -976,7 +978,7 @@ static int mixart_monitor_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_ changed = 1; } } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -995,10 +997,10 @@ static struct snd_kcontrol_new mixart_control_monitor_vol = { static int mixart_monitor_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->monitoring_active[0]; ucontrol->value.integer.value[1] = chip->monitoring_active[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -1007,7 +1009,7 @@ static int mixart_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct snd_mixart *chip = snd_kcontrol_chip(kcontrol); int changed = 0; int i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for(i=0; i<2; i++) { if(chip->monitoring_active[i] != ucontrol->value.integer.value[i]) { chip->monitoring_active[i] = ucontrol->value.integer.value[i]; @@ -1029,7 +1031,7 @@ static int mixart_monitor_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e } } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return (changed != 0); } @@ -1059,7 +1061,7 @@ int snd_mixart_create_mixer(struct mixart_mgr *mgr) struct snd_mixart *chip; int err, i; - init_MUTEX(&mgr->mixer_mutex); /* can be in another place */ + mutex_init(&mgr->mixer_mutex); /* can be in another place */ for(i=0; inum_cards; i++) { struct snd_kcontrol_new temp; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 0d0ff54f0fc60d..3a3ba6f547bc3b 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -32,6 +32,8 @@ #include #include #include +#include + #include #include #include @@ -235,7 +237,7 @@ struct nm256 { int irq_acks; irqreturn_t (*interrupt)(int, void *, struct pt_regs *); int badintrcount; /* counter to check bogus interrupts */ - struct semaphore irq_mutex; + struct mutex irq_mutex; struct nm256_stream streams[2]; @@ -459,32 +461,32 @@ snd_nm256_set_format(struct nm256 *chip, struct nm256_stream *s, /* acquire interrupt */ static int snd_nm256_acquire_irq(struct nm256 *chip) { - down(&chip->irq_mutex); + mutex_lock(&chip->irq_mutex); if (chip->irq < 0) { if (request_irq(chip->pci->irq, chip->interrupt, SA_INTERRUPT|SA_SHIRQ, chip->card->driver, chip)) { snd_printk(KERN_ERR "unable to grab IRQ %d\n", chip->pci->irq); - up(&chip->irq_mutex); + mutex_unlock(&chip->irq_mutex); return -EBUSY; } chip->irq = chip->pci->irq; } chip->irq_acks++; - up(&chip->irq_mutex); + mutex_unlock(&chip->irq_mutex); return 0; } /* release interrupt */ static void snd_nm256_release_irq(struct nm256 *chip) { - down(&chip->irq_mutex); + mutex_lock(&chip->irq_mutex); if (chip->irq_acks > 0) chip->irq_acks--; if (chip->irq_acks == 0 && chip->irq >= 0) { free_irq(chip->irq, chip); chip->irq = -1; } - up(&chip->irq_mutex); + mutex_unlock(&chip->irq_mutex); } /* @@ -1407,7 +1409,7 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, chip->use_cache = use_cache; spin_lock_init(&chip->reg_lock); chip->irq = -1; - init_MUTEX(&chip->irq_mutex); + mutex_init(&chip->irq_mutex); /* store buffer sizes in bytes */ chip->streams[SNDRV_PCM_STREAM_PLAYBACK].bufsize = playback_bufsize * 1024; diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index b2cba75b6b167f..31a3e8e1b2343e 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -28,6 +28,8 @@ #include #include #include +#include + #include #include #include @@ -518,7 +520,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) struct timeval my_tv1, my_tv2; do_gettimeofday(&my_tv1); #endif - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); /* check the pipes concerned and build pipe_array */ for (i = 0; i < mgr->num_cards; i++) { @@ -537,7 +539,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) } } if (capture_mask == 0 && playback_mask == 0) { - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); snd_printk(KERN_ERR "pcxhr_trigger_tasklet : no pipes\n"); return; } @@ -548,7 +550,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) /* synchronous stop of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); if (err) { - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error stop pipes (P%x C%x)\n", playback_mask, capture_mask); return; @@ -592,7 +594,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) /* synchronous start of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); if (err) { - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); snd_printk(KERN_ERR "pcxhr_trigger_tasklet : error start pipes (P%x C%x)\n", playback_mask, capture_mask); return; @@ -619,7 +621,7 @@ static void pcxhr_trigger_tasklet(unsigned long arg) } spin_unlock_irqrestore(&mgr->lock, flags); - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); #ifdef CONFIG_SND_DEBUG_DETECT do_gettimeofday(&my_tv2); @@ -728,7 +730,7 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) } */ - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); do { /* if the stream was stopped before, format and buffer were reset */ @@ -755,7 +757,7 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) } } while(0); /* do only once (so we can use break instead of goto) */ - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return err; } @@ -780,7 +782,7 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs, /* set up format for the stream */ format = params_format(hw); - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); stream->channels = channels; stream->format = format; @@ -789,7 +791,7 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs, /* err = pcxhr_set_format(stream); if(err) { - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return err; } */ @@ -801,7 +803,7 @@ static int pcxhr_hw_params(struct snd_pcm_substream *subs, err = pcxhr_update_r_buffer(stream); } */ - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return err; } @@ -847,7 +849,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) struct pcxhr_stream *stream; int is_capture; - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); /* copy the struct snd_pcm_hardware struct */ runtime->hw = pcxhr_caps; @@ -871,7 +873,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) /* streams in use */ snd_printk(KERN_ERR "pcxhr_open chip%d subs%d in use\n", chip->chip_idx, subs->number); - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return -EBUSY; } @@ -887,7 +889,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) &external_rate) || external_rate == 0) { /* cannot detect the external clock rate */ - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return -EBUSY; } runtime->hw.rate_min = runtime->hw.rate_max = external_rate; @@ -905,7 +907,7 @@ static int pcxhr_open(struct snd_pcm_substream *subs) mgr->ref_count_rate++; - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return 0; } @@ -916,7 +918,7 @@ static int pcxhr_close(struct snd_pcm_substream *subs) struct pcxhr_mgr *mgr = chip->mgr; struct pcxhr_stream *stream = subs->runtime->private_data; - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); snd_printdd("pcxhr_close chip%d subs%d\n", chip->chip_idx, subs->number); @@ -929,7 +931,7 @@ static int pcxhr_close(struct snd_pcm_substream *subs) stream->status = PCXHR_STREAM_STATUS_FREE; stream->substream = NULL; - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); return 0; } @@ -1264,7 +1266,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id spin_lock_init(&mgr->msg_lock); /* init setup mutex*/ - init_MUTEX(&mgr->setup_mutex); + mutex_init(&mgr->setup_mutex); /* init taslket */ tasklet_init(&mgr->msg_taskq, pcxhr_msg_tasklet, (unsigned long) mgr); diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 049f2b3f2867fb..652064787a55b3 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -24,6 +24,7 @@ #define __SOUND_PCXHR_H #include +#include #include #define PCXHR_DRIVER_VERSION 0x000804 /* 0.8.4 */ @@ -76,8 +77,8 @@ struct pcxhr_mgr { spinlock_t lock; /* interrupt spinlock */ spinlock_t msg_lock; /* message spinlock */ - struct semaphore setup_mutex; /* mutex used in hw_params, open and close */ - struct semaphore mixer_mutex; /* mutex for mixer */ + struct mutex setup_mutex; /* mutex used in hw_params, open and close */ + struct mutex mixer_mutex; /* mutex for mixer */ /* hardware interface */ unsigned int dsp_loaded; /* bit flags of loaded dsp indices */ diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 760e733ac25e31..94e63a1e90d9bf 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include "pcxhr.h" #include "pcxhr_hwdep.h" @@ -92,7 +93,7 @@ static int pcxhr_analog_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if (kcontrol->private_value == 0) { /* playback */ ucontrol->value.integer.value[0] = chip->analog_playback_volume[0]; ucontrol->value.integer.value[1] = chip->analog_playback_volume[1]; @@ -100,7 +101,7 @@ static int pcxhr_analog_vol_get(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = chip->analog_capture_volume[0]; ucontrol->value.integer.value[1] = chip->analog_capture_volume[1]; } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -111,7 +112,7 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol, int changed = 0; int is_capture, i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); is_capture = (kcontrol->private_value != 0); for (i = 0; i < 2; i++) { int new_volume = ucontrol->value.integer.value[i]; @@ -123,7 +124,7 @@ static int pcxhr_analog_vol_put(struct snd_kcontrol *kcontrol, pcxhr_update_analog_audio_level(chip, is_capture, i); } } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -150,10 +151,10 @@ static int pcxhr_audio_sw_get(struct snd_kcontrol *kcontrol, { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->analog_playback_active[0]; ucontrol->value.integer.value[1] = chip->analog_playback_active[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -162,7 +163,7 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int i, changed = 0; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for(i = 0; i < 2; i++) { if (chip->analog_playback_active[i] != ucontrol->value.integer.value[i]) { chip->analog_playback_active[i] = ucontrol->value.integer.value[i]; @@ -170,7 +171,7 @@ static int pcxhr_audio_sw_put(struct snd_kcontrol *kcontrol, pcxhr_update_analog_audio_level(chip, 0, i); /* update playback levels */ } } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -299,14 +300,14 @@ static int pcxhr_pcm_vol_get(struct snd_kcontrol *kcontrol, int *stored_volume; int is_capture = kcontrol->private_value; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if (is_capture) stored_volume = chip->digital_capture_volume; /* digital capture */ else stored_volume = chip->digital_playback_volume[idx]; /* digital playback */ ucontrol->value.integer.value[0] = stored_volume[0]; ucontrol->value.integer.value[1] = stored_volume[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -320,7 +321,7 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol, int *stored_volume; int i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if (is_capture) stored_volume = chip->digital_capture_volume; /* digital capture */ else @@ -335,7 +336,7 @@ static int pcxhr_pcm_vol_put(struct snd_kcontrol *kcontrol, } if (! is_capture && changed) pcxhr_update_playback_stream_level(chip, idx); /* update playback volume */ - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -356,10 +357,10 @@ static int pcxhr_pcm_sw_get(struct snd_kcontrol *kcontrol, struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->digital_playback_active[idx][0]; ucontrol->value.integer.value[1] = chip->digital_playback_active[idx][1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -370,7 +371,7 @@ static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); /* index */ int i, j; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); j = idx; for (i = 0; i < 2; i++) { if (chip->digital_playback_active[j][i] != ucontrol->value.integer.value[i]) { @@ -380,7 +381,7 @@ static int pcxhr_pcm_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v } if (changed) pcxhr_update_playback_stream_level(chip, idx); - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -402,10 +403,10 @@ static int pcxhr_monitor_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->monitoring_volume[0]; ucontrol->value.integer.value[1] = chip->monitoring_volume[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -416,7 +417,7 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, int changed = 0; int i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 2; i++) { if (chip->monitoring_volume[i] != ucontrol->value.integer.value[i]) { chip->monitoring_volume[i] = ucontrol->value.integer.value[i]; @@ -426,7 +427,7 @@ static int pcxhr_monitor_vol_put(struct snd_kcontrol *kcontrol, changed = 1; } } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -446,10 +447,10 @@ static int pcxhr_monitor_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); ucontrol->value.integer.value[0] = chip->monitoring_active[0]; ucontrol->value.integer.value[1] = chip->monitoring_active[1]; - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -460,7 +461,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol, int changed = 0; int i; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 2; i++) { if (chip->monitoring_active[i] != ucontrol->value.integer.value[i]) { chip->monitoring_active[i] = ucontrol->value.integer.value[i]; @@ -474,7 +475,7 @@ static int pcxhr_monitor_sw_put(struct snd_kcontrol *kcontrol, /* update right monitoring volume and mute */ pcxhr_update_audio_pipe_level(chip, 0, 1); - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return (changed != 0); } @@ -571,13 +572,13 @@ static int pcxhr_audio_src_put(struct snd_kcontrol *kcontrol, struct snd_pcxhr *chip = snd_kcontrol_chip(kcontrol); int ret = 0; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); if (chip->audio_capture_source != ucontrol->value.enumerated.item[0]) { chip->audio_capture_source = ucontrol->value.enumerated.item[0]; pcxhr_set_audio_source(chip); ret = 1; } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return ret; } @@ -636,9 +637,9 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); int rate, ret = 0; - down(&mgr->mixer_mutex); + mutex_lock(&mgr->mixer_mutex); if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) { - down(&mgr->setup_mutex); + mutex_lock(&mgr->setup_mutex); mgr->use_clock_type = ucontrol->value.enumerated.item[0]; if (mgr->use_clock_type) pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate); @@ -649,10 +650,10 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, if (mgr->sample_rate) mgr->sample_rate = rate; } - up(&mgr->setup_mutex); + mutex_unlock(&mgr->setup_mutex); ret = 1; /* return 1 even if the set was not done. ok ? */ } - up(&mgr->mixer_mutex); + mutex_unlock(&mgr->mixer_mutex); return ret; } @@ -685,7 +686,7 @@ static int pcxhr_clock_rate_get(struct snd_kcontrol *kcontrol, struct pcxhr_mgr *mgr = snd_kcontrol_chip(kcontrol); int i, err, rate; - down(&mgr->mixer_mutex); + mutex_lock(&mgr->mixer_mutex); for(i = 0; i < 3 + mgr->capture_chips; i++) { if (i == PCXHR_CLOCK_TYPE_INTERNAL) rate = mgr->sample_rate_real; @@ -696,7 +697,7 @@ static int pcxhr_clock_rate_get(struct snd_kcontrol *kcontrol, } ucontrol->value.integer.value[i] = rate; } - up(&mgr->mixer_mutex); + mutex_unlock(&mgr->mixer_mutex); return 0; } @@ -765,7 +766,7 @@ static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v unsigned char aes_bits; int i, err; - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for(i = 0; i < 5; i++) { if (kcontrol->private_value == 0) /* playback */ aes_bits = chip->aes_bits[i]; @@ -776,7 +777,7 @@ static int pcxhr_iec958_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v } ucontrol->value.iec958.status[i] = aes_bits; } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return 0; } @@ -828,14 +829,14 @@ static int pcxhr_iec958_put(struct snd_kcontrol *kcontrol, int i, changed = 0; /* playback */ - down(&chip->mgr->mixer_mutex); + mutex_lock(&chip->mgr->mixer_mutex); for (i = 0; i < 5; i++) { if (ucontrol->value.iec958.status[i] != chip->aes_bits[i]) { pcxhr_iec958_update_byte(chip, i, ucontrol->value.iec958.status[i]); changed = 1; } } - up(&chip->mgr->mixer_mutex); + mutex_unlock(&chip->mgr->mixer_mutex); return changed; } @@ -916,7 +917,7 @@ int pcxhr_create_mixer(struct pcxhr_mgr *mgr) struct snd_pcxhr *chip; int err, i; - init_MUTEX(&mgr->mixer_mutex); /* can be in another place */ + mutex_init(&mgr->mixer_mutex); /* can be in another place */ for (i = 0; i < mgr->num_cards; i++) { struct snd_kcontrol_new temp; diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index cf09ea99755c38..46c6982c9e88b8 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -27,6 +27,8 @@ #include #include #include +#include + #include #include @@ -201,16 +203,16 @@ snd_trident_alloc_sg_pages(struct snd_trident *trident, - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); blk = search_empty(hdr, runtime->dma_bytes); if (blk == NULL) { - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } if (lastpg(blk) - firstpg(blk) >= sgbuf->pages) { snd_printk(KERN_ERR "page calculation doesn't match: allocated pages = %d, trident = %d/%d\n", sgbuf->pages, firstpg(blk), lastpg(blk)); __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } @@ -221,12 +223,12 @@ snd_trident_alloc_sg_pages(struct snd_trident *trident, unsigned long ptr = (unsigned long)sgbuf->table[idx].buf; if (! is_valid_page(addr)) { __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } set_tlb_bus(trident, page, ptr, addr); } - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return blk; } @@ -248,10 +250,10 @@ snd_trident_alloc_cont_pages(struct snd_trident *trident, hdr = trident->tlb.memhdr; snd_assert(hdr != NULL, return NULL); - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); blk = search_empty(hdr, runtime->dma_bytes); if (blk == NULL) { - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } @@ -262,12 +264,12 @@ snd_trident_alloc_cont_pages(struct snd_trident *trident, ptr += SNDRV_TRIDENT_PAGE_SIZE, addr += SNDRV_TRIDENT_PAGE_SIZE) { if (! is_valid_page(addr)) { __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } set_tlb_bus(trident, page, ptr, addr); } - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return blk; } @@ -300,13 +302,13 @@ int snd_trident_free_pages(struct snd_trident *trident, snd_assert(blk != NULL, return -EINVAL); hdr = trident->tlb.memhdr; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); /* reset TLB entries */ for (page = firstpg(blk); page <= lastpg(blk); page++) set_silent_tlb(trident, page); /* free memory block */ __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return 0; } @@ -332,18 +334,18 @@ snd_trident_synth_alloc(struct snd_trident *hw, unsigned int size) struct snd_util_memblk *blk; struct snd_util_memhdr *hdr = hw->tlb.memhdr; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); blk = __snd_util_mem_alloc(hdr, size); if (blk == NULL) { - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } if (synth_alloc_pages(hw, blk)) { __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return NULL; } - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return blk; } @@ -356,10 +358,10 @@ snd_trident_synth_free(struct snd_trident *hw, struct snd_util_memblk *blk) { struct snd_util_memhdr *hdr = hw->tlb.memhdr; - down(&hdr->block_mutex); + mutex_lock(&hdr->block_mutex); synth_free_pages(hw, blk); __snd_util_mem_free(hdr, blk); - up(&hdr->block_mutex); + mutex_unlock(&hdr->block_mutex); return 0; } diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index c705af409b0fd3..9b6d345b83a666 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -24,6 +24,8 @@ #include #include #include +#include + #include #include #include @@ -861,10 +863,10 @@ static int vx_input_level_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vx222 *chip = (struct snd_vx222 *)_chip; - down(&_chip->mixer_mutex); + mutex_lock(&_chip->mixer_mutex); ucontrol->value.integer.value[0] = chip->input_level[0]; ucontrol->value.integer.value[1] = chip->input_level[1]; - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 0; } @@ -872,16 +874,16 @@ static int vx_input_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vx222 *chip = (struct snd_vx222 *)_chip; - down(&_chip->mixer_mutex); + mutex_lock(&_chip->mixer_mutex); if (chip->input_level[0] != ucontrol->value.integer.value[0] || chip->input_level[1] != ucontrol->value.integer.value[1]) { chip->input_level[0] = ucontrol->value.integer.value[0]; chip->input_level[1] = ucontrol->value.integer.value[1]; vx2_set_input_level(chip); - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 1; } - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 0; } @@ -907,14 +909,14 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vx222 *chip = (struct snd_vx222 *)_chip; - down(&_chip->mixer_mutex); + mutex_lock(&_chip->mixer_mutex); if (chip->mic_level != ucontrol->value.integer.value[0]) { chip->mic_level = ucontrol->value.integer.value[0]; vx2_set_input_level(chip); - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 1; } - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 0; } -- cgit 1.2.3-korg From 12aa757905d09b1dc2c1c3d0de3fa8f4c9726f2b Mon Sep 17 00:00:00 2001 From: Ingo Molnar Date: Mon, 16 Jan 2006 16:36:05 +0100 Subject: [ALSA] semaphore -> mutex (Archs, misc buses) Semaphore to mutex conversion. The conversion was generated via scripts, and the result was validated automatically via a script as well. Signed-off-by: Ingo Molnar Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 10 +++++----- sound/arm/aaci.h | 2 +- sound/arm/pxa2xx-ac97.c | 12 ++++++------ sound/pcmcia/vx/vxp_mixer.c | 12 ++++++------ sound/sparc/cs4231.c | 34 +++++++++++++++++----------------- sound/usb/usbaudio.c | 15 ++++++++------- sound/usb/usx2y/usbusx2y.c | 2 +- sound/usb/usx2y/usbusx2y.h | 2 +- sound/usb/usx2y/usbusx2yaudio.c | 8 ++++---- sound/usb/usx2y/usx2yhwdeppcm.c | 12 ++++++------ 10 files changed, 55 insertions(+), 54 deletions(-) diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 149feb41065481..5f22d70fefc0fa 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -73,7 +73,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned if (ac97->num >= 4) return; - down(&aaci->ac97_sem); + mutex_lock(&aaci->ac97_sem); aaci_ac97_select_codec(aaci, ac97); @@ -91,7 +91,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned v = readl(aaci->base + AACI_SLFR); } while (v & (SLFR_1TXB|SLFR_2TXB)); - up(&aaci->ac97_sem); + mutex_unlock(&aaci->ac97_sem); } /* @@ -105,7 +105,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (ac97->num >= 4) return ~0; - down(&aaci->ac97_sem); + mutex_lock(&aaci->ac97_sem); aaci_ac97_select_codec(aaci, ac97); @@ -145,7 +145,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) v = ~0; } - up(&aaci->ac97_sem); + mutex_unlock(&aaci->ac97_sem); return v; } @@ -783,7 +783,7 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev) card->shortname, dev->res.start, dev->irq[0]); aaci = card->private_data; - init_MUTEX(&aaci->ac97_sem); + mutex_init(&aaci->ac97_sem); spin_lock_init(&aaci->lock); aaci->card = card; aaci->dev = dev; diff --git a/sound/arm/aaci.h b/sound/arm/aaci.h index 83f73c2505c6ea..06295190606ce0 100644 --- a/sound/arm/aaci.h +++ b/sound/arm/aaci.h @@ -227,7 +227,7 @@ struct aaci { unsigned int fifosize; /* AC'97 */ - struct semaphore ac97_sem; + struct mutex ac97_sem; ac97_bus_t *ac97_bus; u32 maincr; diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 3acbc6023d1979..599aff8290e8d4 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -25,7 +25,7 @@ #include #include -#include +#include #include #include #include @@ -33,7 +33,7 @@ #include "pxa2xx-pcm.h" -static DECLARE_MUTEX(car_mutex); +static DEFINE_MUTEX(car_mutex); static DECLARE_WAIT_QUEUE_HEAD(gsr_wq); static volatile long gsr_bits; @@ -52,7 +52,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg unsigned short val = -1; volatile u32 *reg_addr; - down(&car_mutex); + mutex_lock(&car_mutex); /* set up primary or secondary codec space */ reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -79,7 +79,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg /* but we've just started another cycle... */ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); -out: up(&car_mutex); +out: mutex_unlock(&car_mutex); return val; } @@ -87,7 +87,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigne { volatile u32 *reg_addr; - down(&car_mutex); + mutex_lock(&car_mutex); /* set up primary or secondary codec space */ reg_addr = (ac97->num & 1) ? &SAC_REG_BASE : &PAC_REG_BASE; @@ -101,7 +101,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigne printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", __FUNCTION__, reg, GSR | gsr_bits); - up(&car_mutex); + mutex_unlock(&car_mutex); } static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) diff --git a/sound/pcmcia/vx/vxp_mixer.c b/sound/pcmcia/vx/vxp_mixer.c index 9450149b931cbb..e237f6c2018f0b 100644 --- a/sound/pcmcia/vx/vxp_mixer.c +++ b/sound/pcmcia/vx/vxp_mixer.c @@ -52,14 +52,14 @@ static int vx_mic_level_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip; - down(&_chip->mixer_mutex); + mutex_lock(&_chip->mixer_mutex); if (chip->mic_level != ucontrol->value.integer.value[0]) { vx_set_mic_level(_chip, ucontrol->value.integer.value[0]); chip->mic_level = ucontrol->value.integer.value[0]; - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 1; } - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 0; } @@ -95,14 +95,14 @@ static int vx_mic_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v { struct vx_core *_chip = snd_kcontrol_chip(kcontrol); struct snd_vxpocket *chip = (struct snd_vxpocket *)_chip; - down(&_chip->mixer_mutex); + mutex_lock(&_chip->mixer_mutex); if (chip->mic_level != ucontrol->value.integer.value[0]) { vx_set_mic_boost(_chip, ucontrol->value.integer.value[0]); chip->mic_level = ucontrol->value.integer.value[0]; - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 1; } - up(&_chip->mixer_mutex); + mutex_unlock(&_chip->mixer_mutex); return 0; } diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index fd654399878854..53a148b01b6b0d 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -115,8 +115,8 @@ struct snd_cs4231 { unsigned char image[32]; /* registers image */ int mce_bit; int calibrate_mute; - struct semaphore mce_mutex; - struct semaphore open_mutex; + struct mutex mce_mutex; + struct mutex open_mutex; union { #ifdef SBUS_SUPPORT @@ -775,7 +775,7 @@ static void snd_cs4231_playback_format(struct snd_cs4231 *chip, struct snd_pcm_h { unsigned long flags; - down(&chip->mce_mutex); + mutex_lock(&chip->mce_mutex); snd_cs4231_calibrate_mute(chip, 1); snd_cs4231_mce_up(chip); @@ -790,7 +790,7 @@ static void snd_cs4231_playback_format(struct snd_cs4231 *chip, struct snd_pcm_h snd_cs4231_mce_down(chip); snd_cs4231_calibrate_mute(chip, 0); - up(&chip->mce_mutex); + mutex_unlock(&chip->mce_mutex); } static void snd_cs4231_capture_format(struct snd_cs4231 *chip, struct snd_pcm_hw_params *params, @@ -798,7 +798,7 @@ static void snd_cs4231_capture_format(struct snd_cs4231 *chip, struct snd_pcm_hw { unsigned long flags; - down(&chip->mce_mutex); + mutex_lock(&chip->mce_mutex); snd_cs4231_calibrate_mute(chip, 1); snd_cs4231_mce_up(chip); @@ -819,7 +819,7 @@ static void snd_cs4231_capture_format(struct snd_cs4231 *chip, struct snd_pcm_hw snd_cs4231_mce_down(chip); snd_cs4231_calibrate_mute(chip, 0); - up(&chip->mce_mutex); + mutex_unlock(&chip->mce_mutex); } /* @@ -933,14 +933,14 @@ static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) { unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); if ((chip->mode & mode)) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return -EAGAIN; } if (chip->mode & CS4231_MODE_OPEN) { chip->mode |= mode; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } /* ok. now enable and ack CODEC IRQ */ @@ -960,7 +960,7 @@ static int snd_cs4231_open(struct snd_cs4231 *chip, unsigned int mode) spin_unlock_irqrestore(&chip->lock, flags); chip->mode = mode; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return 0; } @@ -968,10 +968,10 @@ static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) { unsigned long flags; - down(&chip->open_mutex); + mutex_lock(&chip->open_mutex); chip->mode &= ~mode; if (chip->mode & CS4231_MODE_OPEN) { - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); return; } snd_cs4231_calibrate_mute(chip, 1); @@ -1008,7 +1008,7 @@ static void snd_cs4231_close(struct snd_cs4231 *chip, unsigned int mode) snd_cs4231_calibrate_mute(chip, 0); chip->mode = 0; - up(&chip->open_mutex); + mutex_unlock(&chip->open_mutex); } /* @@ -1969,8 +1969,8 @@ static int __init snd_cs4231_sbus_create(struct snd_card *card, spin_lock_init(&chip->lock); spin_lock_init(&chip->c_dma.sbus_info.lock); spin_lock_init(&chip->p_dma.sbus_info.lock); - init_MUTEX(&chip->mce_mutex); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->mce_mutex); + mutex_init(&chip->open_mutex); chip->card = card; chip->dev_u.sdev = sdev; chip->regs_size = sdev->reg_addrs[0].reg_size; @@ -2157,8 +2157,8 @@ static int __init snd_cs4231_ebus_create(struct snd_card *card, spin_lock_init(&chip->lock); spin_lock_init(&chip->c_dma.ebus_info.lock); spin_lock_init(&chip->p_dma.ebus_info.lock); - init_MUTEX(&chip->mce_mutex); - init_MUTEX(&chip->open_mutex); + mutex_init(&chip->mce_mutex); + mutex_init(&chip->open_mutex); chip->flags |= CS4231_FLAG_EBUS; chip->card = card; chip->dev_u.pdev = edev->bus->self; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c9476c237c4222..53009bb642f671 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -47,6 +47,7 @@ #include #include #include +#include #include #include #include @@ -202,7 +203,7 @@ struct snd_usb_stream { * the all interfaces on the same card as one sound device. */ -static DECLARE_MUTEX(register_mutex); +static DEFINE_MUTEX(register_mutex); static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; @@ -3285,7 +3286,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, /* check whether it's already registered */ chip = NULL; - down(®ister_mutex); + mutex_lock(®ister_mutex); for (i = 0; i < SNDRV_CARDS; i++) { if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { @@ -3338,13 +3339,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, usb_chip[chip->index] = chip; chip->num_interfaces++; - up(®ister_mutex); + mutex_unlock(®ister_mutex); return chip; __error: if (chip && !chip->num_interfaces) snd_card_free(chip->card); - up(®ister_mutex); + mutex_unlock(®ister_mutex); __err_val: return NULL; } @@ -3364,7 +3365,7 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) chip = ptr; card = chip->card; - down(®ister_mutex); + mutex_lock(®ister_mutex); chip->shutdown = 1; chip->num_interfaces--; if (chip->num_interfaces <= 0) { @@ -3382,10 +3383,10 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr) snd_usb_mixer_disconnect(p); } usb_chip[chip->index] = NULL; - up(®ister_mutex); + mutex_unlock(®ister_mutex); snd_card_free(card); } else { - up(®ister_mutex); + mutex_unlock(®ister_mutex); } } diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index e0abb56bbe4920..cfec38d7839b09 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -351,7 +351,7 @@ static struct snd_card *usX2Y_create_card(struct usb_device *device) usX2Y(card)->chip.dev = device; usX2Y(card)->chip.card = card; init_waitqueue_head(&usX2Y(card)->prepare_wait_queue); - init_MUTEX (&usX2Y(card)->prepare_mutex); + mutex_init(&usX2Y(card)->prepare_mutex); INIT_LIST_HEAD(&usX2Y(card)->chip.midi_list); strcpy(card->driver, "USB "NAME_ALLCAPS""); sprintf(card->shortname, "TASCAM "NAME_ALLCAPS""); diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index 435c1feda9df62..456b5fdbc339af 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -34,7 +34,7 @@ struct usX2Ydev { unsigned int rate, format; int chip_status; - struct semaphore prepare_mutex; + struct mutex prepare_mutex; struct us428ctls_sharedmem *us428ctls_sharedmem; int wait_iso_frame; wait_queue_head_t us428ctls_wait_queue_head; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index a6bbc7a6348f81..f6bd0dee563ce3 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -811,7 +811,7 @@ static int snd_usX2Y_pcm_hw_free(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usX2Y_substream *subs = runtime->private_data; - down(&subs->usX2Y->prepare_mutex); + mutex_lock(&subs->usX2Y->prepare_mutex); snd_printdd("snd_usX2Y_hw_free(%p)\n", substream); if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { @@ -832,7 +832,7 @@ static int snd_usX2Y_pcm_hw_free(struct snd_pcm_substream *substream) usX2Y_urbs_release(subs); } } - up(&subs->usX2Y->prepare_mutex); + mutex_unlock(&subs->usX2Y->prepare_mutex); return snd_pcm_lib_free_pages(substream); } /* @@ -849,7 +849,7 @@ static int snd_usX2Y_pcm_prepare(struct snd_pcm_substream *substream) int err = 0; snd_printdd("snd_usX2Y_pcm_prepare(%p)\n", substream); - down(&usX2Y->prepare_mutex); + mutex_lock(&usX2Y->prepare_mutex); usX2Y_subs_prepare(subs); // Start hardware streams // SyncStream first.... @@ -869,7 +869,7 @@ static int snd_usX2Y_pcm_prepare(struct snd_pcm_substream *substream) err = usX2Y_urbs_start(subs); up_prepare_mutex: - up(&usX2Y->prepare_mutex); + mutex_unlock(&usX2Y->prepare_mutex); return err; } diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 796a7dcef09d59..315855082fe181 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -366,7 +366,7 @@ static int snd_usX2Y_usbpcm_hw_free(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct snd_usX2Y_substream *subs = runtime->private_data, *cap_subs2 = subs->usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; - down(&subs->usX2Y->prepare_mutex); + mutex_lock(&subs->usX2Y->prepare_mutex); snd_printdd("snd_usX2Y_usbpcm_hw_free(%p)\n", substream); if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) { @@ -395,7 +395,7 @@ static int snd_usX2Y_usbpcm_hw_free(struct snd_pcm_substream *substream) usX2Y_usbpcm_urbs_release(cap_subs2); } } - up(&subs->usX2Y->prepare_mutex); + mutex_unlock(&subs->usX2Y->prepare_mutex); return snd_pcm_lib_free_pages(substream); } @@ -503,7 +503,7 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) memset(usX2Y->hwdep_pcm_shm, 0, sizeof(struct snd_usX2Y_hwdep_pcm_shm)); } - down(&usX2Y->prepare_mutex); + mutex_lock(&usX2Y->prepare_mutex); usX2Y_subs_prepare(subs); // Start hardware streams // SyncStream first.... @@ -544,7 +544,7 @@ static int snd_usX2Y_usbpcm_prepare(struct snd_pcm_substream *substream) usX2Y->hwdep_pcm_shm->capture_iso_start = -1; up_prepare_mutex: - up(&usX2Y->prepare_mutex); + mutex_unlock(&usX2Y->prepare_mutex); return err; } @@ -621,7 +621,7 @@ static int usX2Y_pcms_lock_check(struct snd_card *card) if (dev->type != SNDRV_DEV_PCM) continue; pcm = dev->device_data; - down(&pcm->open_mutex); + mutex_lock(&pcm->open_mutex); } list_for_each(list, &card->devices) { int s; @@ -650,7 +650,7 @@ static void usX2Y_pcms_unlock(struct snd_card *card) if (dev->type != SNDRV_DEV_PCM) continue; pcm = dev->device_data; - up(&pcm->open_mutex); + mutex_unlock(&pcm->open_mutex); } } -- cgit 1.2.3-korg From 5a25c5cfd4f61f514decca3c4106210fb168ce19 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 18 Jan 2006 08:02:24 +0100 Subject: [ALSA] ymfpci - make rear channel swap optional Modules: YMFPCI driver Added rear_swap module option / kernel parameter to configure the rear channel swapping. Default value is enable to make the AC3 passthrough working, but analog only users might revert the previous behaviour. Signed-off-by: Jaroslav Kysela --- include/sound/ymfpci.h | 10 ++++++---- sound/pci/ymfpci/ymfpci.c | 5 ++++- sound/pci/ymfpci/ymfpci_main.c | 42 ++++++++++++++++++++++++++++-------------- 3 files changed, 38 insertions(+), 19 deletions(-) diff --git a/include/sound/ymfpci.h b/include/sound/ymfpci.h index d567bfdbf513af..d41cda97e952de 100644 --- a/include/sound/ymfpci.h +++ b/include/sound/ymfpci.h @@ -269,9 +269,10 @@ struct snd_ymfpci_pcm { enum snd_ymfpci_pcm_type type; struct snd_pcm_substream *substream; struct snd_ymfpci_voice *voices[2]; /* playback only */ - unsigned int running: 1; - unsigned int output_front: 1; - unsigned int output_rear: 1; + unsigned int running: 1, + output_front: 1, + output_rear: 1, + swap_rear: 1; unsigned int update_pcm_vol; u32 period_size; /* cached from runtime->period_size */ u32 buffer_size; /* cached from runtime->buffer_size */ @@ -344,6 +345,7 @@ struct snd_ymfpci { struct snd_kcontrol *spdif_pcm_ctl; int mode_dup4ch; int rear_opened; + int rear_swap; int spdif_opened; struct { u16 left; @@ -376,7 +378,7 @@ int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); int snd_ymfpci_pcm_spdif(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); int snd_ymfpci_pcm_4ch(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm); -int snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch); +int snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch, int rear_swap); int snd_ymfpci_timer(struct snd_ymfpci *chip, int device); #endif /* __SOUND_YMFPCI_H */ diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index dab9b831034189..db57ce939fa8fd 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -49,6 +49,7 @@ static long mpu_port[SNDRV_CARDS]; static long joystick_port[SNDRV_CARDS]; #endif static int rear_switch[SNDRV_CARDS]; +static int rear_swap[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 1 }; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the Yamaha DS-1 PCI soundcard."); @@ -66,6 +67,8 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); #endif module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); +module_param_array(rear_swap, bool, NULL, 0444); +MODULE_PARM_DESC(rear_swap, "Swap rear channels (must be enabled for correct IEC958 (S/PDIF)) output"); static struct pci_device_id snd_ymfpci_ids[] = { { 0x1073, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724 */ @@ -295,7 +298,7 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, snd_card_free(card); return err; } - if ((err = snd_ymfpci_mixer(chip, rear_switch[dev])) < 0) { + if ((err = snd_ymfpci_mixer(chip, rear_switch[dev], rear_swap[dev])) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 30ee53adb49413..8ac5ab50b5c769 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -536,19 +536,30 @@ static void snd_ymfpci_pcm_init_voice(struct snd_ymfpci_pcm *ypcm, unsigned int } } if (ypcm->output_rear) { - /* The SPDIF out channels seem to be swapped, so we have - * to swap them here, too. The rear analog out channels - * will be wrong, but otherwise AC3 would not work. - */ - if (use_left) { - bank->eff3_gain = - bank->eff3_gain_end = vol_left; - } - if (use_right) { - bank->eff2_gain = - bank->eff2_gain_end = vol_right; - } - } + if (!ypcm->swap_rear) { + if (use_left) { + bank->eff2_gain = + bank->eff2_gain_end = vol_left; + } + if (use_right) { + bank->eff3_gain = + bank->eff3_gain_end = vol_right; + } + } else { + /* The SPDIF out channels seem to be swapped, so we have + * to swap them here, too. The rear analog out channels + * will be wrong, but otherwise AC3 would not work. + */ + if (use_left) { + bank->eff3_gain = + bank->eff3_gain_end = vol_left; + } + if (use_right) { + bank->eff2_gain = + bank->eff2_gain_end = vol_right; + } + } + } } } @@ -898,6 +909,7 @@ static int snd_ymfpci_playback_open(struct snd_pcm_substream *substream) ypcm = runtime->private_data; ypcm->output_front = 1; ypcm->output_rear = chip->mode_dup4ch ? 1 : 0; + ypcm->swap_rear = chip->rear_swap; spin_lock_irq(&chip->reg_lock); if (ypcm->output_rear) { ymfpci_open_extension(chip); @@ -1738,7 +1750,7 @@ static void snd_ymfpci_mixer_free_ac97(struct snd_ac97 *ac97) chip->ac97 = NULL; } -int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch) +int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch, int rear_swap) { struct snd_ac97_template ac97; struct snd_kcontrol *kctl; @@ -1750,6 +1762,7 @@ int __devinit snd_ymfpci_mixer(struct snd_ymfpci *chip, int rear_switch) .read = snd_ymfpci_codec_read, }; + chip->rear_swap = rear_swap; if ((err = snd_ac97_bus(chip->card, 0, &ops, chip, &chip->ac97_bus)) < 0) return err; chip->ac97_bus->private_free = snd_ymfpci_mixer_free_ac97_bus; @@ -2297,6 +2310,7 @@ int __devinit snd_ymfpci_create(struct snd_card *card, return -EIO; } + chip->rear_swap = 1; if ((err = snd_ymfpci_ac3_init(chip)) < 0) { snd_ymfpci_free(chip); return err; -- cgit 1.2.3-korg From cdca881d79dc4133b35db5c855b688ee9cba12a9 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 18 Jan 2006 08:53:32 +0100 Subject: [ALSA] usb-audio: add Roland G-70 support Modules: USB generic driver Add a quirk entry for the Roland G-70. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 15 ++++++++++++++- 1 file changed, 14 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a433c6f2128d4b..42079dadbe8c9e 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1091,7 +1091,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, /* TODO: add Edirol UA-101 support */ - /* TODO: add Roland G-70 support */ +{ + /* has ID 0x0081 when not in "Advanced Driver" mode */ + USB_DEVICE(0x0582, 0x0080), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Roland", + .product_name = "G-70", + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + } +}, /* TODO: add Roland V-SYNTH XT support */ /* TODO: add BOSS GT-PRO support */ { -- cgit 1.2.3-korg From 3f72a30ccfd5054c01fc3bb0d37241df5380a54c Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 18 Jan 2006 11:50:40 +0100 Subject: [ALSA] snd_pcm_format_name() is no longer exported Modules: PCM Midlevel,USB generic driver Because snd_pcm_format_name() function is used only for informational purposes, it is no longer exported from the PCM midlevel to reduce space and dependency. usbaudio module shows only numeric value for format. Signed-off-by: Jaroslav Kysela --- sound/core/pcm.c | 1 - sound/usb/usbaudio.c | 2 +- 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index f903d1bd74d5f7..3da6a38c2d0fd6 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1105,7 +1105,6 @@ EXPORT_SYMBOL(snd_pcm_new_stream); EXPORT_SYMBOL(snd_pcm_notify); EXPORT_SYMBOL(snd_pcm_open_substream); EXPORT_SYMBOL(snd_pcm_release_substream); -EXPORT_SYMBOL(snd_pcm_format_name); /* pcm_native.c */ EXPORT_SYMBOL(snd_pcm_link_rwlock); #ifdef CONFIG_PM diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 53009bb642f671..51a862637f206c 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2029,7 +2029,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: %s\n", snd_pcm_format_name(fp->format)); + snd_iprintf(buffer, " Format: 0x%x\n", fp->format); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, -- cgit 1.2.3-korg From 05422575d4bc5d1a6aff22dfa9a0b5bc701ccf6f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 18 Jan 2006 15:44:53 +0100 Subject: [ALSA] usb-audio: fix number of G-70 ports Modules: USB generic driver The G-70 has one MIDI port in each direction, no two. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 42079dadbe8c9e..edb88871ff4a81 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1100,8 +1100,8 @@ YAMAHA_DEVICE(0x7010, "UB99"), .ifnum = 0, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 + .out_cables = 0x0001, + .in_cables = 0x0001 } } }, -- cgit 1.2.3-korg From 119c4ff52e5250a476054e6faca9a85422a0402b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 19 Jan 2006 08:25:19 +0100 Subject: [ALSA] usb-audio: add UM-1EX/UM-2EX information Modules: USB generic driver Add comments with information about UM-1EX/UM-2EX devices. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index edb88871ff4a81..bb7829cf98dccd 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -294,7 +294,8 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* a later revision uses ID 0x0099 */ + /* Has ID 0x0099 when not in "Advanced Driver" mode. + * The UM-2EX has only one input, but we cannot detect this. */ USB_DEVICE(0x0582, 0x0005), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "EDIROL", @@ -385,7 +386,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - /* a later revision uses ID 0x009d */ + /* has ID 0x009d when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0009), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "EDIROL", -- cgit 1.2.3-korg From d6ec894b6d6bf12885a34a4667bccb7f67e2916c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Jan 2006 14:05:06 +0100 Subject: [ALSA] Add the notes on PM to ens1370/ens1371 sections Modules: Documentation Add the notes on PM to ens1370/ens1371 sections. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index b45f7be47816d5..f28c45ca8395f0 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -513,6 +513,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards and autoprobe. + The power-management is supported. + Module snd-ens1371 ------------------ @@ -526,6 +528,8 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards and autoprobe. + The power-management is supported. + Module snd-es968 ---------------- -- cgit 1.2.3-korg From c82590d2b2877ec846c5313ea897420209b91ea7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 20 Jan 2006 17:13:45 +0100 Subject: [ALSA] Fix mulaw -> linear conversion in OSS PCM emulation Modules: ALSA<-OSS emulation Fixed the missing mulaw -> linear conversion in OSS PCM emulation code. Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/oss/pcm_plugin.c b/sound/core/oss/pcm_plugin.c index cec2774425ec4b..0e67dd280a5d4e 100644 --- a/sound/core/oss/pcm_plugin.c +++ b/sound/core/oss/pcm_plugin.c @@ -470,7 +470,8 @@ int snd_pcm_plug_format_plugins(struct snd_pcm_substream *plug, /* format change */ if (srcformat.format != dstformat.format) { tmpformat.format = dstformat.format; - if (tmpformat.format == SNDRV_PCM_FORMAT_MU_LAW) { + if (srcformat.format == SNDRV_PCM_FORMAT_MU_LAW || + tmpformat.format == SNDRV_PCM_FORMAT_MU_LAW) { err = snd_pcm_plugin_build_mulaw(plug, &srcformat, &tmpformat, &plugin); -- cgit 1.2.3-korg From ea50888d83cfb797ff7efadedc033b33bc2064bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 23 Jan 2006 15:49:18 +0100 Subject: [ALSA] Use dma_alloc_coherent() hack on i386 only Modules: Memalloc module Use dma_alloc_coherent() hack on i386 only (as a valid arch). Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 9f2b88cb5877fc..8360418d333ed1 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -83,7 +83,7 @@ struct snd_mem_list { * Hacks */ -#if defined(__i386__) || defined(__ppc__) || defined(__x86_64__) +#if defined(__i386__) /* * A hack to allocate large buffers via dma_alloc_coherent() * -- cgit 1.2.3-korg From 2ba8c15c738b64b4d3acaace1e19750362ff2b69 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 31 Jan 2006 14:44:28 +0100 Subject: [ALSA] Removed unneeded page-reserve Modules: Memalloc module Removed unneeded page-reservation. Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 32 ++------------------------------ 1 file changed, 2 insertions(+), 30 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 8360418d333ed1..3fc6f97075ed87 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -141,10 +141,6 @@ static void *snd_dma_hack_alloc_coherent(struct device *dev, size_t size, #endif /* arch */ -#if ! defined(__arm__) -#define NEED_RESERVE_PAGES -#endif - /* * * Generic memory allocators @@ -163,20 +159,6 @@ static inline void dec_snd_pages(int order) snd_allocated_pages -= 1 << order; } -static void mark_pages(struct page *page, int order) -{ - struct page *last_page = page + (1 << order); - while (page < last_page) - SetPageReserved(page++); -} - -static void unmark_pages(struct page *page, int order) -{ - struct page *last_page = page + (1 << order); - while (page < last_page) - ClearPageReserved(page++); -} - /** * snd_malloc_pages - allocate pages with the given size * @size: the size to allocate in bytes @@ -195,10 +177,8 @@ void *snd_malloc_pages(size_t size, gfp_t gfp_flags) snd_assert(gfp_flags != 0, return NULL); gfp_flags |= __GFP_COMP; /* compound page lets parts be mapped */ pg = get_order(size); - if ((res = (void *) __get_free_pages(gfp_flags, pg)) != NULL) { - mark_pages(virt_to_page(res), pg); + if ((res = (void *) __get_free_pages(gfp_flags, pg)) != NULL) inc_snd_pages(pg); - } return res; } @@ -217,7 +197,6 @@ void snd_free_pages(void *ptr, size_t size) return; pg = get_order(size); dec_snd_pages(pg); - unmark_pages(virt_to_page(ptr), pg); free_pages((unsigned long) ptr, pg); } @@ -242,12 +221,8 @@ static void *snd_malloc_dev_pages(struct device *dev, size_t size, dma_addr_t *d | __GFP_NORETRY /* don't trigger OOM-killer */ | __GFP_NOWARN; /* no stack trace print - this call is non-critical */ res = dma_alloc_coherent(dev, PAGE_SIZE << pg, dma, gfp_flags); - if (res != NULL) { -#ifdef NEED_RESERVE_PAGES - mark_pages(virt_to_page(res), pg); /* should be dma_to_page() */ -#endif + if (res != NULL) inc_snd_pages(pg); - } return res; } @@ -262,9 +237,6 @@ static void snd_free_dev_pages(struct device *dev, size_t size, void *ptr, return; pg = get_order(size); dec_snd_pages(pg); -#ifdef NEED_RESERVE_PAGES - unmark_pages(virt_to_page(ptr), pg); /* should be dma_to_page() */ -#endif dma_free_coherent(dev, PAGE_SIZE << pg, ptr, dma); } -- cgit 1.2.3-korg From e957ebf164e880ddb0c057418195db47d013c4ac Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 2 Feb 2006 07:56:54 +0100 Subject: [ALSA] ice1712 & cs8427 - fix problem for S/PDIF input setup Modules: I2C cs8427,ICE1712 driver See ALSA bug#1785 for more details. Signed-off-by: Jaroslav Kysela --- sound/i2c/cs8427.c | 7 +++++-- sound/pci/ice1712/ice1712.c | 33 ++++++++++++++++++--------------- 2 files changed, 23 insertions(+), 17 deletions(-) diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 9deba80a587cb8..cb89f7eb9236df 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -291,11 +291,13 @@ static void snd_cs8427_reset(struct snd_i2c_device *cs8427) { struct cs8427 *chip; unsigned long end_time; - int data; + int data, aes3input = 0; snd_assert(cs8427, return); chip = cs8427->private_data; snd_i2c_lock(cs8427->bus); + if ((chip->regmap[CS8427_REG_CLOCKSOURCE] & CS8427_RXDAES3INPUT) == CS8427_RXDAES3INPUT) /* AES3 bit is set */ + aes3input = 1; chip->regmap[CS8427_REG_CLOCKSOURCE] &= ~(CS8427_RUN | CS8427_RXDMASK); snd_cs8427_reg_write(cs8427, CS8427_REG_CLOCKSOURCE, chip->regmap[CS8427_REG_CLOCKSOURCE]); @@ -316,7 +318,8 @@ static void snd_cs8427_reset(struct snd_i2c_device *cs8427) } snd_i2c_lock(cs8427->bus); chip->regmap[CS8427_REG_CLOCKSOURCE] &= ~CS8427_RXDMASK; - chip->regmap[CS8427_REG_CLOCKSOURCE] |= CS8427_RXDAES3INPUT; + if (aes3input) + chip->regmap[CS8427_REG_CLOCKSOURCE] |= CS8427_RXDAES3INPUT; snd_cs8427_reg_write(cs8427, CS8427_REG_CLOCKSOURCE, chip->regmap[CS8427_REG_CLOCKSOURCE]); snd_i2c_unlock(cs8427->bus); diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 3156a31329903f..6682e1fc144983 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -317,7 +317,6 @@ static void snd_ice1712_set_gpio_data(struct snd_ice1712 *ice, unsigned int val) inb(ICEREG(ice, DATA)); /* dummy read for pci-posting */ } - /* * * CS8427 interface @@ -397,6 +396,20 @@ int __devinit snd_ice1712_init_cs8427(struct snd_ice1712 *ice, int addr) return 0; } +static void snd_ice1712_set_input_clock_source(struct snd_ice1712 *ice, int spdif_is_master) +{ + /* change CS8427 clock source too */ + if (ice->cs8427) + snd_ice1712_cs8427_set_input_clock(ice, spdif_is_master); + /* notify ak4524 chip as well */ + if (spdif_is_master) { + unsigned int i; + for (i = 0; i < ice->akm_codecs; i++) { + if (ice->akm[i].ops.set_rate_val) + ice->akm[i].ops.set_rate_val(&ice->akm[i], 0); + } + } +} /* * Interrupt handler @@ -1857,20 +1870,8 @@ static int snd_ice1712_pro_internal_clock_put(struct snd_kcontrol *kcontrol, spin_unlock_irq(&ice->reg_lock); if ((oval & ICE1712_SPDIF_MASTER) != - (inb(ICEMT(ice, RATE)) & ICE1712_SPDIF_MASTER)) { - /* change CS8427 clock source too */ - if (ice->cs8427) { - snd_ice1712_cs8427_set_input_clock(ice, is_spdif_master(ice)); - } - /* notify ak4524 chip as well */ - if (is_spdif_master(ice)) { - unsigned int i; - for (i = 0; i < ice->akm_codecs; i++) { - if (ice->akm[i].ops.set_rate_val) - ice->akm[i].ops.set_rate_val(&ice->akm[i], 0); - } - } - } + (inb(ICEMT(ice, RATE)) & ICE1712_SPDIF_MASTER)) + snd_ice1712_set_input_clock_source(ice, is_spdif_master(ice)); return change; } @@ -2736,6 +2737,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, } } + snd_ice1712_set_input_clock_source(ice, 0); + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, ice->port, ice->irq); -- cgit 1.2.3-korg From cc7a59bd8dcee9b01e0c9ab21380071f0ae6ef0d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Feb 2006 17:11:06 +0100 Subject: [ALSA] usb-audio: rename QUIRK_MIDI_MIDITECH to QUIRK_MIDI_CME Modules: USB generic driver Rename QUIRK_MIDI_MIDITECH to QUIRK_MIDI_CME because Miditech keyboards are built by CME and use the same protocol, and don't force a Miditech product name for the USB ID used by both Miditech and CME UF-x keyboards. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 2 +- sound/usb/usbaudio.h | 4 ++-- sound/usb/usbmidi.c | 2 +- sound/usb/usbquirks.h | 10 ++++++---- 4 files changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 51a862637f206c..7b1b9f29f3779f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3074,7 +3074,7 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_NOVATION] = snd_usb_create_midi_interface, [QUIRK_MIDI_RAW] = snd_usb_create_midi_interface, [QUIRK_MIDI_EMAGIC] = snd_usb_create_midi_interface, - [QUIRK_MIDI_MIDITECH] = snd_usb_create_midi_interface, + [QUIRK_MIDI_CME] = snd_usb_create_midi_interface, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA700_UA25] = create_ua700_ua25_quirk, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index ecd724bfe5a524..88733524d0fb1c 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,7 +161,7 @@ enum quirk_type { QUIRK_MIDI_NOVATION, QUIRK_MIDI_RAW, QUIRK_MIDI_EMAGIC, - QUIRK_MIDI_MIDITECH, + QUIRK_MIDI_CME, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA700_UA25, @@ -209,7 +209,7 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_MIDI_EMAGIC, data points to a snd_usb_midi_endpoint_info * structure (out_cables and in_cables only) */ -/* for QUIRK_MIDI_MIDITECH, data is NULL */ +/* for QUIRK_MIDI_CME, data is NULL */ /* */ diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 7f7e371c5df19b..7580339dba221f 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1576,7 +1576,7 @@ int snd_usb_create_midi_interface(struct snd_usb_audio* chip, sizeof(struct snd_usb_midi_endpoint_info)); err = snd_usbmidi_detect_endpoints(umidi, &endpoints[0], 1); break; - case QUIRK_MIDI_MIDITECH: + case QUIRK_MIDI_CME: err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints); break; default: diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index bb7829cf98dccd..0485e21861a193 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1537,22 +1537,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Miditech devices */ { USB_DEVICE(0x4752, 0x0011), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Miditech", .product_name = "Midistart-2", .ifnum = 0, - .type = QUIRK_MIDI_MIDITECH + .type = QUIRK_MIDI_CME } }, + +/* Central Music devices */ { + /* this ID used by both Miditech MidiStudio-2 and CME UF-x */ USB_DEVICE(0x7104, 0x2202), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Miditech", - .product_name = "MidiStudio-2", .ifnum = 0, - .type = QUIRK_MIDI_MIDITECH + .type = QUIRK_MIDI_CME } }, -- cgit 1.2.3-korg From 531af46279fe113f9e41bd5167c8868c936813b5 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Wed, 8 Feb 2006 07:40:33 +0100 Subject: [ALSA] ice1712 - disable unused ADCs & DACs on DMX6fire Modules: ICE1712 driver Change the value written to hardware configuration register of envy24 chip in the case of Terratec DMX6fire to restrict the active ADCs & DACs to 6. Also add the dxr_enable module option to eventually leave the old behaviour when user requests. Signed-off-by: Alan Horstmann Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/ice1712.c | 16 ++++++++++++++-- sound/pci/ice1712/ice1712.h | 1 + 2 files changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 6682e1fc144983..d6dfe001fde3d2 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -87,6 +87,7 @@ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card static char *model[SNDRV_CARDS]; static int omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ static int cs8427_timeout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* CS8427 S/PDIF transciever reset timeout value in msec */ +static int dxr_enable[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* DXR enable for DMX6FIRE */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for ICE1712 soundcard."); @@ -100,6 +101,8 @@ module_param_array(cs8427_timeout, int, NULL, 0444); MODULE_PARM_DESC(cs8427_timeout, "Define reset timeout for cs8427 chip in msec resolution."); module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); +module_param_array(dxr_enable, int, NULL, 0444); +MODULE_PARM_DESC(dsr_enable, "Enable DXR support for Terratec DMX6FIRE."); static struct pci_device_id snd_ice1712_ids[] = { @@ -2390,7 +2393,13 @@ static int __devinit snd_ice1712_chip_init(struct snd_ice1712 *ice) udelay(200); outb(ICE1712_NATIVE, ICEREG(ice, CONTROL)); udelay(200); - pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]); + if (ice->eeprom.subvendor == ICE1712_SUBDEVICE_DMX6FIRE && !ice->dxr_enable) { + /* Limit active ADCs and DACs to 6; */ + /* Note: DXR extension not supported */ + pci_write_config_byte(ice->pci, 0x60, 0x0a); + } else { + pci_write_config_byte(ice->pci, 0x60, ice->eeprom.data[ICE_EEP1_CODEC]); + } pci_write_config_byte(ice->pci, 0x61, ice->eeprom.data[ICE_EEP1_ACLINK]); pci_write_config_byte(ice->pci, 0x62, ice->eeprom.data[ICE_EEP1_I2SID]); pci_write_config_byte(ice->pci, 0x63, ice->eeprom.data[ICE_EEP1_SPDIF]); @@ -2526,6 +2535,7 @@ static int __devinit snd_ice1712_create(struct snd_card *card, const char *modelname, int omni, int cs8427_timeout, + int dxr_enable, struct snd_ice1712 ** r_ice1712) { struct snd_ice1712 *ice; @@ -2558,6 +2568,7 @@ static int __devinit snd_ice1712_create(struct snd_card *card, else if (cs8427_timeout > 1000) cs8427_timeout = 1000; ice->cs8427_timeout = cs8427_timeout; + ice->dxr_enable = dxr_enable; spin_lock_init(&ice->reg_lock); mutex_init(&ice->gpio_mutex); mutex_init(&ice->i2c_mutex); @@ -2660,7 +2671,8 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, strcpy(card->shortname, "ICEnsemble ICE1712"); if ((err = snd_ice1712_create(card, pci, model[dev], omni[dev], - cs8427_timeout[dev], &ice)) < 0) { + cs8427_timeout[dev], dxr_enable[dev], + &ice)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/ice1712/ice1712.h b/sound/pci/ice1712/ice1712.h index d7416a83fcacdf..f9b22d4a39323b 100644 --- a/sound/pci/ice1712/ice1712.h +++ b/sound/pci/ice1712/ice1712.h @@ -325,6 +325,7 @@ struct snd_ice1712 { unsigned int pro_volumes[20]; unsigned int omni: 1; /* Delta Omni I/O */ + unsigned int dxr_enable: 1; /* Terratec DXR enable for DMX6FIRE */ unsigned int vt1724: 1; unsigned int vt1720: 1; unsigned int has_spdif: 1; /* VT1720/4 - has SPDIF I/O */ -- cgit 1.2.3-korg From abf58f095525c0e46b4ee64a4f9c2084b4c08f4c Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 8 Feb 2006 09:10:36 +0100 Subject: [ALSA] bt848 - added Leadtek Winfast tv 2000xp delux to whitelist Modules: BT87x driver - added 0x107d:0x6606 to whitelist - print also the pci device ID for developers when model is not known Signed-off-by: Jaroslav Kysela --- sound/pci/bt87x.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index c840a4c08e9814..7f1a19c13f10b8 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -783,6 +783,8 @@ static struct pci_device_id snd_bt87x_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100), /* AVerMedia Studio No. 103, 203, ...? */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), + /* Leadtek Winfast tv 2000xp delux */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000), { } }; MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); @@ -816,13 +818,13 @@ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci) for (i = 0; i < ARRAY_SIZE(blacklist); ++i) if (blacklist[i].subvendor == pci->subsystem_vendor && blacklist[i].subdevice == pci->subsystem_device) { - snd_printdd(KERN_INFO "card %#04x:%#04x has no audio\n", - pci->subsystem_vendor, pci->subsystem_device); + snd_printdd(KERN_INFO "card %#04x-%#04x:%#04x has no audio\n", + pci->device, pci->subsystem_vendor, pci->subsystem_device); return -EBUSY; } - snd_printk(KERN_INFO "unknown card %#04x:%#04x, using default rate 32000\n", - pci->subsystem_vendor, pci->subsystem_device); + snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x, using default rate 32000\n", + pci->device, pci->subsystem_vendor, pci->subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " "\n"); -- cgit 1.2.3-korg From b2b8229dde970b95e407d90a140e8a8753e1f0f6 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 8 Feb 2006 12:38:23 +0100 Subject: [ALSA] usb-audio: add Miditech Play'n Roll support Modules: USB generic driver Add a quirk entry for the Miditech Play'n Roll. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 0485e21861a193..7820d7bd025fd5 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1507,6 +1507,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_MIDI_STANDARD_INTERFACE } }, +{ + USB_DEVICE(0x0ccd, 0x0035), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Miditech", + .product_name = "Play'n Roll", + .ifnum = 0, + .type = QUIRK_MIDI_CME + } +}, /* Novation EMS devices */ { -- cgit 1.2.3-korg From 50dabc2d1139ba01362418874152aeeb591a4544 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Feb 2006 11:45:20 +0100 Subject: [ALSA] ac97 - Add support of static resolution tables Modules: AC97 Codec Added the support of static resolution table support for codecs that the driver cannot probe the volume resolution properly. The table pointer should be set in each codec patch. Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 7 +++++++ sound/pci/ac97/ac97_codec.c | 12 ++++++++++++ 2 files changed, 19 insertions(+) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index ad3fe046f6cf81..9036d25e1baca3 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -444,6 +444,12 @@ struct snd_ac97_template { DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */ }; +/* static resolution table */ +struct snd_ac97_res_table { + unsigned short reg; /* register */ + unsigned short bits; /* resolution bitmask */ +}; + struct snd_ac97 { /* -- lowlevel (hardware) driver specific -- */ struct snd_ac97_build_ops * build_ops; @@ -464,6 +470,7 @@ struct snd_ac97 { unsigned short caps; /* capabilities (register 0) */ unsigned short ext_id; /* extended feature identification (register 28) */ unsigned short ext_mid; /* extended modem ID (register 3C) */ + const struct snd_ac97_res_table *res_table; /* static resolution */ unsigned int scaps; /* driver capabilities */ unsigned int flags; /* specific code */ unsigned int rates[6]; /* see AC97_RATES_* defines */ diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6108cdc5efb655..124c1bc4cb9272 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1030,6 +1030,18 @@ static void check_volume_resolution(struct snd_ac97 *ac97, int reg, unsigned cha unsigned char max[3] = { 63, 31, 15 }; int i; + /* first look up the static resolution table */ + if (ac97->res_table) { + const struct snd_ac97_res_table *tbl; + for (tbl = ac97->res_table; tbl->reg; tbl++) { + if (tbl->reg == reg) { + *lo_max = tbl->bits & 0xff; + *hi_max = (tbl->bits >> 8) & 0xff; + return; + } + } + } + *lo_max = *hi_max = 0; for (i = 0 ; i < ARRAY_SIZE(cbit); i++) { unsigned short val; -- cgit 1.2.3-korg From ba22429d3ea3b9945735b88d4dde74711171ffab Mon Sep 17 00:00:00 2001 From: Charl Coetzee Date: Thu, 9 Feb 2006 11:48:21 +0100 Subject: [ALSA] ac97 - Added a codec patch for LM4550 Modules: AC97 Codec Added a codec patch for LM4550. It sets up a static volume resolution table. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_codec.c | 2 +- sound/pci/ac97/ac97_patch.c | 31 +++++++++++++++++++++++++++++++ sound/pci/ac97/ac97_patch.h | 1 + 3 files changed, 33 insertions(+), 1 deletion(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 124c1bc4cb9272..38b6c65d40c37d 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -150,7 +150,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x49544561, 0xffffffff, "IT2646E", patch_it2646, NULL }, { 0x4e534300, 0xffffffff, "LM4540,43,45,46,48", NULL, NULL }, // only guess --jk { 0x4e534331, 0xffffffff, "LM4549", NULL, NULL }, -{ 0x4e534350, 0xffffffff, "LM4550", NULL, NULL }, +{ 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix { 0x50534304, 0xffffffff, "UCB1400", NULL, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 8bc79cbe321523..5c03b4b6f5c155 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -2825,3 +2825,34 @@ int mpatch_si3036(struct snd_ac97 * ac97) snd_ac97_write_cache(ac97, 0x68, 0); return 0; } + +/* + * LM 4550 Codec + * + * We use a static resolution table since LM4550 codec cannot be + * properly autoprobed to determine the resolution via + * check_volume_resolution(). + */ + +static struct snd_ac97_res_table lm4550_restbl[] = { + { AC97_MASTER, 0x1f1f }, + { AC97_HEADPHONE, 0x1f1f }, + { AC97_MASTER_MONO, 0x001f }, + { AC97_PC_BEEP, 0x001f }, /* LSB is ignored */ + { AC97_PHONE, 0x001f }, + { AC97_PHONE, 0x001f }, + { AC97_MIC, 0x001f }, + { AC97_LINE, 0x1f1f }, + { AC97_CD, 0x1f1f }, + { AC97_VIDEO, 0x1f1f }, + { AC97_AUX, 0x1f1f }, + { AC97_PCM, 0x1f1f }, + { AC97_REC_GAIN, 0x0f0f }, + { } /* terminator */ +}; + +int patch_lm4550(struct snd_ac97 *ac97) +{ + ac97->res_table = lm4550_restbl; + return 0; +} diff --git a/sound/pci/ac97/ac97_patch.h b/sound/pci/ac97/ac97_patch.h index 5060cb6f2ec370..adcaa04586cbb5 100644 --- a/sound/pci/ac97/ac97_patch.h +++ b/sound/pci/ac97/ac97_patch.h @@ -59,3 +59,4 @@ int patch_vt1616(struct snd_ac97 * ac97); int patch_vt1617a(struct snd_ac97 * ac97); int patch_it2646(struct snd_ac97 * ac97); int mpatch_si3036(struct snd_ac97 * ac97); +int patch_lm4550(struct snd_ac97 * ac97); -- cgit 1.2.3-korg From 4c5186ed6b25278df595edf2d355ee87b00c4426 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Thu, 9 Feb 2006 11:53:48 +0100 Subject: [ALSA] hda: add PCM for 2nd ADC on ALC260 Modules: HDA Codec driver The following patch against alsa 1.0.11rc3 creates a PCM device (pcm1c) for the second ADC present on the ALC260 codec used by the hda driver. It also defines a new mixer control allowing the mode of retasking pins to be set; this means a user can (for example) designate the headphone jack to be a second input. With this patch in place it is possible to do 4 channel recording on laptops equipped with an ALC260 codec assuming both a stereo line-in jack is provided in addition to a headphone jack. Mixer controls are provided to allow the headphone jack to be switched as an input. In addition, an (input only) mode control is configured for the line-in jack to allow a bias voltage to be requested (VREF80 or VREF50) so headsets based on condensor microphones have a chance of working. This patch has been tested on a Fujitsu S7020 laptop and as such these features are currently only configured for the 'fujitsu' model. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 149 +++++++++++++++++++++++++++++++++--------- 1 file changed, 119 insertions(+), 30 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b76755264730d6..c8fc6269b03c89 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -132,7 +132,7 @@ struct alc_spec { int num_channel_mode; /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ + struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; @@ -218,56 +218,96 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va spec->num_channel_mode, &spec->multiout.max_channels); } - /* - * Control of pin widget settings via the mixer. Only boolean settings are - * supported, so VrefEn can't be controlled using these functions as they - * stand. + * Control the mode of pin widget settings via the mixer. "pc" is used + * instead of "%" to avoid consequences of accidently treating the % as + * being part of a format specifier. Maximum allowed length of a value is + * 63 characters plus NULL terminator. + */ +static char *alc_pin_mode_names[] = { + "Line in", "Mic 80pc bias", "Mic 50pc bias", + "Line out", "Headphone out", +}; +static unsigned char alc_pin_mode_values[] = { + PIN_IN, PIN_VREF80, PIN_VREF50, PIN_OUT, PIN_HP, +}; +/* The control can present all 5 options, or it can limit the options based + * in the pin being assumed to be exclusively an input or an output pin. */ -static int alc_pinctl_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +#define ALC_PIN_DIR_IN 0x00 +#define ALC_PIN_DIR_OUT 0x01 +#define ALC_PIN_DIR_INOUT 0x02 + +/* Info about the pin modes supported by the three different pin directions. + * For each direction the minimum and maximum values are given. + */ +static signed char alc_pin_mode_dir_info[3][2] = { + { 0, 2 }, /* ALC_PIN_DIR_IN */ + { 3, 4 }, /* ALC_PIN_DIR_OUT */ + { 0, 4 }, /* ALC_PIN_DIR_INOUT */ +}; +#define alc_pin_mode_min(_dir) (alc_pin_mode_dir_info[_dir][0]) +#define alc_pin_mode_max(_dir) (alc_pin_mode_dir_info[_dir][1]) +#define alc_pin_mode_n_items(_dir) \ + (alc_pin_mode_max(_dir)-alc_pin_mode_min(_dir)+1) + +static int alc_pin_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + unsigned int item_num = uinfo->value.enumerated.item; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; + uinfo->value.enumerated.items = alc_pin_mode_n_items(dir); + + if (item_numalc_pin_mode_max(dir)) + item_num = alc_pin_mode_min(dir); + strcpy(uinfo->value.enumerated.name, alc_pin_mode_names[item_num]); return 0; } -static int alc_pinctl_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_pin_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + unsigned int i; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value & 0xffff; - long mask = (kcontrol->private_value >> 16) & 0xff; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; + unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); - *valp = 0; - if (snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00) & mask) - *valp = 1; + /* Find enumerated value for current pinctl setting */ + i = alc_pin_mode_min(dir); + while (alc_pin_mode_values[i]!=pinctl && i<=alc_pin_mode_max(dir)) + i++; + *valp = i<=alc_pin_mode_max(dir)?i:alc_pin_mode_min(dir); return 0; } -static int alc_pinctl_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { + signed int change; struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = kcontrol->private_value & 0xffff; - long mask = (kcontrol->private_value >> 16) & 0xff; - long *valp = ucontrol->value.integer.value; + unsigned char dir = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; unsigned int pinctl = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_PIN_WIDGET_CONTROL,0x00); - int change = ((pinctl & mask)!=0) != *valp; + if (valalc_pin_mode_max(dir)) + val = alc_pin_mode_min(dir); + + change = pinctl != alc_pin_mode_values[val]; if (change) snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL, - *valp?(pinctl|mask):(pinctl&~mask)); + alc_pin_mode_values[val]); return change; } -#define ALC_PINCTL_SWITCH(xname, nid, mask) \ +#define ALC_PIN_MODE(xname, nid, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ - .info = alc_pinctl_switch_info, \ - .get = alc_pinctl_switch_get, \ - .put = alc_pinctl_switch_put, \ - .private_value = (nid) | (mask<<16) } - + .info = alc_pin_mode_info, \ + .get = alc_pin_mode_get, \ + .put = alc_pin_mode_put, \ + .private_value = nid | (dir<<16) } /* * set up from the preset table @@ -1250,6 +1290,13 @@ static struct hda_pcm_stream alc880_pcm_digital_capture = { /* NID is set in alc_build_pcms */ }; +/* Used by alc_build_pcms to flag that a PCM has no playback stream */ +static struct hda_pcm_stream alc_pcm_null_playback = { + .substreams = 0, + .channels_min = 0, + .channels_max = 0, +}; + static int alc_build_pcms(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -1280,6 +1327,23 @@ static int alc_build_pcms(struct hda_codec *codec) } } + /* If the use of more than one ADC is requested for the current + * model, configure a second analog capture-only PCM. + */ + if (spec->num_adc_nids > 1) { + codec->num_pcms++; + info++; + info->name = spec->stream_name_analog; + /* No playback stream for second PCM */ + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = alc_pcm_null_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = 0; + if (spec->stream_analog_capture) { + snd_assert(spec->adc_nids, return -EINVAL); + info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); + info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[1]; + } + } + if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms++; info++; @@ -2322,6 +2386,11 @@ static hda_nid_t alc260_hp_adc_nids[2] = { 0x05, 0x04 }; +static hda_nid_t alc260_fujitsu_adc_nids[2] = { + /* ADC0, ADC1 */ + 0x04, 0x05 +}; + #define ALC260_DIGOUT_NID 0x03 #define ALC260_DIGIN_NID 0x06 @@ -2339,10 +2408,11 @@ static struct hda_input_mux alc260_capture_source = { * and the internal CD lines. */ static struct hda_input_mux alc260_fujitsu_capture_source = { - .num_items = 2, + .num_items = 3, .items = { { "Mic/Line", 0x0 }, { "CD", 0x4 }, + { "Headphone", 0x2 }, }, }; @@ -2408,11 +2478,12 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PINCTL_SWITCH("Headphone Amp Switch", 0x14, PIN_HP_AMP), + ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Internal Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), @@ -2645,6 +2716,11 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + * when acting as an output. + */ + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + /* Start with mixer outputs muted */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -2654,14 +2730,27 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Unmute Line1 pin widget amp left and right (no equiv mixer ctrl) */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Line1 pin widget input for when this pin is used as input + * (no equiv mixer ctrl). Having input and output unmuted doesn't + * seem to cause a problem. + */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Unmute pin widget used for Line-in (no equiv mixer ctrl) */ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Mute capture amp left and right */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to line in (on mic1 pin) */ + /* Set ADC connection select to match default mixer setting - line + * in (on mic1 pin) + */ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to line in + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Mute all inputs to mixer widget (even unconnected ones) */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ @@ -3009,8 +3098,8 @@ static struct alc_config_preset alc260_presets[] = { .init_verbs = { alc260_fujitsu_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_adc_nids), - .adc_nids = alc260_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_fujitsu_adc_nids), + .adc_nids = alc260_fujitsu_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_fujitsu_capture_source, -- cgit 1.2.3-korg From 8b33a5aa08c4e462360d812386dcecc0467efd7b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Feb 2006 11:57:01 +0100 Subject: [ALSA] hda: minor correction to fujitsu ALC260 initverbs Modules: HDA Codec driver Jonathan Woithe The following patch changes the initverbs associated with the output sum widgets 0x08, 0x09 and 0x0a used with the fujitsu model in patch_realtek.c. It is against alsa 1.0.11rc3. Currently these widgets have their output muted, but a close reading of the datasheet suggests that they don't actually have an output mute. They have an *input* mute for each of their two inputs and a single output gain. This patch therefore activates the input mutes and zeros the output gain. It has not yet been tested but the above interpretation of the ALC260 datasheet seems correct. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c8fc6269b03c89..8ec2db2e9b6016 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2721,10 +2721,16 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { */ {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - /* Start with mixer outputs muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, -- cgit 1.2.3-korg From 7cf51e48315d87b4c1cf600d611894f45f661142 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Thu, 9 Feb 2006 12:01:26 +0100 Subject: [ALSA] hda: ALC260 test model implementation Modules: HDA Codec driver The following patch adds a 'test' ALC260 model specification to the patch_realtek.c driver if CONFIG_SND_DEBUG is set. This is similar to the 'test' ALC880 model in that it sets up mixer controls for almost everything to make it easier for people to test their laptop/soundcard when working out what pin widgets are connected to which real-world devices. This patch assumes my previous patch (adding the second PCM to the ALC260) has previously been applied since it uses infrastructure added by that patch. In developing this patch it was found that not all retasking pins accept all the modes - in particular, some ignore the VREFxx variants. The pin mode control has therefore been tweaked to prevent this becoming a problem in mixer applications. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 173 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 170 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8ec2db2e9b6016..ea4200a4ee428e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6,6 +6,7 @@ * Copyright (c) 2004 Kailang Yang * PeiSen Hou * Takashi Iwai + * Jonathan Woithe * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -63,6 +64,9 @@ enum { ALC260_HP, ALC260_HP_3013, ALC260_FUJITSU_S702X, +#ifdef CONFIG_SND_DEBUG + ALC260_TEST, +#endif ALC260_AUTO, ALC260_MODEL_LAST /* last tag */ }; @@ -223,13 +227,19 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_va * instead of "%" to avoid consequences of accidently treating the % as * being part of a format specifier. Maximum allowed length of a value is * 63 characters plus NULL terminator. + * + * Note: some retasking pin complexes seem to ignore requests for input + * states other than HiZ (eg: PIN_VREFxx) and revert to HiZ if any of these + * are requested. Therefore order this list so that this behaviour will not + * cause problems when mixer clients move through the enum sequentially. + * NIDs 0x0f and 0x10 have been observed to have this behaviour. */ static char *alc_pin_mode_names[] = { - "Line in", "Mic 80pc bias", "Mic 50pc bias", - "Line out", "Headphone out", + "Mic 50pc bias", "Mic 80pc bias", + "Line in", "Line out", "Headphone out", }; static unsigned char alc_pin_mode_values[] = { - PIN_IN, PIN_VREF80, PIN_VREF50, PIN_OUT, PIN_HP, + PIN_VREF50, PIN_VREF80, PIN_IN, PIN_OUT, PIN_HP, }; /* The control can present all 5 options, or it can limit the options based * in the pin being assumed to be exclusively an input or an output pin. @@ -2770,6 +2780,146 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { { } }; +/* Test configuration for debugging, modelled after the ALC880 test + * configuration. + */ +#ifdef CONFIG_SND_DEBUG +static hda_nid_t alc260_test_dac_nids[1] = { + 0x02, +}; +static hda_nid_t alc260_test_adc_nids[2] = { + 0x04, 0x05, +}; +static struct hda_input_mux alc260_test_capture_source = { + .num_items = 7, + .items = { + { "MIC1 pin", 0x0 }, + { "MIC2 pin", 0x1 }, + { "LINE1 pin", 0x2 }, + { "LINE2 pin", 0x3 }, + { "CD pin", 0x4 }, + { "LINE-OUT pin", 0x5 }, + { "HP-OUT pin", 0x6 }, + }, +}; +static struct snd_kcontrol_new alc260_test_mixer[] = { + /* Output driver widgets */ + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT2 Playback Volume", 0x09, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT2 Playback Switch", 0x09, 2, HDA_INPUT), + HDA_CODEC_VOLUME("LOUT1 Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("LOUT1 Playback Switch", 0x08, 2, HDA_INPUT), + + /* Modes for retasking pin widgets */ + ALC_PIN_MODE("HP-OUT pin mode", 0x10, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE-OUT pin mode", 0x0f, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE2 pin mode", 0x15, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("LINE1 pin mode", 0x14, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC2 pin mode", 0x13, ALC_PIN_DIR_INOUT), + ALC_PIN_MODE("MIC1 pin mode", 0x12, ALC_PIN_DIR_INOUT), + + /* Loopback mixer controls */ + HDA_CODEC_VOLUME("MIC1 Playback Volume", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 Playback Switch", 0x07, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("MIC2 Playback Volume", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("MIC2 Playback Switch", 0x07, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("LINE2 Playback Volume", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), + { } /* end */ +}; +static struct hda_verb alc260_test_init_verbs[] = { + /* Disable all GPIOs */ + {0x01, AC_VERB_SET_GPIO_MASK, 0}, + /* Enable retasking pins as output, initially without power amp */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* Disable digital (SPDIF) pins for now */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure mic1, mic2, line1 and line2 pin widget take input from the + * OUT1 sum bus when acting as an output. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0c, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute retasking pin widget output amp left/right (no mixer ctrl) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Also unmute the mono-out pin widget */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Also unmute the retasking pin input amps. Having the input and + * output amps unmuted at the same time doesn't appear to cause any + * trouble. + */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - line + * in (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to line in + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; +#endif + static struct hda_pcm_stream alc260_pcm_analog_playback = { .substreams = 1, .channels_min = 2, @@ -3053,6 +3203,9 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .pci_subvendor = 0x103c, .pci_subdevice = 0x3016, .config = ALC260_HP }, { .modelname = "fujitsu", .config = ALC260_FUJITSU_S702X }, { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702X }, +#ifdef CONFIG_SND_DEBUG + { .modelname = "test", .config = ALC260_TEST }, +#endif { .modelname = "auto", .config = ALC260_AUTO }, {} }; @@ -3110,6 +3263,20 @@ static struct alc_config_preset alc260_presets[] = { .channel_mode = alc260_modes, .input_mux = &alc260_fujitsu_capture_source, }, +#ifdef CONFIG_SND_DEBUG + [ALC260_TEST] = { + .mixers = { alc260_test_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_test_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_test_dac_nids), + .dac_nids = alc260_test_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_test_adc_nids), + .adc_nids = alc260_test_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_test_capture_source, + }, +#endif }; static int patch_alc260(struct hda_codec *codec) -- cgit 1.2.3-korg From bd84b0cc99d267d6512f01a55af7dbb673f784af Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Feb 2006 12:04:36 +0100 Subject: [ALSA] via82xx - Add dxs entry for FSC Amilo L7300 Modules: VIA82xx driver Added dxs_support entry for FSC Amilo L7300. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 423741371191ee..5062b053608693 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2375,6 +2375,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1695, .subdevice = 0x3005, .action = VIA_DXS_ENABLE }, /* EPoX EP-8K9A */ { .subvendor = 0x1695, .subdevice = 0x300e, .action = VIA_DXS_SRC }, /* EPoX 9HEAI */ { .subvendor = 0x16f3, .subdevice = 0x6405, .action = VIA_DXS_SRC }, /* Jetway K8M8MS */ + { .subvendor = 0x1734, .subdevice = 0x1078, .action = VIA_DXS_SRC }, /* FSC Amilo L7300 */ { .subvendor = 0x1734, .subdevice = 0x1093, .action = VIA_DXS_SRC }, /* FSC */ { .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */ { .subvendor = 0x1849, .subdevice = 0x9761, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */ -- cgit 1.2.3-korg From d08a23e2509e8e80637b4dfa5607ea00b9151b0a Mon Sep 17 00:00:00 2001 From: Ken Arromdee Date: Thu, 9 Feb 2006 13:50:26 +0100 Subject: [ALSA] ad1816a - Fix PCM trigger direction Modules: AD1816A driver Fixed the bug of capture with Shark Predator ISA resulting in: arecord: pcm_read:1196: read error: Input/output error Signed-off-by: Takashi Iwai --- sound/isa/ad1816a/ad1816a_lib.c | 15 +++++++++------ 1 file changed, 9 insertions(+), 6 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index ac0d808fff570c..fd8fe16c09ee02 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -1,4 +1,3 @@ - /* ad1816a.c - lowlevel code for Analog Devices AD1816A chip. Copyright (C) 1999-2000 by Massimo Piccioni @@ -175,7 +174,7 @@ static void snd_ad1816a_close(struct snd_ad1816a *chip, unsigned int mode) static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what, - int channel, int cmd) + int channel, int cmd, int iscapture) { int error = 0; @@ -184,10 +183,14 @@ static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what, case SNDRV_PCM_TRIGGER_STOP: spin_lock(&chip->lock); cmd = (cmd == SNDRV_PCM_TRIGGER_START) ? 0xff: 0x00; - if (what & AD1816A_PLAYBACK_ENABLE) + /* if (what & AD1816A_PLAYBACK_ENABLE) */ + /* That is not valid, because playback and capture enable + * are the same bit pattern, just to different addresses + */ + if (! iscapture) snd_ad1816a_out_mask(chip, AD1816A_PLAYBACK_CONFIG, AD1816A_PLAYBACK_ENABLE, cmd); - if (what & AD1816A_CAPTURE_ENABLE) + else snd_ad1816a_out_mask(chip, AD1816A_CAPTURE_CONFIG, AD1816A_CAPTURE_ENABLE, cmd); spin_unlock(&chip->lock); @@ -204,14 +207,14 @@ static int snd_ad1816a_playback_trigger(struct snd_pcm_substream *substream, int { struct snd_ad1816a *chip = snd_pcm_substream_chip(substream); return snd_ad1816a_trigger(chip, AD1816A_PLAYBACK_ENABLE, - SNDRV_PCM_STREAM_PLAYBACK, cmd); + SNDRV_PCM_STREAM_PLAYBACK, cmd, 0); } static int snd_ad1816a_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_ad1816a *chip = snd_pcm_substream_chip(substream); return snd_ad1816a_trigger(chip, AD1816A_CAPTURE_ENABLE, - SNDRV_PCM_STREAM_CAPTURE, cmd); + SNDRV_PCM_STREAM_CAPTURE, cmd, 1); } static int snd_ad1816a_hw_params(struct snd_pcm_substream *substream, -- cgit 1.2.3-korg From dbc967760b075aef3b181deda6639d73bbd0ad07 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 9 Feb 2006 16:29:44 +0100 Subject: [ALSA] ac97 - Remove duplicated entry in lm4550_restbl Modules: AC97 Codec Removed the duplicated entry in lm4550_restbl. Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 5c03b4b6f5c155..4d9cf37300f75a 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -2840,7 +2840,6 @@ static struct snd_ac97_res_table lm4550_restbl[] = { { AC97_MASTER_MONO, 0x001f }, { AC97_PC_BEEP, 0x001f }, /* LSB is ignored */ { AC97_PHONE, 0x001f }, - { AC97_PHONE, 0x001f }, { AC97_MIC, 0x001f }, { AC97_LINE, 0x1f1f }, { AC97_CD, 0x1f1f }, -- cgit 1.2.3-korg From 9c4be3d334808d1dedea7db0ae84759bfac8e18e Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 9 Feb 2006 20:04:16 +0100 Subject: [ALSA] no need to check pointers passed to vfree() for NULL Modules: Digigram VX core,USB generic driver There's no need to check pointers passed to vfree() for NULL. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_pcm.c | 7 +++---- sound/usb/usbaudio.c | 7 +++---- 2 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 464109e421d4ce..2195e25087b41f 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -98,10 +98,9 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) { struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - vfree(runtime->dma_area); - runtime->dma_area = NULL; - } + + vfree(runtime->dma_area); + runtime->dma_area = NULL; return 0; } diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 7b1b9f29f3779f..039f27dc1f08e6 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -720,10 +720,9 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) { struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - vfree(runtime->dma_area); - runtime->dma_area = NULL; - } + + vfree(runtime->dma_area); + runtime->dma_area = NULL; return 0; } -- cgit 1.2.3-korg From f7004f3975591f3d4cdf457f879e9ede57394500 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 10 Feb 2006 08:42:17 +0100 Subject: [ALSA] ice1712 - Delta 1010LT S/PDIF fixes Modules: ICE1712 driver See ALSA bug#1806 for details. Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/delta.c | 36 +++++++++++++++++++++++++++++++++++- sound/pci/ice1712/ice1712.c | 3 +++ 2 files changed, 38 insertions(+), 1 deletion(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 4bbf1e2ae65372..74c4d03720bb21 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -393,6 +393,37 @@ static void delta_setup_spdif(struct snd_ice1712 *ice, int rate) snd_ice1712_delta_cs8403_spdif_write(ice, tmp); } +int snd_ice1712_delta1010lt_wordclock_status_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + char reg = 0x10; // cs8427 receiver error register + struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); + + if (snd_i2c_sendbytes(ice->cs8427, ®, 1) != 1) + snd_printk(KERN_ERR "unable to send register 0x%x byte to CS8427\n", reg); + snd_i2c_readbytes(ice->cs8427, ®, 1); + ucontrol->value.integer.value[0] = (reg ? 0 : 1); + return 0; +} + +static struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_status __devinitdata = +{ + .access = (SNDRV_CTL_ELEM_ACCESS_READ), + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Word Clock Status", + .info = snd_ice1712_delta1010lt_wordclock_status_info, + .get = snd_ice1712_delta1010lt_wordclock_status_get, +}; /* * initialize the chips on M-Audio cards @@ -622,7 +653,7 @@ static int __devinit snd_ice1712_delta_init(struct snd_ice1712 *ice) static struct snd_kcontrol_new snd_ice1712_delta1010_wordclock_select __devinitdata = ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_WORD_CLOCK_SELECT, 1, 0); static struct snd_kcontrol_new snd_ice1712_delta1010lt_wordclock_select __devinitdata = -ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_1010LT_WORDCLOCK, 1, 0); +ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Sync", 0, ICE1712_DELTA_1010LT_WORDCLOCK, 0, 0); static struct snd_kcontrol_new snd_ice1712_delta1010_wordclock_status __devinitdata = ICE1712_GPIO(SNDRV_CTL_ELEM_IFACE_MIXER, "Word Clock Status", 0, ICE1712_DELTA_WORD_CLOCK_STATUS, 1, SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE); static struct snd_kcontrol_new snd_ice1712_deltadio2496_spdif_in_select __devinitdata = @@ -653,6 +684,9 @@ static int __devinit snd_ice1712_delta_add_controls(struct snd_ice1712 *ice) break; case ICE1712_SUBDEVICE_DELTA1010LT: err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_select, ice)); + if (err < 0) + return err; + err = snd_ctl_add(ice->card, snd_ctl_new1(&snd_ice1712_delta1010lt_wordclock_status, ice)); if (err < 0) return err; break; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index d6dfe001fde3d2..78098319209cf5 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1584,6 +1584,9 @@ static void snd_ice1712_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, " CAPTURE : 0x%08x\n", inl(ICEMT(ice, ROUTE_CAPTURE))); snd_iprintf(buffer, " SPDOUT : 0x%04x\n", (unsigned)inw(ICEMT(ice, ROUTE_SPDOUT))); snd_iprintf(buffer, " RATE : 0x%02x\n", (unsigned)inb(ICEMT(ice, RATE))); + snd_iprintf(buffer, " GPIO_DATA : 0x%02x\n", (unsigned)snd_ice1712_get_gpio_data(ice)); + snd_iprintf(buffer, " GPIO_WRITE_MASK : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_WRITE_MASK)); + snd_iprintf(buffer, " GPIO_DIRECTION : 0x%02x\n", (unsigned)snd_ice1712_read(ice, ICE1712_IREG_GPIO_DIRECTION)); } static void __devinit snd_ice1712_proc_init(struct snd_ice1712 * ice) -- cgit 1.2.3-korg From 253b999f5a620be81db4cfa31f76873b639ec9a2 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 15 Feb 2006 13:31:23 +0100 Subject: [ALSA] intel8x0 - wait for ICH_RESETREGS Modules: Intel8x0 driver It seems that hardware requires some time to reset bus master registers. We need to wait until ICH_RESETREGS bit is not released. The suggestion and symptom was described by Mike Gorchak . Signed-off-by: Jaroslav Kysela --- sound/pci/intel8x0.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index da024ffe96c252..ebbf2cf4ca0f46 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -2351,7 +2351,7 @@ static int snd_intel8x0_ali_chip_init(struct intel8x0 *chip, int probing) static int snd_intel8x0_chip_init(struct intel8x0 *chip, int probing) { - unsigned int i; + unsigned int i, timeout; int err; if (chip->device_type != DEVICE_ALI) { @@ -2369,6 +2369,15 @@ static int snd_intel8x0_chip_init(struct intel8x0 *chip, int probing) /* reset channels */ for (i = 0; i < chip->bdbars_count; i++) iputbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset, ICH_RESETREGS); + for (i = 0; i < chip->bdbars_count; i++) { + timeout = 100000; + while (--timeout != 0) { + if ((igetbyte(chip, ICH_REG_OFF_CR + chip->ichd[i].reg_offset) & ICH_RESETREGS) == 0) + break; + } + if (timeout == 0) + printk(KERN_ERR "intel8x0: reset of registers failed?\n"); + } /* initialize Buffer Descriptor Lists */ for (i = 0; i < chip->bdbars_count; i++) iputdword(chip, ICH_REG_OFF_BDBAR + chip->ichd[i].reg_offset, -- cgit 1.2.3-korg From 353b9e667042d6faa15a41df022bf38c949a7b2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2006 18:16:17 +0100 Subject: [ALSA] hda-codec - Add missing model entries for Intel 945 boards Modules: HDA Codec driver Added the missing entries for Intel 945 boards. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 35c2823a0a2b40..45ddf548d6fb40 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -302,6 +302,12 @@ static struct hda_board_config stac922x_cfg_tbl[] = { { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0101, .config = STAC_D945GTP3 }, /* Intel D945GTP - 3 Stack */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0202, + .config = STAC_D945GTP3 }, /* Intel D945GNT - 3 Stack, 9221 A1 */ + { .pci_subvendor = PCI_VENDOR_ID_INTEL, + .pci_subdevice = 0x0b0b, + .config = STAC_D945GTP3 }, /* Intel D945PSN - 3 Stack, 9221 A1 */ { .pci_subvendor = PCI_VENDOR_ID_INTEL, .pci_subdevice = 0x0404, .config = STAC_D945GTP5 }, /* Intel D945GTP - 5 Stack */ -- cgit 1.2.3-korg From 111d3af5f5fbf0e28570f1c01e83444d73c68a25 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2006 18:17:58 +0100 Subject: [ALSA] hda-intel - Automatic correction to single_cmd mode Modules: HDA Codec driver,HDA Intel driver Switch to single_cmd mode automatically as a fallback when CORB/RIRB communication doesn't work well. It may make the driver working on some devices with broken BIOS/ACPI support. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +++++ sound/pci/hda/hda_intel.c | 60 +++++++++++++++++++++++++++++++++-------------- 2 files changed, 49 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 208a3341ec201b..0d1566a399657d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -531,6 +531,12 @@ int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, bus->caddr_tbl[codec_addr] = codec; codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_VENDOR_ID); + if (codec->vendor_id == -1) + /* read again, hopefully the access method was corrected + * in the last read... + */ + codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, + AC_PAR_VENDOR_ID); codec->subsystem_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_SUBSYSTEM_ID); codec->revision_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_REV_ID); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index dbed2644a1920b..016fbc263e5509 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -446,8 +446,8 @@ static void azx_free_cmd_io(struct azx *chip) } /* send a command */ -static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int para) +static int azx_corb_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, + unsigned int verb, unsigned int para) { struct azx *chip = codec->bus->private_data; unsigned int wp; @@ -503,18 +503,21 @@ static void azx_update_rirb(struct azx *chip) } /* receive a response */ -static unsigned int azx_get_response(struct hda_codec *codec) +static unsigned int azx_rirb_get_response(struct hda_codec *codec) { struct azx *chip = codec->bus->private_data; int timeout = 50; while (chip->rirb.cmds) { if (! --timeout) { - if (printk_ratelimit()) - snd_printk(KERN_ERR - "azx_get_response timeout\n"); + snd_printk(KERN_ERR + "hda_intel: azx_get_response timeout, " + "switching to single_cmd mode...\n"); chip->rirb.rp = azx_readb(chip, RIRBWP); chip->rirb.cmds = 0; + /* switch to single_cmd mode */ + chip->single_cmd = 1; + azx_free_cmd_io(chip); return -1; } msleep(1); @@ -578,6 +581,36 @@ static unsigned int azx_single_get_response(struct hda_codec *codec) return (unsigned int)-1; } +/* + * The below are the main callbacks from hda_codec. + * + * They are just the skeleton to call sub-callbacks according to the + * current setting of chip->single_cmd. + */ + +/* send a command */ +static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, + int direct, unsigned int verb, + unsigned int para) +{ + struct azx *chip = codec->bus->private_data; + if (chip->single_cmd) + return azx_single_send_cmd(codec, nid, direct, verb, para); + else + return azx_corb_send_cmd(codec, nid, direct, verb, para); +} + +/* get a response */ +static unsigned int azx_get_response(struct hda_codec *codec) +{ + struct azx *chip = codec->bus->private_data; + if (chip->single_cmd) + return azx_single_get_response(codec); + else + return azx_rirb_get_response(codec); +} + + /* reset codec link */ static int azx_reset(struct azx *chip) { @@ -900,13 +933,8 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model) bus_temp.private_data = chip; bus_temp.modelname = model; bus_temp.pci = chip->pci; - if (chip->single_cmd) { - bus_temp.ops.command = azx_single_send_cmd; - bus_temp.ops.get_response = azx_single_get_response; - } else { - bus_temp.ops.command = azx_send_cmd; - bus_temp.ops.get_response = azx_get_response; - } + bus_temp.ops.command = azx_send_cmd; + bus_temp.ops.get_response = azx_get_response; if ((err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus)) < 0) return err; @@ -1308,8 +1336,7 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) for (i = 0; i < chip->pcm_devs; i++) snd_pcm_suspend_all(chip->pcm[i]); snd_hda_suspend(chip->bus, state); - if (! chip->single_cmd) - azx_free_cmd_io(chip); + azx_free_cmd_io(chip); pci_disable_device(pci); pci_save_state(pci); return 0; @@ -1347,8 +1374,7 @@ static int azx_free(struct azx *chip) azx_int_clear(chip); /* disable CORB/RIRB */ - if (! chip->single_cmd) - azx_free_cmd_io(chip); + azx_free_cmd_io(chip); /* disable position buffer */ azx_writel(chip, DPLBASE, 0); -- cgit 1.2.3-korg From 1a56f8d662ec7fc86f2c408d289fa07cdb781746 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Feb 2006 19:51:10 +0100 Subject: [ALSA] hda-intel - Auto-correction of the DMA position mode Modules: HDA Intel driver Switch the method to measure the current DMA position automatically from position-buffer mode to LPIB-read mode with a sanity check. Some hardwares seems to have problem with the position buffer. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 016fbc263e5509..c096606970ff0c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -250,7 +250,6 @@ struct azx_dev { unsigned int fragsize; /* size of each period in bytes */ unsigned int frags; /* number for period in the play buffer */ unsigned int fifo_size; /* FIFO size */ - unsigned int last_pos; /* last updated period position */ void __iomem *sd_addr; /* stream descriptor pointer */ @@ -261,10 +260,11 @@ struct azx_dev { unsigned int format_val; /* format value to be set in the controller and the codec */ unsigned char stream_tag; /* assigned stream */ unsigned char index; /* stream index */ + /* for sanity check of position buffer */ + unsigned int period_intr; unsigned int opened: 1; unsigned int running: 1; - unsigned int period_updating: 1; }; /* CORB/RIRB */ @@ -804,11 +804,10 @@ static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs) if (status & azx_dev->sd_int_sta_mask) { azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); if (azx_dev->substream && azx_dev->running) { - azx_dev->period_updating = 1; + azx_dev->period_intr++; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(azx_dev->substream); spin_lock(&chip->reg_lock); - azx_dev->period_updating = 0; } } } @@ -1119,7 +1118,6 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; else azx_dev->fifo_size = 0; - azx_dev->last_pos = 0; return hinfo->ops.prepare(hinfo, apcm->codec, azx_dev->stream_tag, azx_dev->format_val, substream); @@ -1167,10 +1165,20 @@ static snd_pcm_uframes_t azx_pcm_pointer(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); unsigned int pos; - if (chip->position_fix == POS_FIX_POSBUF) { + if (chip->position_fix == POS_FIX_POSBUF || + chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ pos = *azx_dev->posbuf; + if (chip->position_fix == POS_FIX_AUTO && + azx_dev->period_intr == 1 && ! pos) { + printk(KERN_WARNING + "hda-intel: Invalid position buffer, " + "using LPIB read method instead.\n"); + chip->position_fix = POS_FIX_NONE; + goto read_lpib; + } } else { + read_lpib: /* read LPIB */ pos = azx_sd_readl(azx_dev, SD_LPIB); if (chip->position_fix == POS_FIX_FIFO) @@ -1441,7 +1449,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, chip->irq = -1; chip->driver_type = driver_type; - chip->position_fix = position_fix ? position_fix : POS_FIX_POSBUF; + chip->position_fix = position_fix; chip->single_cmd = single_cmd; #if BITS_PER_LONG != 64 -- cgit 1.2.3-korg From 8278ca8feb2748cf02d756ac6c5b9ab2e047c84a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Feb 2006 11:57:34 +0100 Subject: [ALSA] Fix check of enable module option Fix the check of enable module option in probe of platform_device drivers. It shouldn't break the loop but just ignore if enable[i] is false. Signed-off-by: Takashi Iwai --- sound/drivers/dummy.c | 4 +++- sound/drivers/mpu401/mpu401.c | 4 +++- sound/drivers/serial-u16550.c | 4 +++- sound/drivers/virmidi.c | 4 +++- sound/isa/ad1848/ad1848.c | 4 +++- sound/isa/cs423x/cs4231.c | 4 +++- sound/isa/cs423x/cs4236.c | 4 ++-- sound/isa/es1688/es1688.c | 4 +++- sound/isa/es18xx.c | 4 ++-- sound/isa/gus/gusclassic.c | 4 +++- sound/isa/gus/gusextreme.c | 4 +++- sound/isa/gus/gusmax.c | 4 +++- sound/isa/gus/interwave.c | 4 +++- sound/isa/opl3sa2.c | 4 +++- sound/isa/sb/sb16.c | 4 ++-- sound/isa/sb/sb8.c | 4 +++- sound/isa/sgalaxy.c | 4 +++- sound/isa/wavefront/wavefront.c | 4 +++- 18 files changed, 51 insertions(+), 21 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 14e1a671b5cf71..e35fd5779a9d7b 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -669,8 +669,10 @@ static int __init alsa_card_dummy_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_DUMMY_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 915589a402ab39..9d10d79e27afca 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -240,8 +240,10 @@ static int __init alsa_card_mpu401_init(void) return err; devices = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; #ifdef CONFIG_PNP if (pnp[i]) continue; diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 112ddf70540289..13b46d12564aae 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -989,8 +989,10 @@ static int __init alsa_card_serial_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_SERIAL_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 4258723de2ab85..a3ee306239c929 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -163,8 +163,10 @@ static int __init alsa_card_virmidi_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_VIRMIDI_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c index e091bbeffd2a80..326a057f752ff7 100644 --- a/sound/isa/ad1848/ad1848.c +++ b/sound/isa/ad1848/ad1848.c @@ -187,8 +187,10 @@ static int __init alsa_card_ad1848_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_AD1848_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index ab67b5c2590d97..a30dcd96252525 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -203,8 +203,10 @@ static int __init alsa_card_cs4231_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_CS4231_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 99a42138bea047..4060918e0327f5 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -771,9 +771,9 @@ static int __init alsa_card_cs423x_init(void) if ((err = platform_driver_register(&cs423x_nonpnp_driver)) < 0) return err; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; - if (is_isapnp_selected(i)) + if (! enable[i] || is_isapnp_selected(i)) continue; device = platform_device_register_simple(CS423X_DRIVER, i, NULL, 0); diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 50d23cf3d7ccea..2b69fc829265df 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -207,8 +207,10 @@ static int __init alsa_card_es1688_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(ES1688_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 79785808dd60b9..bb709264216f91 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2381,9 +2381,9 @@ static int __init alsa_card_es18xx_init(void) if ((err = platform_driver_register(&snd_es18xx_nonpnp_driver)) < 0) return err; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; - if (is_isapnp_selected(i)) + if (! enable[i] || is_isapnp_selected(i)) continue; device = platform_device_register_simple(ES18XX_DRIVER, i, NULL, 0); diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 91c219116d7a29..26dccfea243745 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -247,8 +247,10 @@ static int __init alsa_card_gusclassic_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(GUSCLASSIC_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 239f16e6b9eec0..31dc20501d0c51 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -357,8 +357,10 @@ static int __init alsa_card_gusextreme_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(GUSEXTREME_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index d4d2b2a517d59c..cafb9b67fa721d 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -384,8 +384,10 @@ static int __init alsa_card_gusmax_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(GUSMAX_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9838d992b10166..2cacd0fa687113 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -935,8 +935,10 @@ static int __init alsa_card_interwave_init(void) if ((err = platform_driver_register(&snd_interwave_driver)) < 0) return err; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; #ifdef CONFIG_PNP if (isapnp[i]) continue; diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 9d84319785012c..56fcd8a946a485 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -949,8 +949,10 @@ static int __init alsa_card_opl3sa2_init(void) if ((err = platform_driver_register(&snd_opl3sa2_nonpnp_driver)) < 0) return err; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; #ifdef CONFIG_PNP if (isapnp[i]) continue; diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 0667bd14ad603a..5737ab76160c57 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -712,9 +712,9 @@ static int __init alsa_card_sb16_init(void) if ((err = platform_driver_register(&snd_sb16_nonpnp_driver)) < 0) return err; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; - if (is_isapnp_selected(i)) + if (! enable[i] || is_isapnp_selected(i)) continue; device = platform_device_register_simple(SND_SB16_DRIVER, i, NULL, 0); diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 60ee79cd14a367..3efa23d303c3be 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -258,8 +258,10 @@ static int __init alsa_card_sb8_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_SB8_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c index 0dbbb35b242cae..a60e66afbf9070 100644 --- a/sound/isa/sgalaxy.c +++ b/sound/isa/sgalaxy.c @@ -360,8 +360,10 @@ static int __init alsa_card_sgalaxy_init(void) return err; cards = 0; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; device = platform_device_register_simple(SND_SGALAXY_DRIVER, i, NULL, 0); if (IS_ERR(device)) { diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index fa3ab960de17c5..c0115bf9065e68 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -710,8 +710,10 @@ static int __init alsa_card_wavefront_init(void) if ((err = platform_driver_register(&snd_wavefront_driver)) < 0) return err; - for (i = 0; i < SNDRV_CARDS && enable[i]; i++) { + for (i = 0; i < SNDRV_CARDS; i++) { struct platform_device *device; + if (! enable[i]) + continue; #ifdef CONFIG_PNP if (isapnp[i]) continue; -- cgit 1.2.3-korg From e661d0ddc58740eca41313fbb01f1612ff9c7878 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Feb 2006 14:14:50 +0100 Subject: [ALSA] pcm - Move PAUSE ioctl to common ioctl handler Modules: PCM Midlevel Moved PAUSE ioctl to the common ioctl handler. A capture stream may issue PAUSE, too. Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ce1956a5528d53..01f150f0990ee7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2539,6 +2539,14 @@ static int snd_pcm_common_ioctl1(struct snd_pcm_substream *substream, return snd_pcm_drain(substream); case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); + case SNDRV_PCM_IOCTL_PAUSE: + { + int res; + snd_pcm_stream_lock_irq(substream); + res = snd_pcm_pause(substream, (int)(unsigned long)arg); + snd_pcm_stream_unlock_irq(substream); + return res; + } } snd_printd("unknown ioctl = 0x%x\n", cmd); return -ENOTTY; @@ -2619,14 +2627,6 @@ static int snd_pcm_playback_ioctl1(struct snd_pcm_substream *substream, __put_user(result, _frames); return result < 0 ? result : 0; } - case SNDRV_PCM_IOCTL_PAUSE: - { - int res; - snd_pcm_stream_lock_irq(substream); - res = snd_pcm_pause(substream, (int)(unsigned long)arg); - snd_pcm_stream_unlock_irq(substream); - return res; - } } return snd_pcm_common_ioctl1(substream, cmd, arg); } -- cgit 1.2.3-korg From 62c5549ee246fa30606f918f97c6b3cde2831292 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 22 Feb 2006 17:14:34 +0100 Subject: [ALSA] Fix sleep in atomic in virmidi driver Modules: ALSA sequencer Debug: sleeping function called from invalid context at /usr/src/linux/include/linux/rwsem.h:43 in_atomic():1, irqs_disabled():0 [] snd_seq_deliver_event+0xb4/0x1a8 [snd_seq] [] snd_seq_kernel_client_dispatch+0x6c/0x7c [snd_seq] [] snd_virmidi_output_trigger+0xca/0xe5 [snd_seq_virmidi] Signed-off-by: Takashi Iwai --- sound/core/seq/seq_virmidi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c index 14fd1a608e14ff..f4edec603b8f70 100644 --- a/sound/core/seq/seq_virmidi.c +++ b/sound/core/seq/seq_virmidi.c @@ -167,7 +167,7 @@ static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, return; /* ignored */ } if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, 0, 0) < 0) + if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) return; vmidi->event.type = SNDRV_SEQ_EVENT_NONE; } @@ -186,7 +186,7 @@ static void snd_virmidi_output_trigger(struct snd_rawmidi_substream *substream, pbuf += res; count -= res; if (vmidi->event.type != SNDRV_SEQ_EVENT_NONE) { - if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, 0, 0) < 0) + if (snd_seq_kernel_client_dispatch(vmidi->client, &vmidi->event, in_atomic(), 0) < 0) return; vmidi->event.type = SNDRV_SEQ_EVENT_NONE; } -- cgit 1.2.3-korg From 62f09c3d321c82981ae3f2dd7e32bc4c73379a7e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 27 Feb 2006 09:53:03 +0100 Subject: [ALSA] usb-audio: optimize snd_usbmidi_count_bits() Modules: USB generic driver Reduce the code size of the snd_usbmidi_count_bits() function by using simpler operations. Signed-off-by: Clemens Ladisch --- sound/usb/usbmidi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 7580339dba221f..2b9d940c8064e7 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -871,10 +871,10 @@ static int snd_usbmidi_in_endpoint_create(struct snd_usb_midi* umidi, static unsigned int snd_usbmidi_count_bits(unsigned int x) { - unsigned int bits = 0; + unsigned int bits; - for (; x; x >>= 1) - bits += x & 1; + for (bits = 0; x; ++bits) + x &= x - 1; return bits; } -- cgit 1.2.3-korg From a5a6bbd9b2481dd556236ea866424590f5f987a4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Feb 2006 17:20:41 +0100 Subject: [ALSA] cs4236 - Fix a typo Modules: CS4236+ driver Fixed a typo in snd_cs4236_put_master_digital(), resulting in silence right channel. Signed-off-by: Takashi Iwai --- sound/isa/cs423x/cs4236_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 1125ddb2b1aa82..7a5a6c71f5e489 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -644,7 +644,7 @@ static int snd_cs4236_put_master_digital(struct snd_kcontrol *kcontrol, struct s val2 = (chip->eimage[CS4236_REG(CS4236_RIGHT_MASTER)] & ~0x7f) | val2; change = val1 != chip->eimage[CS4236_REG(CS4236_LEFT_MASTER)] || val2 != chip->eimage[CS4236_REG(CS4236_RIGHT_MASTER)]; snd_cs4236_ext_out(chip, CS4236_LEFT_MASTER, val1); - snd_cs4236_ext_out(chip, CS4236_RIGHT_MASTER, val1); + snd_cs4236_ext_out(chip, CS4236_RIGHT_MASTER, val2); spin_unlock_irqrestore(&chip->reg_lock, flags); return change; } -- cgit 1.2.3-korg From 0bbbc4ca7e2459f2750bd4bd8dda2ccbf7a71f02 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 Feb 2006 17:23:46 +0100 Subject: [ALSA] opti9x - Fix compile without CONFIG_PNP Modules: Opti9xx drivers Fix compile errors without CONFIG_PNP. Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 63d96be11b2b29..65b28cbc0ebd75 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -2088,9 +2088,11 @@ static int __init alsa_card_opti9xx_init(void) int error; struct platform_device *device; +#ifdef CONFIG_PNP pnp_register_card_driver(&opti9xx_pnpc_driver); if (snd_opti9xx_pnp_is_probed) return 0; +#endif if (! is_isapnp_selected()) { error = platform_driver_register(&snd_opti9xx_driver); if (error < 0) @@ -2102,7 +2104,9 @@ static int __init alsa_card_opti9xx_init(void) } platform_driver_unregister(&snd_opti9xx_driver); } +#ifdef CONFIG_PNP pnp_unregister_card_driver(&opti9xx_pnpc_driver); +#endif #ifdef MODULE printk(KERN_ERR "no OPTi " CHIP_NAME " soundcard found\n"); #endif @@ -2115,7 +2119,9 @@ static void __exit alsa_card_opti9xx_exit(void) platform_device_unregister(snd_opti9xx_platform_device); platform_driver_unregister(&snd_opti9xx_driver); } +#ifdef CONFIG_PNP pnp_unregister_card_driver(&opti9xx_pnpc_driver); +#endif } module_init(alsa_card_opti9xx_init) -- cgit 1.2.3-korg From 2a578f3b4c5dbac4828de52e72bb79f3e9f4d1a2 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 27 Feb 2006 18:35:46 +0100 Subject: [ALSA] Don't NULL check vfree argument in pdaudiocf_pcm.c Modules: PDAudioCF driver Don't check pointers passed to vfree for null in pdaudiocf_pcm.c Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 962e6d525564a0..7f2a4de1d35d5b 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -66,10 +66,9 @@ static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t s static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) { struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - vfree(runtime->dma_area); - runtime->dma_area = NULL; - } + + vfree(runtime->dma_area); + runtime->dma_area = NULL; return 0; } -- cgit 1.2.3-korg From 17e7aec680a252ec0cea6d886d3983319af43ac4 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:35:18 +0100 Subject: [ALSA] HDA/ALC260: 1/7 - Fix test model input mux label Modules: HDA Codec driver This patch fixes input mux labels used for the ALC260 'test' model. This is needed to avoid confusion which comes about because the two ADCs in the ALC260 have slightly different mappings for their input selectors. Since this is just the test model it's sufficient to simply report both options where they exist. If a model comes along for which this becomes an issue, the ALC260 input mux code will have to be extended to allow different mux layouts for different ADCs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 20 +++++++++++++++----- 1 file changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ea4200a4ee428e..b34d2448d17266 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2414,8 +2414,8 @@ static struct hda_input_mux alc260_capture_source = { }, }; -/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack - * and the internal CD lines. +/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, + * headphone jack and the internal CD lines. */ static struct hda_input_mux alc260_fujitsu_capture_source = { .num_items = 3, @@ -2790,16 +2790,26 @@ static hda_nid_t alc260_test_dac_nids[1] = { static hda_nid_t alc260_test_adc_nids[2] = { 0x04, 0x05, }; +/* This is a bit messy since the two input muxes in the ALC260 have slight + * variations in their signal assignments. The ideal way to deal with this + * is to extend alc_spec.input_mux to allow a different input MUX for each + * ADC. For the purposes of the test model it's sufficient to just list + * both options for affected signal indices. The separate input mux + * functionality only needs to be considered if a model comes along which + * actually uses signals 0x5, 0x6 and 0x7 for something which makes sense to + * record. + */ static struct hda_input_mux alc260_test_capture_source = { - .num_items = 7, + .num_items = 8, .items = { { "MIC1 pin", 0x0 }, { "MIC2 pin", 0x1 }, { "LINE1 pin", 0x2 }, { "LINE2 pin", 0x3 }, { "CD pin", 0x4 }, - { "LINE-OUT pin", 0x5 }, - { "HP-OUT pin", 0x6 }, + { "LINE-OUT pin (cap1), Mixer (cap2)", 0x5 }, + { "HP-OUT pin (cap1), LINE-OUT pin (cap2)", 0x6 }, + { "HP-OUT pin (cap2 only)", 0x7 }, }, }; static struct snd_kcontrol_new alc260_test_mixer[] = { -- cgit 1.2.3-korg From cdcd9268a1b03c6695ea07983ce029b2b4a099d8 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:36:42 +0100 Subject: [ALSA] HDA/ALC260: 2/7 - switch pin buffer enables Modules: HDA Codec driver This patch adds functionality which switches the input/output buffer enables of retasking pins when the user changes their mode with the mode control. This probably reduces noise *slightly* for recording compared to the case where both input and output buffers were enabled simultaneously. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 58 ++++++++++++++++++++++++++++--------------- 1 file changed, 38 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b34d2448d17266..600d0a037842e1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,9 +306,32 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v val = alc_pin_mode_min(dir); change = pinctl != alc_pin_mode_values[val]; - if (change) + if (change) { + /* Set pin mode to that requested */ snd_hda_codec_write(codec,nid,0,AC_VERB_SET_PIN_WIDGET_CONTROL, alc_pin_mode_values[val]); + + /* Also enable the retasking pin's input/output as required + * for the requested pin mode. Enum values of 2 or less are + * input modes. + * + * Dynamically switching the input/output buffers probably + * reduces noise slightly, particularly on input. However, + * havingboth input and output buffers enabled + * simultaneously doesn't seem to be problematic. + */ + if (val <= 2) { + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_MUTE); + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + } else { + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_AMP_GAIN_MUTE, + AMP_OUT_UNMUTE); + } + } return change; } @@ -2744,14 +2767,16 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget amp left and right (no equiv mixer ctrl) */ + /* Unmute Line1 pin widget output buffer since it starts as an output. + * If the pin mode is changed by the user the pin mode control will + * take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget input for when this pin is used as input - * (no equiv mixer ctrl). Having input and output unmuted doesn't - * seem to cause a problem. - */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Unmute pin widget used for Line-in (no equiv mixer ctrl) */ + /* Unmute input buffer of pin widget used for Line-in (no equiv + * mixer ctrl) + */ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Mute capture amp left and right */ @@ -2882,7 +2907,11 @@ static struct hda_verb alc260_test_init_verbs[] = { {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Unmute retasking pin widget output amp left/right (no mixer ctrl) */ + /* Unmute retasking pin widget output buffers since the default + * state appears to be output. As the pin mode is changed by the + * user the pin mode control will take care of enabling the pin's + * input/output buffers as needed. + */ {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -2892,17 +2921,6 @@ static struct hda_verb alc260_test_init_verbs[] = { /* Also unmute the mono-out pin widget */ {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Also unmute the retasking pin input amps. Having the input and - * output amps unmuted at the same time doesn't appear to cause any - * trouble. - */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Mute capture amp left and right */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Set ADC connection select to match default mixer setting - line -- cgit 1.2.3-korg From d57fdac0691d500d5c697e452f769335b22a75e3 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:38:35 +0100 Subject: [ALSA] HDA/ALC260: 3/7 - generalise some structures Modules: HDA Codec driver This patch generalises a structure added for the 'fujitsu' model but which is potentially useful for other models as well. It turns the 'alc260_fujitsu_adc_nids' array into 'alc260_dual_adc_nids'; for other models which decide to utilise the dual ADC functionality there's really no reason why they need to define their own list of ADC nids. The 'fujitsu' model preset is adjusted accordingly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 600d0a037842e1..80ba6c753c335a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2419,7 +2419,10 @@ static hda_nid_t alc260_hp_adc_nids[2] = { 0x05, 0x04 }; -static hda_nid_t alc260_fujitsu_adc_nids[2] = { +/* NIDs used when simultaneous access to both ADCs makes sense. Note that + * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. + */ +static hda_nid_t alc260_dual_adc_nids[2] = { /* ADC0, ADC1 */ 0x04, 0x05 }; @@ -3285,8 +3288,8 @@ static struct alc_config_preset alc260_presets[] = { .init_verbs = { alc260_fujitsu_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, - .num_adc_nids = ARRAY_SIZE(alc260_fujitsu_adc_nids), - .adc_nids = alc260_fujitsu_adc_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, .num_channel_mode = ARRAY_SIZE(alc260_modes), .channel_mode = alc260_modes, .input_mux = &alc260_fujitsu_capture_source, -- cgit 1.2.3-korg From 5c8f858d71054960f08c46703a3f0fb36a752079 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:43:27 +0100 Subject: [ALSA] HDA/ALC260: 4/7 - add GPIO switches to test model Modules: HDA Codec driver This patch adds 'test' model mixer switches for the GPIO controls found on the ALC260. It has been found that some laptops (eg: Acer) can use these to enable particular controls, so it would be useful to have access to these via the 'test' model. It will make testing new models easy, especially if certain outputs cannot be made to work any other way. This patch *should* work, but because the GPIO pins don't do anything in my laptop I cannot personally verify that all this works as expected. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 66 +++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 64 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 80ba6c753c335a..4f6dea23d8b25d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -342,6 +342,58 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v .put = alc_pin_mode_put, \ .private_value = nid | (dir<<16) } +/* A switch control for ALC260 GPIO pins. Multiple GPIOs can be ganged + * together using a mask with more than one bit set. This control is + * currently used only by the ALC260 test model. At this stage they are not + * needed for any "production" models. + */ +#ifdef CONFIG_SND_DEBUG +static int alc_gpio_data_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} +static int alc_gpio_data_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int gpio_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_GPIO_DATA,0x00); + + /* Set/unset the masked GPIO bit(s) as needed */ + change = (val==0?0:mask) != (gpio_data & mask); + if (val==0) + gpio_data &= ~mask; + else + gpio_data |= mask; + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_GPIO_DATA,gpio_data); + + return change; +} +#define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_gpio_data_info, \ + .get = alc_gpio_data_get, \ + .put = alc_gpio_data_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + /* * set up from the preset table */ @@ -2874,11 +2926,21 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), HDA_CODEC_MUTE("HP-OUT loopback Playback Switch", 0x07, 0x7, HDA_INPUT), + + /* Controls for GPIO pins, assuming they are configured as outputs */ + ALC_GPIO_DATA_SWITCH("GPIO pin 0", 0x01, 0x01), + ALC_GPIO_DATA_SWITCH("GPIO pin 1", 0x01, 0x02), + ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), + ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), + { } /* end */ }; static struct hda_verb alc260_test_init_verbs[] = { - /* Disable all GPIOs */ - {0x01, AC_VERB_SET_GPIO_MASK, 0}, + /* Enable all GPIOs as outputs with an initial value of 0 */ + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x0f}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x00}, + {0x01, AC_VERB_SET_GPIO_MASK, 0x0f}, + /* Enable retasking pins as output, initially without power amp */ {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, -- cgit 1.2.3-korg From 0bfc90e95946ac420e2de049707232ce18ddeba9 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:45:11 +0100 Subject: [ALSA] HDA/ALC260: 5/7 - add 'acer' model Modules: HDA Codec driver This patch adds a new 'acer' model. It is based on work by Rimas and many other dedicated Acer owners over the past few weeks (see bug number 1618) - kudos to them. This adds support for a variety of Acer laptops, although we are still in the process of collecting pci/subsystem IDs to add. There are still some potentially outstanding issues: there are reports that the CD control might not yet be functional for example. However, the time is probably right to get a 'first cut' into the kernel which can be refined as more test reports come in. From the reports by various Acer owners, this code allows them to record from internal mics and external jacks. Playback also seems to work to all external jacks and the internal speaker. The 'beep' control doesn't appear functional at the moment. This patch depends on the changes made in patch 3/7. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 129 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 129 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f6dea23d8b25d..104d2a37f45a05 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -64,6 +64,7 @@ enum { ALC260_HP, ALC260_HP_3013, ALC260_FUJITSU_S702X, + ALC260_ACER, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -2504,6 +2505,19 @@ static struct hda_input_mux alc260_fujitsu_capture_source = { }, }; +/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configutation to + * the Fujitsu S702x, but jacks are marked differently. We won't allow + * retasking the Headphone jack, so it won't be available here. + */ +static struct hda_input_mux alc260_acer_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + { "CD", 0x4 }, + }, +}; + /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -2521,6 +2535,7 @@ static struct hda_channel_mode alc260_modes[1] = { * HP: base_output + input + capture_alt * HP_3013: hp_3013 + input + capture * fujitsu: fujitsu + capture + * acer: acer + capture */ static struct snd_kcontrol_new alc260_base_output_mixer[] = { @@ -2579,6 +2594,22 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc260_acer_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), + HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), + ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc260_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT), @@ -2860,6 +2891,90 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { { } }; +/* Initialisation sequence for ALC260 as configured in Acer TravelMate and + * similar laptops (adapted from Fujitsu init verbs). + */ +static struct hda_verb alc260_acer_init_verbs[] = { + /* On TravelMate laptops, GPIO 0 enables the internal speaker and + * the headphone jack. Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Internal speaker/Headphone jack is connected to Line-out pin */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Internal microphone/Mic jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Line In jack is connected to Line1 pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right (no equiv mixer ctrl) */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to line (on line1 pin) + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x02}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + /* Test configuration for debugging, modelled after the ALC880 test * configuration. */ @@ -3296,6 +3411,8 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .pci_subvendor = 0x103c, .pci_subdevice = 0x3016, .config = ALC260_HP }, { .modelname = "fujitsu", .config = ALC260_FUJITSU_S702X }, { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1326, .config = ALC260_FUJITSU_S702X }, + { .modelname = "acer", .config = ALC260_ACER }, + { .pci_subvendor = 0x1025, .pci_subdevice = 0x008f, .config = ALC260_ACER }, #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC260_TEST }, #endif @@ -3356,6 +3473,18 @@ static struct alc_config_preset alc260_presets[] = { .channel_mode = alc260_modes, .input_mux = &alc260_fujitsu_capture_source, }, + [ALC260_ACER] = { + .mixers = { alc260_acer_mixer, + alc260_capture_mixer }, + .init_verbs = { alc260_acer_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_acer_capture_source, + }, #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = { .mixers = { alc260_test_mixer, -- cgit 1.2.3-korg From f7ace40d14c46dca9a12ab753677adc17b1b906c Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:46:14 +0100 Subject: [ALSA] HDA/ALC260: 6/7 - Fujitsu/test model tweaks Modules: HDA Codec driver This patch is janitorial - it cleans up a number of cosmetic issues with the 'fujitsu' and 'test' models. Issues addressed: * spaces instead of tabs used for some indents, * clarified/corrected selected comments. Unlike a patch from earlier this week, this keeps both ADCs connected to the mic1 pin widget by default. I believe this is the better default, since some laptops don't bring the line1 widget to the outside world. Most (if not all) do bring mic1 out though, so it seems to me that this default should remain in place. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 108 +++++++++++++++++++++--------------------- 1 file changed, 54 insertions(+), 54 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 104d2a37f45a05..c8b0ec814375c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2819,28 +2819,28 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, /* Headphone/Line-out jack connects to Line1 pin; make it an output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Ensure all other unused pins are disabled and muted. - * Note: trying to set widget 0x15 to anything blocks all audio - * output for some reason, so just leave that at the default. - */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Mic/Line-in jack is connected to mic1 pin, so make it an input */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Ensure all other unused pins are disabled and muted. */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Disable digital (SPDIF) pins */ - {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, - {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure Line1 pin widget takes its input from the OUT1 sum bus - * when acting as an output. - */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + * when acting as an output. + */ + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, - /* Start with output sum widgets muted and their output gains at min */ + /* Start with output sum widgets muted and their output gains at min */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, @@ -2851,42 +2851,42 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute Line1 pin widget output buffer since it starts as an output. - * If the pin mode is changed by the user the pin mode control will - * take care of enabling the pin's input/output buffers as needed. - * Therefore there's no need to enable the input buffer at this - * stage. + /* Unmute HP pin widget amp left and right (no equiv mixer ctrl) */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Line1 pin widget output buffer since it starts as an output. + * If the pin mode is changed by the user the pin mode control will + * take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Unmute input buffer of pin widget used for Line-in (no equiv * mixer ctrl) */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - - /* Mute capture amp left and right */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - line - * in (on mic1 pin) - */ - {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Do the same for the second ADC: mute capture input amp and - * set ADC connection to line in - */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, - - /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - line + * in (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do the same for the second ADC: mute capture input amp and + * set ADC connection to line in (on mic1 pin) + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ { } }; @@ -3068,7 +3068,7 @@ static struct hda_verb alc260_test_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure mic1, mic2, line1 and line2 pin widget take input from the + /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the * OUT1 sum bus when acting as an output. */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, @@ -3103,13 +3103,13 @@ static struct hda_verb alc260_test_init_verbs[] = { /* Mute capture amp left and right */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - line - * in (on mic1 pin) + /* Set ADC connection select to match default mixer setting (mic1 + * pin) */ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Do the same for the second ADC: mute capture input amp and - * set ADC connection to line in + * set ADC connection to mic1 pin */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, -- cgit 1.2.3-korg From 92621f130e60da865de2f5aa6d3dec2ee844b803 Mon Sep 17 00:00:00 2001 From: Jonathan Woithe Date: Tue, 28 Feb 2006 11:47:47 +0100 Subject: [ALSA] HDA/ALC260: 7/7 - add SPDIF enable to test model Modules: HDA Codec driver This patch adds mixer controls to the 'test' ALC260 model which allow the user to selectively enable or disable the SPDIF output pins. This should assist people identify digital outputs on machines which bring them to the outside world. Note that while the patch *should* work, I cannot personally verify it since my laptop doesn't bring the SPDIF lines out. As for the GPIO switches added in patch 4, these controls are currently only compiled in if debug mode is selected. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 68 ++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 67 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c8b0ec814375c9..219ddf0b8d437c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -395,6 +395,60 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_ .private_value = nid | (mask<<16) } #endif /* CONFIG_SND_DEBUG */ +/* A switch control to allow the enabling of the digital IO pins on the + * ALC260. This is incredibly simplistic; the intention of this control is + * to provide something in the test model allowing digital outputs to be + * identified if present. If models are found which can utilise these + * outputs a more complete mixer control can be devised for those models if + * necessary. + */ +#ifdef CONFIG_SND_DEBUG +static int alc_spdif_ctrl_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} +static int alc_spdif_ctrl_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long *valp = ucontrol->value.integer.value; + unsigned int val = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); + + *valp = (val & mask) != 0; + return 0; +} +static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + signed int change; + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = kcontrol->private_value & 0xffff; + unsigned char mask = (kcontrol->private_value >> 16) & 0xff; + long val = *ucontrol->value.integer.value; + unsigned int ctrl_data = snd_hda_codec_read(codec,nid,0,AC_VERB_GET_DIGI_CONVERT,0x00); + + /* Set/unset the masked control bit(s) as needed */ + change = (val==0?0:mask) != (ctrl_data & mask); + if (val==0) + ctrl_data &= ~mask; + else + ctrl_data |= mask; + snd_hda_codec_write(codec,nid,0,AC_VERB_SET_DIGI_CONVERT_1,ctrl_data); + + return change; +} +#define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .info = alc_spdif_ctrl_info, \ + .get = alc_spdif_ctrl_get, \ + .put = alc_spdif_ctrl_put, \ + .private_value = nid | (mask<<16) } +#endif /* CONFIG_SND_DEBUG */ + /* * set up from the preset table */ @@ -3048,6 +3102,13 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { ALC_GPIO_DATA_SWITCH("GPIO pin 2", 0x01, 0x04), ALC_GPIO_DATA_SWITCH("GPIO pin 3", 0x01, 0x08), + /* Switches to allow the digital IO pins to be enabled. The datasheet + * is ambigious as to which NID is which; testing on laptops which + * make this output available should provide clarification. + */ + ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), + ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), + { } /* end */ }; static struct hda_verb alc260_test_init_verbs[] = { @@ -3064,7 +3125,12 @@ static struct hda_verb alc260_test_init_verbs[] = { {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Disable digital (SPDIF) pins for now */ + /* Disable digital (SPDIF) pins initially, but users can enable + * them via a mixer switch. In the case of SPDIF-out, this initverb + * payload also sets the generation to 0, output to be in "consumer" + * PCM format, copyright asserted, no pre-emphasis and no validity + * control. + */ {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, -- cgit 1.2.3-korg From 381e3cda682abcef2acd7828e9639f4ceb95880f Mon Sep 17 00:00:00 2001 From: Doug McLain Date: Tue, 28 Feb 2006 11:53:06 +0100 Subject: [ALSA] ice1712 - Fix wordclock status on Delta1010LT Modules: ICE1712 driver I posted this patch to bug 1806 a while back, and have been awaiting a reply or commit. It currently reports the Locked/No Signal text backwards in envy24control, since i was using an older version of envy24control when I wrote the original code. The Locked/No Signal test was recently reversed in envy24control cvs, so the test in my code needs to be reversed as well. Here is the patch, once again. Signed-off-by: Takashi Iwai --- sound/pci/ice1712/delta.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 74c4d03720bb21..465eeb50eb981a 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -412,7 +412,7 @@ int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kcontrol, if (snd_i2c_sendbytes(ice->cs8427, ®, 1) != 1) snd_printk(KERN_ERR "unable to send register 0x%x byte to CS8427\n", reg); snd_i2c_readbytes(ice->cs8427, ®, 1); - ucontrol->value.integer.value[0] = (reg ? 0 : 1); + ucontrol->value.integer.value[0] = (reg ? 1 : 0); return 0; } -- cgit 1.2.3-korg From 834be88d136ee82828e3ce1b34fa7a1dcf947b81 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Mar 2006 14:16:17 +0100 Subject: [ALSA] hda-codec - Fix ALC262 for Fujitsu laptop Modules: HDA Codec driver,HDA generic driver Add 'fujitsu' model for ALC262 patch to support a FSC laptop. The internal speaker is turned on/off with jack sensing. Also fixed alc262 'basic' model. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 6 +- sound/pci/hda/hda_local.h | 5 ++ sound/pci/hda/patch_realtek.c | 159 ++++++++++++++++++++++++++++++++++++++---- 3 files changed, 155 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 0d1566a399657d..fc91256e42eb6d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -729,7 +729,8 @@ static void put_vol_mute(struct hda_codec *codec, struct hda_amp_info *info, /* * read AMP value. The volume is between 0 to 0x7f, 0x80 = mute bit. */ -static int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index) +int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int index) { struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index)); if (! info) @@ -740,7 +741,8 @@ static int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch /* * update the AMP value, mask = bit mask to set, val = the value */ -static int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int mask, int val) +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int idx, int mask, int val) { struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx)); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index c82d2a72d13e4f..548a8b6984871a 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -66,6 +66,11 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +/* lowlevel accessor with caching; use carefully */ +int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int index); +int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid, int ch, + int direction, int idx, int mask, int val); /* mono switch binding multiple inputs */ #define HDA_BIND_MUTE_MONO(xname, nid, channel, indices, direction) \ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 219ddf0b8d437c..5de754a51fc71e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -75,6 +75,7 @@ enum { /* ALC262 models */ enum { ALC262_BASIC, + ALC262_FUJITSU, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -145,6 +146,10 @@ struct alc_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_imux; hda_nid_t private_dac_nids[5]; + + /* for pin sensing */ + unsigned int sense_updated: 1; + unsigned int jack_present: 1; }; /* @@ -4194,19 +4199,9 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, { } /* end */ -}; - +}; + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -4289,6 +4284,129 @@ static struct hda_verb alc262_init_verbs[] = { { } }; +/* + * fujitsu model + * 0x14 = headphone/spdif-out, 0x15 = internal speaker + */ + +#define ALC_HP_EVENT 0x37 + +static struct hda_verb alc262_fujitsu_unsol_verbs[] = { + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {} +}; + +static struct hda_input_mux alc262_fujitsu_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "CD", 0x4 }, + }, +}; + +/* mute/unmute internal speaker according to the hp jack and mute state */ +static void alc262_fujitsu_automute(struct hda_codec *codec, int force) +{ + struct alc_spec *spec = codec->spec; + unsigned int mute; + + if (force || ! spec->sense_updated) { + unsigned int present; + /* need to execute and sync at first */ + snd_hda_codec_read(codec, 0x14, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0); + spec->jack_present = (present & 0x80000000) != 0; + spec->sense_updated = 1; + } + if (spec->jack_present) { + /* mute internal speaker */ + snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + 0x80, 0x80); + snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + 0x80, 0x80); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_update(codec, 0x15, 0, HDA_OUTPUT, 0, + 0x80, mute & 0x80); + mute = snd_hda_codec_amp_read(codec, 0x14, 1, HDA_OUTPUT, 0); + snd_hda_codec_amp_update(codec, 0x15, 1, HDA_OUTPUT, 0, + 0x80, mute & 0x80); + } +} + +/* unsolicited event for HP jack sensing */ +static void alc262_fujitsu_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC_HP_EVENT) + return; + alc262_fujitsu_automute(codec, 1); +} + +/* bind volumes of both NID 0x0c and 0x0d */ +static int alc262_fujitsu_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x0c, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x0c, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x0d, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x0d, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* bind hp and internal speaker mute (with plug check) */ +static int alc262_fujitsu_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x14, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + change |= snd_hda_codec_amp_update(codec, 0x14, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + if (change || codec->in_resume) + alc262_fujitsu_automute(codec, codec->in_resume); + return change; +} + +static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = alc262_fujitsu_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_fujitsu_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { @@ -4479,6 +4597,8 @@ static int alc262_auto_init(struct hda_codec *codec) */ static struct hda_board_config alc262_cfg_tbl[] = { { .modelname = "basic", .config = ALC262_BASIC }, + { .modelname = "fujitsu", .config = ALC262_FUJITSU }, + { .pci_subvendor = 0x10cf, .pci_subdevice = 0x1397, .config = ALC262_FUJITSU }, { .modelname = "auto", .config = ALC262_AUTO }, {} }; @@ -4494,6 +4614,17 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, }, + [ALC262_FUJITSU] = { + .mixers = { alc262_fujitsu_mixer }, + .init_verbs = { alc262_init_verbs, alc262_fujitsu_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_capture_source, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -4568,7 +4699,9 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) codec->patch_ops.init = alc262_auto_init; - + if (board_config == ALC262_FUJITSU) + codec->patch_ops.unsol_event = alc262_fujitsu_unsol_event; + return 0; } -- cgit 1.2.3-korg From ed345f8f8630d6a3f52d899e181968e2bf8e4be8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Mar 2006 14:16:53 +0100 Subject: [ALSA] Update description of hda-intel models Modules: Documentation Updated the description of hda-intel models for realtek codecs. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index f28c45ca8395f0..0daba0a8c4fd97 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -703,10 +703,30 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y + auto auto-config reading BIOS (default) ALC260 hp HP machines fujitsu Fujitsu S7020 + acer Acer TravelMate + basic fixed pin assignment (old default model) + auto auto-config reading BIOS (default) + + ALC262 + fujitsu Fujitsu Laptop + basic fixed pin assignment w/o SPDIF + auto auto-config reading BIOS (default) + + ALC882/883/885 + 3stack-dig 3-jack with SPDIF I/O + 6stck-dig 6-jack digital with SPDIF I/O + auto auto-config reading BIOS (default) + + ALC861 + 3stack 3-jack + 3stack-dig 3-jack with SPDIF I/O + 6stack-dig 6-jack with SPDIF I/O + auto auto-config reading BIOS (default) CMI9880 minimal 3-jack in back -- cgit 1.2.3-korg From 42611c02ac31e44abd53893209460ce60205cb8e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Mar 2006 19:52:23 +0100 Subject: [ALSA] via82xx - Add dxs entry for ASRock mobo Modules: VIA82xx driver Add dxs_support entry for ASRock mobo. Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 5062b053608693..1957d29c119e42 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2378,6 +2378,7 @@ static int __devinit check_dxs_list(struct pci_dev *pci) { .subvendor = 0x1734, .subdevice = 0x1078, .action = VIA_DXS_SRC }, /* FSC Amilo L7300 */ { .subvendor = 0x1734, .subdevice = 0x1093, .action = VIA_DXS_SRC }, /* FSC */ { .subvendor = 0x1849, .subdevice = 0x3059, .action = VIA_DXS_NO_VRA }, /* ASRock K7VM2 */ + { .subvendor = 0x1849, .subdevice = 0x9739, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */ { .subvendor = 0x1849, .subdevice = 0x9761, .action = VIA_DXS_SRC }, /* ASRock mobo(?) */ { .subvendor = 0x1919, .subdevice = 0x200a, .action = VIA_DXS_NO_VRA }, /* Soltek SL-K8Tpro-939 */ { .subvendor = 0x4005, .subdevice = 0x4710, .action = VIA_DXS_SRC }, /* MSI K7T266 Pro2 (MS-6380 V2.0) BIOS 3.7 */ -- cgit 1.2.3-korg From 18a815d7426986890b88078ea63b77732baca0b5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Mar 2006 19:54:39 +0100 Subject: [ALSA] hda-codec - Fix AD198x recording and add HP model Modules: HDA Codec driver - Fixed recording of AD1981HD/AD1983/AD1986A - Added model 'hp' to AD1981HD for HP nx6320 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 299 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 255 insertions(+), 44 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5a3821ae93a8b7..b2479268778d54 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -310,7 +310,7 @@ static int ad198x_resume(struct hda_codec *codec) struct ad198x_spec *spec = codec->spec; int i; - ad198x_init(codec); + codec->patch_ops.init(codec); for (i = 0; i < spec->num_mixers; i++) snd_hda_resume_ctls(codec, spec->mixers[i]); if (spec->multiout.dig_out_nid) @@ -332,6 +332,53 @@ static struct hda_codec_ops ad198x_patch_ops = { }; +/* + * EAPD control + * the private value = nid | (invert << 8) + */ +static int ad198x_eapd_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + int invert = (kcontrol->private_value >> 8) & 1; + if (invert) + ucontrol->value.integer.value[0] = ! spec->cur_eapd; + else + ucontrol->value.integer.value[0] = spec->cur_eapd; + return 0; +} + +static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + int invert = (kcontrol->private_value >> 8) & 1; + hda_nid_t nid = kcontrol->private_value & 0xff; + unsigned int eapd; + eapd = ucontrol->value.integer.value[0]; + if (invert) + eapd = !eapd; + if (eapd == spec->cur_eapd && ! codec->in_resume) + return 0; + spec->cur_eapd = eapd; + snd_hda_codec_write(codec, nid, + 0, AC_VERB_SET_EAPD_BTLENABLE, + eapd ? 0x02 : 0x00); + return 1; +} + /* * AD1986A specific */ @@ -346,6 +393,7 @@ static hda_nid_t ad1986a_dac_nids[3] = { AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC }; static hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC }; +static hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 }; static struct hda_input_mux ad1986a_capture_source = { .num_items = 7, @@ -577,6 +625,7 @@ static int patch_ad1986a(struct hda_codec *codec) static hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC }; static hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC }; +static hda_nid_t ad1983_capsrc_nids[1] = { 0x15 }; static struct hda_input_mux ad1983_capture_source = { .num_items = 4, @@ -719,7 +768,7 @@ static int patch_ad1983(struct hda_codec *codec) spec->multiout.dig_out_nid = AD1983_SPDIF_OUT; spec->num_adc_nids = 1; spec->adc_nids = ad1983_adc_nids; - spec->capsrc_nids = ad1983_adc_nids; + spec->capsrc_nids = ad1983_capsrc_nids; spec->input_mux = &ad1983_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1983_mixers; @@ -743,6 +792,7 @@ static int patch_ad1983(struct hda_codec *codec) static hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC }; static hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC }; +static hda_nid_t ad1981_capsrc_nids[1] = { 0x15 }; /* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */ static struct hda_input_mux ad1981_capture_source = { @@ -848,9 +898,192 @@ static struct hda_verb ad1981_init_verbs[] = { { } /* end */ }; +/* + * Patch for HP nx6320 + * + * nx6320 uses EAPD in the reserve way - EAPD-on means the internal + * speaker output enabled _and_ mute-LED off. + */ + +#define AD1981_HP_EVENT 0x37 +#define AD1981_MIC_EVENT 0x38 + +static struct hda_verb ad1981_hp_init_verbs[] = { + {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */ + /* pin sensing on HP and Mic jacks */ + {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, + {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT}, + {} +}; + +/* turn on/off EAPD (+ mute HP) as a master switch */ +static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct ad198x_spec *spec = codec->spec; + + if (! ad198x_eapd_put(kcontrol, ucontrol)) + return 0; + + /* toggle HP mute appropriately */ + snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, + 0x80, spec->cur_eapd ? 0 : 0x80); + snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, + 0x80, spec->cur_eapd ? 0 : 0x80); + return 1; +} + +/* bind volumes of both NID 0x05 and 0x06 */ +static int ad1981_hp_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x06, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x06, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* mute internal speaker if HP is plugged */ +static void ad1981_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x06, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +/* toggle input of built-in and mic jack appropriately */ +static void ad1981_hp_automic(struct hda_codec *codec) +{ + static struct hda_verb mic_jack_on[] = { + {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {} + }; + static struct hda_verb mic_jack_off[] = { + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {} + }; + unsigned int present; + + present = snd_hda_codec_read(codec, 0x08, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) + snd_hda_sequence_write(codec, mic_jack_on); + else + snd_hda_sequence_write(codec, mic_jack_off); +} + +/* unsolicited event for HP jack sensing */ +static void ad1981_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + res >>= 26; + switch (res) { + case AD1981_HP_EVENT: + ad1981_hp_automute(codec); + break; + case AD1981_MIC_EVENT: + ad1981_hp_automic(codec); + break; + } +} + +static struct hda_input_mux ad1981_hp_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Docking-Station", 0x1 }, + { "Mix", 0x2 }, + }, +}; + +static struct snd_kcontrol_new ad1981_hp_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = ad1981_hp_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad1981_hp_master_sw_put, + .private_value = 0x05, + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), +#if 0 + /* FIXME: analog mic/line loopback doesn't work with my tests... + * (although recording is OK) + */ + HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Docking-Station Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Docking-Station Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), + /* FIXME: does this laptop have analog CD connection? */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), +#endif + HDA_CODEC_VOLUME("Mic Boost", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x18, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* initialize jack-sensing, too */ +static int ad1981_hp_init(struct hda_codec *codec) +{ + ad198x_init(codec); + ad1981_hp_automute(codec); + ad1981_hp_automic(codec); + return 0; +} + +/* models */ +enum { AD1981_BASIC, AD1981_HP }; + +static struct hda_board_config ad1981_cfg_tbl[] = { + { .modelname = "hp", .config = AD1981_HP }, + { .pci_subvendor = 0x103c, .pci_subdevice = 0x30aa, + .config = AD1981_HP }, + { .modelname = "basic", .config = AD1981_BASIC }, + {} +}; + static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; + int board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -865,7 +1098,7 @@ static int patch_ad1981(struct hda_codec *codec) spec->multiout.dig_out_nid = AD1981_SPDIF_OUT; spec->num_adc_nids = 1; spec->adc_nids = ad1981_adc_nids; - spec->capsrc_nids = ad1981_adc_nids; + spec->capsrc_nids = ad1981_capsrc_nids; spec->input_mux = &ad1981_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1981_mixers; @@ -875,6 +1108,21 @@ static int patch_ad1981(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, ad1981_cfg_tbl); + switch (board_config) { + case AD1981_HP: + spec->mixers[0] = ad1981_hp_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1981_hp_init_verbs; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1981_hp_capture_source; + + codec->patch_ops.init = ad1981_hp_init; + codec->patch_ops.unsol_event = ad1981_hp_unsol_event; + break; + } + return 0; } @@ -1062,44 +1310,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, spec->num_channel_mode, &spec->multiout.max_channels); } -/* - * EAPD control - */ -static int ad1988_eapd_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - return 0; -} - -static int ad1988_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - ucontrol->value.enumerated.item[0] = ! spec->cur_eapd; - return 0; -} - -static int ad1988_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - unsigned int eapd; - eapd = ! ucontrol->value.enumerated.item[0]; - if (eapd == spec->cur_eapd && ! codec->in_resume) - return 0; - spec->cur_eapd = eapd; - snd_hda_codec_write(codec, 0x12 /* port-D */, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 0; -} - /* 6-stack mode */ static struct snd_kcontrol_new ad1988_6stack_mixers1[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), @@ -1222,9 +1432,10 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", - .info = ad1988_eapd_info, - .get = ad1988_eapd_get, - .put = ad1988_eapd_put, + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad198x_eapd_put, + .private_value = 0x12 | (1 << 8), /* port-D, inversed */ }, { } /* end */ -- cgit 1.2.3-korg From a7ee8201a507fe1f89f0e5fc32516daed6621ffd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Mar 2006 20:05:39 +0100 Subject: [ALSA] Fix missing AD1986a capsrc Modules: HDA Codec driver Fix missing AD1986a capsrc setting in the last patch. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index b2479268778d54..4d363cfa6c8122 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -603,7 +603,7 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT; spec->num_adc_nids = 1; spec->adc_nids = ad1986a_adc_nids; - spec->capsrc_nids = ad1986a_adc_nids; + spec->capsrc_nids = ad1986a_capsrc_nids; spec->input_mux = &ad1986a_capture_source; spec->num_mixers = 1; spec->mixers[0] = ad1986a_mixers; -- cgit 1.2.3-korg From 0b7bed4ec2a16434336e018505b66bd51bb35560 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Mar 2006 15:35:55 +0100 Subject: [ALSA] Fix typos in document Modules: Documentation Fixed typos in document. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl index 4251085d38d3a1..6dc9d9f622ca62 100644 --- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl @@ -1834,7 +1834,7 @@ mychip_set_sample_format(chip, runtime->format); mychip_set_sample_rate(chip, runtime->rate); mychip_set_channels(chip, runtime->channels); - mychip_set_dma_setup(chip, runtime->dma_area, + mychip_set_dma_setup(chip, runtime->dma_addr, chip->buffer_size, chip->period_size); return 0; @@ -3388,7 +3388,7 @@ struct _snd_pcm_runtime { .name = "PCM Playback Switch", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .private_values = 0xffff, + .private_value = 0xffff, .info = my_control_info, .get = my_control_get, .put = my_control_put @@ -3449,7 +3449,7 @@ struct _snd_pcm_runtime { - The private_values field contains + The private_value field contains an arbitrary long integer value for this record. When using generic info, get and -- cgit 1.2.3-korg From b3b0abe11d606fa2344793edd3d69b98b430b0d4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Mar 2006 14:08:43 +0100 Subject: [ALSA] return ENODEV for disconnected devices Modules: ALSA Core Add dummy functions that return -ENODEV for the struct file_operations of a disconnected device. Without such functions, userspace would get ENOTTY. Signed-off-by: Clemens Ladisch --- sound/core/init.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 42 insertions(+) diff --git a/sound/core/init.c b/sound/core/init.c index 17fbd6c14fb879..ad68761abba1a8 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -169,11 +169,44 @@ struct snd_card *snd_card_new(int idx, const char *xid, return NULL; } +static loff_t snd_disconnect_llseek(struct file *file, loff_t offset, int orig) +{ + return -ENODEV; +} + +static ssize_t snd_disconnect_read(struct file *file, char __user *buf, + size_t count, loff_t *offset) +{ + return -ENODEV; +} + +static ssize_t snd_disconnect_write(struct file *file, const char __user *buf, + size_t count, loff_t *offset) +{ + return -ENODEV; +} + static unsigned int snd_disconnect_poll(struct file * file, poll_table * wait) { return POLLERR | POLLNVAL; } +static long snd_disconnect_ioctl(struct file *file, + unsigned int cmd, unsigned long arg) +{ + return -ENODEV; +} + +static int snd_disconnect_mmap(struct file *file, struct vm_area_struct *vma) +{ + return -ENODEV; +} + +static int snd_disconnect_fasync(int fd, struct file *file, int on) +{ + return -ENODEV; +} + /** * snd_card_disconnect - disconnect all APIs from the file-operations (user space) * @card: soundcard structure @@ -224,7 +257,16 @@ int snd_card_disconnect(struct snd_card *card) memset(f_ops, 0, sizeof(*f_ops)); f_ops->owner = file->f_op->owner; f_ops->release = file->f_op->release; + f_ops->llseek = snd_disconnect_llseek; + f_ops->read = snd_disconnect_read; + f_ops->write = snd_disconnect_write; f_ops->poll = snd_disconnect_poll; + f_ops->unlocked_ioctl = snd_disconnect_ioctl; +#ifdef CONFIG_COMPAT + f_ops->compat_ioctl = snd_disconnect_ioctl; +#endif + f_ops->mmap = snd_disconnect_mmap; + f_ops->fasync = snd_disconnect_fasync; s_f_ops->next = card->s_f_ops; card->s_f_ops = s_f_ops; -- cgit 1.2.3-korg From ae6b813a4dbba2713df497c032798b845289653f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Mar 2006 16:47:17 +0100 Subject: [ALSA] hda-codec - Add lg model for LG laptop Modules: Documentation,HDA Codec driver Added a new model 'lg' for LG laptop (m1 express dual) with ALC880 codec. Also clean up the initialization/unsol_event hooks in patch_realtek.c. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_realtek.c | 220 +++++++++++++++++++++--- 2 files changed, 193 insertions(+), 28 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 0daba0a8c4fd97..c12dab05176afe 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -700,6 +700,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. asus 3-jack uniwill 3-jack F1734 2-jack + lg LG laptop (m1 express dual) test for testing/debugging purpose, almost all controls can be adjusted. Appearing only when compiled with $CONFIG_SND_DEBUG=y diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5de754a51fc71e..fcab766862d4df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -51,6 +51,7 @@ enum { ALC880_UNIWILL_DIG, ALC880_CLEVO, ALC880_TCL_S700, + ALC880_LG, #ifdef CONFIG_SND_DEBUG ALC880_TEST, #endif @@ -147,6 +148,10 @@ struct alc_spec { struct hda_input_mux private_imux; hda_nid_t private_dac_nids[5]; + /* hooks */ + void (*init_hook)(struct hda_codec *codec); + void (*unsol_event)(struct hda_codec *codec, unsigned int res); + /* for pin sensing */ unsigned int sense_updated: 1; unsigned int jack_present: 1; @@ -168,6 +173,8 @@ struct alc_config_preset { unsigned int num_channel_mode; const struct hda_channel_mode *channel_mode; const struct hda_input_mux *input_mux; + void (*unsol_event)(struct hda_codec *, unsigned int); + void (*init_hook)(struct hda_codec *); }; @@ -481,6 +488,9 @@ static void setup_preset(struct alc_spec *spec, const struct alc_config_preset * spec->num_adc_nids = preset->num_adc_nids; spec->adc_nids = preset->adc_nids; spec->dig_in_nid = preset->dig_in_nid; + + spec->unsol_event = preset->unsol_event; + spec->init_hook = preset->init_hook; } /* @@ -1283,6 +1293,141 @@ static struct hda_verb alc880_pin_tcl_S700_init_verbs[] = { }; /* + * LG m1 express dual + * + * Pin assignment: + * Rear Line-In/Out (blue): 0x14 + * Build-in Mic-In: 0x15 + * Speaker-out: 0x17 + * HP-Out (green): 0x1b + * Mic-In/Out (red): 0x19 + * SPDIF-Out: 0x1e + */ + +/* To make 5.1 output working (green=Front, blue=Surr, red=CLFE) */ +static hda_nid_t alc880_lg_dac_nids[3] = { + 0x05, 0x02, 0x03 +}; + +/* seems analog CD is not working */ +static struct hda_input_mux alc880_lg_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x1 }, + { "Line", 0x5 }, + { "Internal Mic", 0x6 }, + }, +}; + +/* 2,4,6 channel modes */ +static struct hda_verb alc880_lg_ch2_init[] = { + /* set line-in and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static struct hda_verb alc880_lg_ch4_init[] = { + /* set line-in to out and mic-in to input */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { } +}; + +static struct hda_verb alc880_lg_ch6_init[] = { + /* set line-in and mic-in to output */ + { 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { } +}; + +static struct hda_channel_mode alc880_lg_ch_modes[3] = { + { 2, alc880_lg_ch2_init }, + { 4, alc880_lg_ch4_init }, + { 6, alc880_lg_ch6_init }, +}; + +static struct snd_kcontrol_new alc880_lg_mixer[] = { + /* FIXME: it's not really "master" but front channels */ + HDA_CODEC_VOLUME("Master Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x07, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, + { } /* end */ +}; + +static struct hda_verb alc880_lg_init_verbs[] = { + /* set capture source to mic-in */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* mute all amp mixer inputs */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(6)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(7)}, + /* line-in to input */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* built-in mic */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* speaker-out */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* mic-in to input */ + {0x11, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* HP-out */ + {0x13, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* jack sense */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | 0x1}, + { } +}; + +/* toggle speaker-output according to the hp-jack state */ +static void alc880_lg_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x17, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x17, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) +{ + /* Looks like the unsol event is incompatible with the standard + * definition. 4bit tag is placed at 28 bit! + */ + if ((res >> 28) == 0x01) + alc880_lg_automute(codec); +} + +/* + * Common callbacks */ static int alc_init(struct hda_codec *codec) @@ -1292,9 +1437,21 @@ static int alc_init(struct hda_codec *codec) for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); + + if (spec->init_hook) + spec->init_hook(codec); + return 0; } +static void alc_unsol_event(struct hda_codec *codec, unsigned int res) +{ + struct alc_spec *spec = codec->spec; + + if (spec->unsol_event) + spec->unsol_event(codec, res); +} + #ifdef CONFIG_PM /* * resume @@ -1531,6 +1688,7 @@ static struct hda_codec_ops alc_patch_ops = { .build_pcms = alc_build_pcms, .init = alc_init, .free = alc_free, + .unsol_event = alc_unsol_event, #ifdef CONFIG_PM .resume = alc_resume, #endif @@ -1549,13 +1707,15 @@ static hda_nid_t alc880_test_dac_nids[4] = { }; static struct hda_input_mux alc880_test_capture_source = { - .num_items = 5, + .num_items = 7, .items = { { "In-1", 0x0 }, { "In-2", 0x1 }, { "In-3", 0x2 }, { "In-4", 0x3 }, { "CD", 0x4 }, + { "Front", 0x5 }, + { "Surround", 0x6 }, }, }; @@ -1911,6 +2071,9 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1734, .pci_subdevice = 0x107c, .config = ALC880_F1734 }, { .pci_subvendor = 0x1584, .pci_subdevice = 0x9054, .config = ALC880_F1734 }, + { .modelname = "lg", .config = ALC880_LG }, + { .pci_subvendor = 0x1854, .pci_subdevice = 0x003b, .config = ALC880_LG }, + #ifdef CONFIG_SND_DEBUG { .modelname = "test", .config = ALC880_TEST }, #endif @@ -2088,6 +2251,19 @@ static struct alc_config_preset alc880_presets[] = { .channel_mode = alc880_threestack_modes, .input_mux = &alc880_capture_source, }, + [ALC880_LG] = { + .mixers = { alc880_lg_mixer }, + .init_verbs = { alc880_volume_init_verbs, + alc880_lg_init_verbs }, + .num_dacs = ARRAY_SIZE(alc880_lg_dac_nids), + .dac_nids = alc880_lg_dac_nids, + .dig_out_nid = ALC880_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc880_lg_ch_modes), + .channel_mode = alc880_lg_ch_modes, + .input_mux = &alc880_lg_capture_source, + .unsol_event = alc880_lg_unsol_event, + .init_hook = alc880_lg_automute, + }, #ifdef CONFIG_SND_DEBUG [ALC880_TEST] = { .mixers = { alc880_test_mixer }, @@ -2427,14 +2603,12 @@ static int alc880_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc880_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc880_auto_init(struct hda_codec *codec) { - alc_init(codec); alc880_auto_init_multi_out(codec); alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); - return 0; } /* @@ -2501,7 +2675,7 @@ static int patch_alc880(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC880_AUTO) - codec->patch_ops.init = alc880_auto_init; + spec->init_hook = alc880_auto_init; return 0; } @@ -3456,13 +3630,11 @@ static int alc260_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc260_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc260_auto_init(struct hda_codec *codec) { - alc_init(codec); alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); - return 0; } /* @@ -3614,7 +3786,7 @@ static int patch_alc260(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC260_AUTO) - codec->patch_ops.init = alc260_auto_init; + spec->init_hook = alc260_auto_init; return 0; } @@ -4089,14 +4261,12 @@ static int alc882_parse_auto_config(struct hda_codec *codec) return err; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc882_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc882_auto_init(struct hda_codec *codec) { - alc_init(codec); alc882_auto_init_multi_out(codec); alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); - return 0; } /* @@ -4163,7 +4333,7 @@ static int patch_alc882(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC882_AUTO) - codec->patch_ops.init = alc882_auto_init; + spec->init_hook = alc882_auto_init; return 0; } @@ -4583,13 +4753,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) /* init callback for auto-configuration model -- overriding the default init */ -static int alc262_auto_init(struct hda_codec *codec) +static void alc262_auto_init(struct hda_codec *codec) { - alc_init(codec); alc262_auto_init_multi_out(codec); alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); - return 0; } /* @@ -4624,6 +4792,7 @@ static struct alc_config_preset alc262_presets[] = { .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, .input_mux = &alc262_fujitsu_capture_source, + .unsol_event = alc262_fujitsu_unsol_event, }, }; @@ -4698,9 +4867,7 @@ static int patch_alc262(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC262_AUTO) - codec->patch_ops.init = alc262_auto_init; - if (board_config == ALC262_FUJITSU) - codec->patch_ops.unsol_event = alc262_fujitsu_unsol_event; + spec->init_hook = alc262_auto_init; return 0; } @@ -5262,15 +5429,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) return 1; } -/* init callback for auto-configuration model -- overriding the default init */ -static int alc861_auto_init(struct hda_codec *codec) +/* additional initialization for auto-configuration model */ +static void alc861_auto_init(struct hda_codec *codec) { - alc_init(codec); alc861_auto_init_multi_out(codec); alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); - - return 0; } @@ -5368,7 +5532,7 @@ static int patch_alc861(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC861_AUTO) - codec->patch_ops.init = alc861_auto_init; + spec->init_hook = alc861_auto_init; return 0; } -- cgit 1.2.3-korg From 1037593c8be9551d5a7835703bcaf6073ffb7ec2 Mon Sep 17 00:00:00 2001 From: David Vrabel Date: Fri, 3 Mar 2006 18:01:57 +0100 Subject: [ALSA] CS5535: shorter delays when accessing AC'97 codec registers Modules: CS5535 driver The 10 ms sleeps while waiting for AC'97 codec register reads/writes to complete are excessive given the maxmium time is one AC'97 frame (~21 us). With AC'97 codecs with integrated touchscreens (like the UCB1400) this improves the interactive performance of the touchscreen. Signed-off-by: David Vrabel Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 02e3721030b709..9fc7f3827461ed 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -62,7 +62,7 @@ static void wait_till_cmd_acked(struct cs5535audio *cs5535au, unsigned long time tmp = cs_readl(cs5535au, ACC_CODEC_CNTL); if (!(tmp & CMD_NEW)) break; - msleep(10); + udelay(1); } while (--timeout); if (!timeout) snd_printk(KERN_ERR "Failure writing to cs5535 codec\n"); @@ -80,14 +80,14 @@ static unsigned short snd_cs5535audio_codec_read(struct cs5535audio *cs5535au, regdata |= CMD_NEW; cs_writel(cs5535au, ACC_CODEC_CNTL, regdata); - wait_till_cmd_acked(cs5535au, 500); + wait_till_cmd_acked(cs5535au, 50); timeout = 50; do { val = cs_readl(cs5535au, ACC_CODEC_STATUS); if ((val & STS_NEW) && reg == (val >> 24)) break; - msleep(10); + udelay(1); } while (--timeout); if (!timeout) snd_printk(KERN_ERR "Failure reading cs5535 codec\n"); -- cgit 1.2.3-korg From ac57b84984859f3e1d567c031556d3de872c1a91 Mon Sep 17 00:00:00 2001 From: Alexey Dobriyan Date: Mon, 6 Mar 2006 13:21:30 +0100 Subject: [ALSA] vx - Fix memory leak on error path Modules: Digigram VX core Noticed by Eric Sesterhenn on kernel-janitors@ Signed-off-by: Alexey Dobriyan Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_pcm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 2195e25087b41f..c4af84995d0544 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -1253,9 +1253,13 @@ static int vx_init_audio_io(struct vx_core *chip) /* allocate pipes */ chip->playback_pipes = kmalloc(sizeof(struct vx_pipe *) * chip->audio_outs, GFP_KERNEL); + if (!chip->playback_pipes) + return -ENOMEM; chip->capture_pipes = kmalloc(sizeof(struct vx_pipe *) * chip->audio_ins, GFP_KERNEL); - if (! chip->playback_pipes || ! chip->capture_pipes) + if (!chip->capture_pipes) { + kfree(chip->playback_pipes); return -ENOMEM; + } memset(chip->playback_pipes, 0, sizeof(struct vx_pipe *) * chip->audio_outs); memset(chip->capture_pipes, 0, sizeof(struct vx_pipe *) * chip->audio_ins); -- cgit 1.2.3-korg From 4fcf0117d599965a5916985c9923776628e16779 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Mar 2006 13:25:14 +0100 Subject: [ALSA] emu10k1 - Add the entry for Audigy4 SB0400 Modules: EMU10K1/EMU10K2 driver Added the missing card entry for Audigy4 SB0400 model. Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 103a3f7708b786..31cb9b48bb593a 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -777,6 +777,14 @@ static int snd_emu10k1_dev_free(struct snd_device *device) static struct snd_emu_chip_details emu_chip_details[] = { /* Audigy 2 Value AC3 out does not work yet. Need to find out how to turn off interpolators.*/ + /* Audigy4 SB0400 */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x10211102, + .driver = "Audigy2", .name = "Audigy 4 [SB0400]", + .id = "Audigy2", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .ac97_chip = 1} , /* Tested by James@superbug.co.uk 3rd July 2005 */ /* DSP: CA0108-IAT * DAC: CS4382-KQ -- cgit 1.2.3-korg From a2df813beab42740fa8043b3fdc9e1d15784b9ec Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Mon, 6 Mar 2006 13:26:10 +0100 Subject: [ALSA] fix usbmixer double kfree Modules: USB generic driver snd_ctl_add() kfree's the kcontrol already if we fail there, so this driver is currently doing a double kfree. Coverity bug #959 Signed-off-by: Dave Jones Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 678dac2d4dba56..0bfea72373957e 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -434,7 +434,6 @@ static int add_control_to_empty(struct mixer_build *state, struct snd_kcontrol * kctl->id.index++; if ((err = snd_ctl_add(state->chip->card, kctl)) < 0) { snd_printd(KERN_ERR "cannot add control (err = %d)\n", err); - snd_ctl_free_one(kctl); return err; } cval->elem_id = &kctl->id; -- cgit 1.2.3-korg From c27e8c591854ef349fdf5bec777355dae04bb48f Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Mon, 6 Mar 2006 13:27:39 +0100 Subject: [ALSA] emu10k1_synth use after free Modules: EMU10K1/EMU10K2 driver Thou shalt not dereference freed memory. Coverity bug #958 Signed-off-by: Dave Jones Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_synth.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/emu10k1/emu10k1_synth.c b/sound/pci/emu10k1/emu10k1_synth.c index 1fa393f22a998f..204995a1dfbd15 100644 --- a/sound/pci/emu10k1/emu10k1_synth.c +++ b/sound/pci/emu10k1/emu10k1_synth.c @@ -62,7 +62,6 @@ static int snd_emu10k1_synth_new_device(struct snd_seq_device *dev) if (snd_emux_register(emu, dev->card, arg->index, "Emu10k1") < 0) { snd_emux_free(emu); - emu->hw = NULL; return -ENOMEM; } -- cgit 1.2.3-korg From bcc54f9a563f146e723ead16c76f842bcaeb694e Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Mon, 6 Mar 2006 13:28:34 +0100 Subject: [ALSA] sound/isa/sb/sb_mixer.c double kfree Modules: SB drivers snd_ctl_add() already does the free on error. Coverity bug #957 Signed-off-by: Dave Jones Signed-off-by: Takashi Iwai --- sound/isa/sb/sb_mixer.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 1a6ee344dddb89..490b1ca5cf5882 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -453,10 +453,8 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty strlcpy(ctl->id.name, name, sizeof(ctl->id.name)); ctl->id.index = index; ctl->private_value = value; - if ((err = snd_ctl_add(chip->card, ctl)) < 0) { - snd_ctl_free_one(ctl); + if ((err = snd_ctl_add(chip->card, ctl)) < 0) return err; - } return 0; } -- cgit 1.2.3-korg From 3de4414e798795ef5d719622dbf12bbe27a9e72e Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Mon, 6 Mar 2006 13:31:18 +0100 Subject: [ALSA] ad1848 double free Modules: AD1848 driver Same again, snd_ctl_add() already kfree's on error. Coverity #956 Signed-off-by: Dave Jones Signed-off-by: Takashi Iwai --- sound/isa/ad1848/ad1848_lib.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index d4b0e580557ede..e0f8baa843b9c0 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -1202,10 +1202,8 @@ int snd_ad1848_add_ctl(struct snd_ad1848 *chip, const char *name, int index, int strlcpy(ctl->id.name, name, sizeof(ctl->id.name)); ctl->id.index = index; ctl->private_value = value; - if ((err = snd_ctl_add(chip->card, ctl)) < 0) { - snd_ctl_free_one(ctl); + if ((err = snd_ctl_add(chip->card, ctl)) < 0) return err; - } return 0; } -- cgit 1.2.3-korg From 5e315e9248329c53a8932b71532c28240125e3c4 Mon Sep 17 00:00:00 2001 From: Dave Jones Date: Mon, 6 Mar 2006 14:03:37 +0100 Subject: [ALSA] Fix use after free in opl3_seq and opl3_oss Modules: OPL3 Don't read from free'd memory. Also make use of the return value, and don't register the device if something went wrong creating the port. Coverity #954, #955 Signed-off-by: Dave Jones Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_oss.c | 12 +++++++----- sound/drivers/opl3/opl3_seq.c | 12 +++++++++--- 2 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 0345ae64768128..fccf019a6d8596 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -104,8 +104,10 @@ static int snd_opl3_oss_create_port(struct snd_opl3 * opl3) voices, voices, name); if (opl3->oss_chset->port < 0) { + int port; + port = opl3->oss_chset->port; snd_midi_channel_free_set(opl3->oss_chset); - return opl3->oss_chset->port; + return port; } return 0; } @@ -136,10 +138,10 @@ void snd_opl3_init_seq_oss(struct snd_opl3 *opl3, char *name) arg->oper = oss_callback; arg->private_data = opl3; - snd_opl3_oss_create_port(opl3); - - /* register to OSS synth table */ - snd_device_register(opl3->card, dev); + if (snd_opl3_oss_create_port(opl3)) { + /* register to OSS synth table */ + snd_device_register(opl3->card, dev); + } } /* unregister */ diff --git a/sound/drivers/opl3/opl3_seq.c b/sound/drivers/opl3/opl3_seq.c index e26556d500e599..57becf34f43efa 100644 --- a/sound/drivers/opl3/opl3_seq.c +++ b/sound/drivers/opl3/opl3_seq.c @@ -207,8 +207,10 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) 16, voices, name); if (opl3->chset->port < 0) { + int port; + port = opl3->chset->port; snd_midi_channel_free_set(opl3->chset); - return opl3->chset->port; + return port; } return 0; } @@ -218,7 +220,7 @@ static int snd_opl3_synth_create_port(struct snd_opl3 * opl3) static int snd_opl3_seq_new_device(struct snd_seq_device *dev) { struct snd_opl3 *opl3; - int client; + int client, err; char name[32]; int opl_ver; @@ -239,7 +241,11 @@ static int snd_opl3_seq_new_device(struct snd_seq_device *dev) if (client < 0) return client; - snd_opl3_synth_create_port(opl3); + if ((err = snd_opl3_synth_create_port(opl3)) < 0) { + snd_seq_delete_kernel_client(client); + opl3->seq_client = -1; + return err; + } /* initialize instrument list */ opl3->ilist = snd_seq_instr_list_new(); -- cgit 1.2.3-korg From c1fc8047310c9d3147f9521af651a5823a72d218 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Mar 2006 14:17:31 +0100 Subject: [ALSA] Add default entry for CTL Travel Master U553W Modules: HDA Codec driver Added the default entry of ALC880 configuration table for CTL Travel Master U553W. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index fcab766862d4df..928141b1d74e5e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3644,6 +3644,8 @@ static struct hda_board_config alc260_cfg_tbl[] = { { .modelname = "basic", .config = ALC260_BASIC }, { .pci_subvendor = 0x104d, .pci_subdevice = 0x81bb, .config = ALC260_BASIC }, /* Sony VAIO */ + { .pci_subvendor = 0x152d, .pci_subdevice = 0x0729, + .config = ALC260_BASIC }, /* CTL Travel Master U553W */ { .modelname = "hp", .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3010, .config = ALC260_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x3011, .config = ALC260_HP }, -- cgit 1.2.3-korg From 0243ef71acc0b5bd734b511ae6d9b4b481c1dc5e Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Mar 2006 07:55:55 +0100 Subject: [ALSA] usb-audio: add Casio AP-80R support Modules: USB generic driver Add a quirk for the Casio AP-80R. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 7820d7bd025fd5..e310af0db1b57a 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1406,6 +1406,17 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Casio devices */ +{ + USB_DEVICE(0x07cf, 0x6802), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Casio", + .product_name = "AP-80R", + .ifnum = 0, + .type = QUIRK_MIDI_YAMAHA + } +}, + /* Mark of the Unicorn devices */ { /* thanks to Robert A. Lerche */ -- cgit 1.2.3-korg From 32e19e882c722b41f37e90956bf12eacdaba6f2b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Mar 2006 07:58:39 +0100 Subject: [ALSA] usb-audio: show USB error descriptions Modules: USB generic driver When usb_submit_urb() fails, show an error description instead of just the error code. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 39 +++++++++++++++++++++++++++++++++++---- 1 file changed, 35 insertions(+), 4 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 039f27dc1f08e6..45157d3cf7896d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -773,6 +773,31 @@ static int deactivate_urbs(struct snd_usb_substream *subs, int force, int can_sl } +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + /* * set up and start data/sync urbs */ @@ -805,16 +830,22 @@ static int start_urbs(struct snd_usb_substream *subs, struct snd_pcm_runtime *ru subs->unlink_mask = 0; subs->running = 1; for (i = 0; i < subs->nurbs; i++) { - if ((err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC)) < 0) { - snd_printk(KERN_ERR "cannot submit datapipe for urb %d, err = %d\n", i, err); + err = usb_submit_urb(subs->dataurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit datapipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); goto __error; } set_bit(i, &subs->active_mask); } if (subs->syncpipe) { for (i = 0; i < SYNC_URBS; i++) { - if ((err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC)) < 0) { - snd_printk(KERN_ERR "cannot submit syncpipe for urb %d, err = %d\n", i, err); + err = usb_submit_urb(subs->syncurb[i].urb, GFP_ATOMIC); + if (err < 0) { + snd_printk(KERN_ERR "cannot submit syncpipe " + "for urb %d, error %d: %s\n", + i, err, usb_error_string(err)); goto __error; } set_bit(i + 16, &subs->active_mask); -- cgit 1.2.3-korg From 318d27f8b424ec9e025aafe4bffa00aab5993f2f Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 9 Mar 2006 08:17:08 +0100 Subject: [ALSA] usb-audio: change Casio quirk product name Modules: USB generic driver Use a generic name for USB device 0x07cf:0x6802 because this ID is used by several devices without a product ID. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e310af0db1b57a..248a11aa8cdfc7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1408,10 +1408,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Casio devices */ { + /* this ID is used by several devices without a product ID */ USB_DEVICE(0x07cf, 0x6802), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Casio", - .product_name = "AP-80R", + .product_name = "Keyboard", .ifnum = 0, .type = QUIRK_MIDI_YAMAHA } -- cgit 1.2.3-korg From 0110f50b6eb7a833f0e1e4e9a58d04a03d58939c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 13 Mar 2006 10:36:44 +0100 Subject: [ALSA] bt87x: add more DVB card IDs Modules: BT87x driver Add more PCI subsystem IDs of DVB cards to the blacklist of cards the driver is to ignore. Signed-off-by: Clemens Ladisch --- sound/pci/bt87x.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 7f1a19c13f10b8..7b44a8db033dc0 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -795,12 +795,15 @@ static struct { unsigned short subvendor, subdevice; } blacklist[] __devinitdata = { {0x0071, 0x0101}, /* Nebula Electronics DigiTV */ + {0x11bd, 0x001c}, /* Pinnacle PCTV Sat */ {0x11bd, 0x0026}, /* Pinnacle PCTV SAT CI */ {0x1461, 0x0761}, /* AVermedia AverTV DVB-T */ {0x1461, 0x0771}, /* AVermedia DVB-T 771 */ {0x1822, 0x0001}, /* Twinhan VisionPlus DVB-T */ + {0x18ac, 0xd500}, /* DVICO FusionHDTV 5 Lite */ {0x18ac, 0xdb10}, /* DVICO FusionHDTV DVB-T Lite */ {0x270f, 0xfc00}, /* Chaintech Digitop DST-1000 DVB-S */ + {0x7063, 0x2000}, /* pcHDTV HD-2000 TV */ }; static struct pci_driver driver; -- cgit 1.2.3-korg From 9230d2148a0c53188c216b446cf17ea213ebca8a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Mar 2006 13:49:49 +0100 Subject: [ALSA] hda-codec - Fix support of laptops with AD1986A codec Modules: Documentation,HDA Codec driver Fix the support of laptops with AD1986A HD-audio codec. Added new models '3stack' and 'laptop'. Currently, fixed for FSC V2060 and Samsung M50. Also fixed the description of missing models in ALSA-Configuration.txt. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 18 ++++ sound/pci/hda/patch_analog.c | 131 ++++++++++++++++++++++++ 2 files changed, 149 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index c12dab05176afe..1065beed8d7542 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -737,6 +737,24 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. allout 5-jack in back, 2-jack in front, SPDIF out auto auto-config reading BIOS (default) + AD1981 + basic 3-jack (default) + hp HP nx6320 + + AD1986A + 6stack 6-jack, separate surrounds (default) + 3stack 3-stack, shared surrounds + laptop 2-channel only (FSC V2060, Samsung M50) + + AD1988 + 6stack 6-jack + 6stack-dig ditto with SPDIF + 3stack 3-jack + 3stack-dig ditto with SPDIF + laptop 3-jack with hp-jack automute + laptop-dig ditto with SPDIF + auto auto-confgi reading BIOS (default) + If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4d363cfa6c8122..cdcc815a9320f0 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -379,6 +379,14 @@ static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, return 1; } +static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); + + /* * AD1986A specific */ @@ -527,6 +535,52 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { { } /* end */ }; +/* additional mixers for 3stack mode */ +static struct snd_kcontrol_new ad1986a_3st_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = ad198x_ch_mode_info, + .get = ad198x_ch_mode_get, + .put = ad198x_ch_mode_put, + }, + { } /* end */ +}; + +/* laptop model - 2ch only */ +static hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; + +static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), */ + HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ + HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + /* * initialization verbs */ @@ -585,10 +639,68 @@ static struct hda_verb ad1986a_init_verbs[] = { { } /* end */ }; +/* additional verbs for 3-stack model */ +static struct hda_verb ad1986a_3st_init_verbs[] = { + /* Mic and line-in selectors */ + {0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + {0x10, AC_VERB_SET_CONNECT_SEL, 0x1}, + { } /* end */ +}; + +static struct hda_verb ad1986a_ch2_init[] = { + /* Surround out -> Line In */ + { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + /* CLFE -> Mic in */ + { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + { } /* end */ +}; + +static struct hda_verb ad1986a_ch4_init[] = { + /* Surround out -> Surround */ + { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* CLFE -> Mic in */ + { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, + { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, + { } /* end */ +}; + +static struct hda_verb ad1986a_ch6_init[] = { + /* Surround out -> Surround out */ + { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* CLFE -> CLFE */ + { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, + { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + { } /* end */ +}; + +static struct hda_channel_mode ad1986a_modes[3] = { + { 2, ad1986a_ch2_init }, + { 4, ad1986a_ch4_init }, + { 6, ad1986a_ch6_init }, +}; + +/* models */ +enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP }; + +static struct hda_board_config ad1986a_cfg_tbl[] = { + { .modelname = "6stack", .config = AD1986A_6STACK }, + { .modelname = "3stack", .config = AD1986A_3STACK }, + { .modelname = "laptop", .config = AD1986A_LAPTOP }, + { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, + .config = AD1986A_LAPTOP }, /* FSC V2060 */ + { .pci_subvendor = 0x17c0, .pci_subdevice = 0x2017, + .config = AD1986A_LAPTOP }, /* Samsung M50 */ + {} +}; static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; + int board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -612,6 +724,25 @@ static int patch_ad1986a(struct hda_codec *codec) codec->patch_ops = ad198x_patch_ops; + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, ad1986a_cfg_tbl); + switch (board_config) { + case AD1986A_3STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1986a_3st_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1986a_3st_init_verbs; + spec->channel_mode = ad1986a_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); + break; + case AD1986A_LAPTOP: + spec->mixers[0] = ad1986a_laptop_mixers; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + break; + } + return 0; } -- cgit 1.2.3-korg From 646494007b48e8897888cd407c2b7d1d69cb2e58 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Mon, 13 Mar 2006 14:11:11 +0100 Subject: [ALSA] make control.c suspend aware Modules: Control Midlevel This patch prevents user-space apps from accessing the hardware via control interface while the soundcard is suspended. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/core/control.c | 20 +++++++++++++++++--- sound/core/control_compat.c | 33 +++++++++++++++++++++++++-------- 2 files changed, 42 insertions(+), 11 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 0c29679a8576b0..9742bdba0de12c 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -658,7 +658,11 @@ static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; - result = snd_ctl_elem_info(ctl, &info); + snd_power_lock(ctl->card); + result = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0, NULL); + if (result >= 0) + result = snd_ctl_elem_info(ctl, &info); + snd_power_unlock(ctl->card); if (result >= 0) if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; @@ -708,7 +712,11 @@ static int snd_ctl_elem_read_user(struct snd_card *card, kfree(control); return -EFAULT; } - result = snd_ctl_elem_read(card, control); + snd_power_lock(card); + result = snd_power_wait(card, SNDRV_CTL_POWER_D0, NULL); + if (result >= 0) + result = snd_ctl_elem_read(card, control); + snd_power_unlock(card); if (result >= 0) if (copy_to_user(_control, control, sizeof(*control))) result = -EFAULT; @@ -758,6 +766,7 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, struct snd_ctl_elem_value __user *_control) { struct snd_ctl_elem_value *control; + struct snd_card *card; int result; control = kmalloc(sizeof(*control), GFP_KERNEL); @@ -767,7 +776,12 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, kfree(control); return -EFAULT; } - result = snd_ctl_elem_write(file->card, file, control); + card = file->card; + snd_power_lock(card); + result = snd_power_wait(card, SNDRV_CTL_POWER_D0, NULL); + if (result >= 0) + result = snd_ctl_elem_write(card, file, control); + snd_power_unlock(card); if (result >= 0) if (copy_to_user(_control, control, sizeof(*control))) result = -EFAULT; diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index a529b62972b46e..84fef5084e1758 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -107,7 +107,13 @@ static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl, */ if (get_user(data->value.enumerated.item, &data32->value.enumerated.item)) goto error; - err = snd_ctl_elem_info(ctl, data); + + snd_power_lock(ctl->card); + err = snd_power_wait(ctl->card, SNDRV_CTL_POWER_D0, NULL); + if (err >= 0) + err = snd_ctl_elem_info(ctl, data); + snd_power_unlock(ctl->card); + if (err < 0) goto error; /* restore info to 32bit */ @@ -286,9 +292,14 @@ static int snd_ctl_elem_read_user_compat(struct snd_card *card, if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) goto error; - if ((err = snd_ctl_elem_read(card, data)) < 0) - goto error; - err = copy_ctl_value_to_user(data32, data, type, count); + + snd_power_lock(card); + err = snd_power_wait(card, SNDRV_CTL_POWER_D0, NULL); + if (err >= 0) + err = snd_ctl_elem_read(card, data); + snd_power_unlock(card); + if (err >= 0) + err = copy_ctl_value_to_user(data32, data, type, count); error: kfree(data); return err; @@ -298,17 +309,23 @@ static int snd_ctl_elem_write_user_compat(struct snd_ctl_file *file, struct snd_ctl_elem_value32 __user *data32) { struct snd_ctl_elem_value *data; + struct snd_card *card = file->card; int err, type, count; data = kzalloc(sizeof(*data), GFP_KERNEL); if (data == NULL) return -ENOMEM; - if ((err = copy_ctl_value_from_user(file->card, data, data32, &type, &count)) < 0) - goto error; - if ((err = snd_ctl_elem_write(file->card, file, data)) < 0) + if ((err = copy_ctl_value_from_user(card, data, data32, &type, &count)) < 0) goto error; - err = copy_ctl_value_to_user(data32, data, type, count); + + snd_power_lock(card); + err = snd_power_wait(card, SNDRV_CTL_POWER_D0, NULL); + if (err >= 0) + err = snd_ctl_elem_write(card, file, data); + snd_power_unlock(card); + if (err >= 0) + err = copy_ctl_value_to_user(data32, data, type, count); error: kfree(data); return err; -- cgit 1.2.3-korg From 3a63e44420932852efd6a7d6d46cdad4d427f880 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Mon, 13 Mar 2006 14:14:10 +0100 Subject: [ALSA] sound/core/: fix 3 off-by-one errors Modules: ALSA Core This patch fixes three off-by-one errors found by the Coverity checker. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/core/sound.c | 4 ++-- sound/core/sound_oss.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/sound.c b/sound/core/sound.c index df4ab94d006e0f..4d28e52126118b 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -121,7 +121,7 @@ void *snd_lookup_minor_data(unsigned int minor, int type) struct snd_minor *mreg; void *private_data; - if (minor > ARRAY_SIZE(snd_minors)) + if (minor >= ARRAY_SIZE(snd_minors)) return NULL; mutex_lock(&sound_mutex); mreg = snd_minors[minor]; @@ -140,7 +140,7 @@ static int snd_open(struct inode *inode, struct file *file) struct file_operations *old_fops; int err = 0; - if (minor > ARRAY_SIZE(snd_minors)) + if (minor >= ARRAY_SIZE(snd_minors)) return -ENODEV; mptr = snd_minors[minor]; if (mptr == NULL) { diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 6b4a4bccd8fee4..4023d3b406de2c 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -46,7 +46,7 @@ void *snd_lookup_oss_minor_data(unsigned int minor, int type) struct snd_minor *mreg; void *private_data; - if (minor > ARRAY_SIZE(snd_oss_minors)) + if (minor >= ARRAY_SIZE(snd_oss_minors)) return NULL; mutex_lock(&sound_oss_mutex); mreg = snd_oss_minors[minor]; -- cgit 1.2.3-korg From 5bab2482083077d1e14644db2546c54534156849 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Mon, 13 Mar 2006 14:15:04 +0100 Subject: [ALSA] sound/pci/rme9652/hdspm.c: fix off-by-one errors Modules: RME9652 driver This patch fixes off-by-one errors found by the Coverity checker. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 103b4d715ff451..980b9cd689dd15 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -474,7 +474,7 @@ static inline unsigned int hdspm_read(struct hdspm * hdspm, unsigned int reg) static inline int hdspm_read_in_gain(struct hdspm * hdspm, unsigned int chan, unsigned int in) { - if (chan > HDSPM_MIXER_CHANNELS || in > HDSPM_MIXER_CHANNELS) + if (chan >= HDSPM_MIXER_CHANNELS || in >= HDSPM_MIXER_CHANNELS) return 0; return hdspm->mixer->ch[chan].in[in]; @@ -483,7 +483,7 @@ static inline int hdspm_read_in_gain(struct hdspm * hdspm, unsigned int chan, static inline int hdspm_read_pb_gain(struct hdspm * hdspm, unsigned int chan, unsigned int pb) { - if (chan > HDSPM_MIXER_CHANNELS || pb > HDSPM_MIXER_CHANNELS) + if (chan >= HDSPM_MIXER_CHANNELS || pb >= HDSPM_MIXER_CHANNELS) return 0; return hdspm->mixer->ch[chan].pb[pb]; } -- cgit 1.2.3-korg From ede3531e8ce2deb81e477e18d72ef10efeb20ebe Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Mar 2006 14:21:16 +0100 Subject: [ALSA] hda-codec - Fix Aopen i915GMm-HFS mobo Modules: HDA Codec driver Add a model entry for Aopen i915GMm-HFS mobo with ALC880 codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 928141b1d74e5e..0697c9bd45cc3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2022,6 +2022,8 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x8086, .pci_subdevice = 0xa100, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1565, .pci_subdevice = 0x8202, .config = ALC880_5ST_DIG }, { .pci_subvendor = 0x1019, .pci_subdevice = 0xa880, .config = ALC880_5ST_DIG }, + { .pci_subvendor = 0xa0a0, .pci_subdevice = 0x0560, + .config = ALC880_5ST_DIG }, /* Aopen i915GMm-HFS */ /* { .pci_subvendor = 0x1019, .pci_subdevice = 0xa884, .config = ALC880_5ST_DIG }, */ /* conflict with 6stack */ { .pci_subvendor = 0x1695, .pci_subdevice = 0x400d, .config = ALC880_5ST_DIG }, /* note subvendor = 0 below */ -- cgit 1.2.3-korg From 3e964432f56f05a8ef639902796e476456b230ad Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 14 Mar 2006 08:06:12 +0100 Subject: [ALSA] usb-audio: add error message about missing split iso support Modules: USB generic driver Add an error message for -ENOSYS for situations when split iso support is needed but not enabled. Signed-off-by: Clemens Ladisch --- sound/usb/usbaudio.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 45157d3cf7896d..6fad2c40c77cf4 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -788,6 +788,10 @@ static const char *usb_error_string(int err) return "device disabled"; case -EHOSTUNREACH: return "device suspended"; +#ifndef CONFIG_USB_EHCI_SPLIT_ISO + case -ENOSYS: + return "enable CONFIG_USB_EHCI_SPLIT_ISO to play through a hub"; +#endif case -EINVAL: case -EAGAIN: case -EFBIG: -- cgit 1.2.3-korg From 202728d783a0fc180e7141d18186eeae167218a1 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 14 Mar 2006 09:44:19 +0100 Subject: [ALSA] fix some memory leaks Modules: Generic drivers,ES18xx driver,CS46xx driver This patch fixes two memory leaks spotted by the Coverity checker. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/drivers/serial-u16550.c | 1 + sound/isa/es18xx.c | 1 + sound/pci/cs46xx/dsp_spos.c | 10 +++++++--- 3 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 13b46d12564aae..1a7fbefe4740de 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -789,6 +789,7 @@ static int __init snd_uart16550_create(struct snd_card *card, if ((err = snd_uart16550_detect(uart)) <= 0) { printk(KERN_ERR "no UART detected at 0x%lx\n", iobase); + snd_uart16550_free(uart); return -ENODEV; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index bb709264216f91..721955d26194b2 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2083,6 +2083,7 @@ static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, err = pnp_activate_dev(acard->devc); if (err < 0) { snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); + kfree(cfg); return -EAGAIN; } snd_printdd("pnp: port=0x%lx\n", pnp_port_start(acard->devc, 0)); diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 8726a68051e793..f407d2a5ce3b6a 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -237,7 +237,7 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) if (ins->symbol_table.symbols == NULL) { cs46xx_dsp_spos_destroy(chip); - return NULL; + goto error; } ins->code.offset = 0; @@ -246,7 +246,7 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) if (ins->code.data == NULL) { cs46xx_dsp_spos_destroy(chip); - return NULL; + goto error; } ins->nscb = 0; @@ -257,7 +257,7 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) if (ins->modules == NULL) { cs46xx_dsp_spos_destroy(chip); - return NULL; + goto error; } /* default SPDIF input sample rate @@ -280,6 +280,10 @@ struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip) /* left and right validity bits */ (1 << 13) | (1 << 12); return ins; + +error: + kfree(ins); + return NULL; } void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) -- cgit 1.2.3-korg From ecefb192525c369dde67bf0addd4f96692a801c3 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Tue, 14 Mar 2006 11:16:26 +0100 Subject: [ALSA] sound/pci/ice1712/delta.c: make 2 functions static Modules: ICE1712 driver This patch makes two needlessly global functions static. Signed-off-by: Adrian Bunk Signed-off-by: Takashi Iwai --- sound/pci/ice1712/delta.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 465eeb50eb981a..af659800c9b040 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -393,7 +393,7 @@ static void delta_setup_spdif(struct snd_ice1712 *ice, int rate) snd_ice1712_delta_cs8403_spdif_write(ice, tmp); } -int snd_ice1712_delta1010lt_wordclock_status_info(struct snd_kcontrol *kcontrol, +static int snd_ice1712_delta1010lt_wordclock_status_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -403,7 +403,7 @@ int snd_ice1712_delta1010lt_wordclock_status_info(struct snd_kcontrol *kcontrol, return 0; } -int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kcontrol, +static int snd_ice1712_delta1010lt_wordclock_status_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { char reg = 0x10; // cs8427 receiver error register -- cgit 1.2.3-korg From e311334ee6bdd173d53be52f4fdffa5f39652e26 Mon Sep 17 00:00:00 2001 From: Thibault LE MEUR Date: Tue, 14 Mar 2006 11:44:53 +0100 Subject: [ALSA] Fixes audiophile usb analog capture with the new device_setup parameter Modules: Documentation,USB generic driver The patch adds the 'device_setup' module parameter and a specific quirk to correctly initialize the audiophile usb device: this fixes the distorted sound bug on the Analog capture port. Backward compatibility is achieved by simply omitting the new parameter. Signed-off-by: Thibault LE MEUR Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 + Documentation/sound/alsa/Audiophile-Usb.txt | 330 ++++++++++++++++++++++++ sound/usb/usbaudio.c | 52 +++- 3 files changed, 384 insertions(+), 1 deletion(-) create mode 100644 Documentation/sound/alsa/Audiophile-Usb.txt diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 1065beed8d7542..f947c4b04ab8b6 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1411,6 +1411,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. vid - Vendor ID for the device (optional) pid - Product ID for the device (optional) + device_setup - Device specific magic number (optional) + - Influence depends on the device + - Default: 0x0000 This module supports multiple devices, autoprobe and hotplugging. diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt new file mode 100644 index 00000000000000..3ba45adbf04052 --- /dev/null +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -0,0 +1,330 @@ + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.1 + ======================================================== + + Thibault Le Meur + +This document is a guide to using the M-Audio Audiophile USB (tm) device with +ALSA and JACK. + +1 - Audiophile USB Specs and correct usage +========================================== +This part is a reminder of important facts about the functions and limitations +of the device. + +The device has 4 audio interfaces, and 2 MIDI ports: + * Analog Stereo Input (Ai) + * Analog Stereo Output (Ao) + * Digital Stereo Input (Di) + * Digital Stereo Output (Do) + * Midi In (Mi) + * Midi Out (Mo) + +The internal DAC/ADC has the following caracteristics: +* sample depth of 16 or 24 bits +* sample rate from 8kHz to 96kHz +* Two ports can't use different sample depths at the same time.Moreover, the +Audiophile USB documentation gives the following Warning: "Please exit any +audio application running before switching between bit depths" + +Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be +activated at the same time depending on the audio mode selected: + * 16-bit/48kHz ==> 4 channels in/ 4 channels out + - Ai+Ao+Di+Do + * 24-bit/48kHz ==> 4 channels in/2 channels out, + or 2 channels in/4 channels out + - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do + * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only) + - Ai or Ao or Di or Do + +Important facts about the Digital interface: +-------------------------------------------- + * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, +though I haven't tested it under linux + - Note that in this setup only the Do interface can be enabled + * Apart from recording an audio digital stream, enabling the Di port is a way +to syncrhonize the device to an external sample clock + - As a consequence, the Di port must be enable only if an active Digital +source is connected + - Enabling Di when no digital source is connected can result in a +synchronization error (for instance sound played at an odd sample rate) + + +2 - Audiophile USB support in ALSA +================================== + +2.1 - MIDI ports +---------------- +The Audiophile USB MIDI ports will be automatically supported once the +following modules have been loaded: + * snd-usb-audio + * snd-seq + * snd-seq-midi + +No additionnal setting is required. + +2.2 - Audio ports +----------------- + +Audio functions of the Audiophile USB device are handled by the snd-usb-audio +module. This module can work in a default mode (without any device-specific +parameter), or in an advanced mode with the device-specific parameter called +"device_setup". + +2.2.1 - Default Alsa driver mode + +The default behaviour of the snd-usb-audio driver is to parse the device +capabilities at startup and enable all functions inside the device (including +all ports at any sample rates and any sample depths supported). This approach +has the advantage to let the driver easily switch from sample rates/depths +automatically according to the need of the application claiming the device. + +In this case the Audiophile ports are mapped to alsa pcm devices in the +following way (I suppose the device's index is 1): + * hw:1,0 is Ao in playback and Di in capture + * hw:1,1 is Do in playback and Ai in capture + * hw:1,2 is Do in AC3/DTS passthrough mode + +You must note as well that the device uses Big Endian byte encoding so that +supported audio format are S16_BE for 16-bit depth modes and S24_3BE for +24-bits depth mode. One exception is the hw:1,2 port which is Little Endian +compliant and thus uses S16_LE. + +Examples: + * playing a S24_3BE encoded raw file to the Ao port + % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw + * recording a S24_3BE encoded raw file from the Ai port + % arecord -D hw:1,1 -c2 -t raw -r48000 -fS24_3BE test.raw + * playing a S16_BE encoded raw file to the Do port + % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw + +If you're happy with the default Alsa driver setup and don't experience any +issue with this mode, then you can skip the following chapter. + +2.2.2 - Advanced module setup + +Due to the hardware constraints described above, the device initialization made +by the Alsa driver in default mode may result in a corrupted state of the +device. For instance, a particularly annoying issue is that the sound captured +from the Ai port sounds distorted (as if boosted with an excessive high volume +gain). + +For people having this problem, the snd-usb-audio module has a new module +parameter called "device_setup". + +2.2.2.1 - Initializing the working mode of the Audiohile USB + +As far as the Audiohile USB device is concerned, this value let the user +specify: + * the sample depth + * the sample rate + * whether the Di port is used or not + +Here is a list of supported device_setup values for this device: + * device_setup=0x00 (or omitted) + - Alsa driver default mode + - maintains backward compatibility with setups that do not use this + parameter by not introducing any change + - results sometimes in corrupted sound as decribed earlier + * device_setup=0x01 + - 16bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + * device_setup=0x11 + - 16bits 48kHz mode with Di enabled + - Ai,Ao,Di,Do can be used at the same time + - hw:1,0 is available in capture mode + - hw:1,2 is not available + * device_setup=0x09 + - 24bits 48kHz mode with Di disabled + - Ai,Ao,Do can be used at the same time + - hw:1,0 is not available in capture mode + - hw:1,2 is not available + * device_setup=0x19 + - 24bits 48kHz mode with Di enabled + - 3 ports from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in capture mode and an active digital source must be + connected to Di + - hw:1,2 is not available + * device_setup=0x0D or 0x10 + - 24bits 96kHz mode + - Di is enabled by default for this mode but does not need to be connected + to an active source + - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time + - hw:1,0 is available in captured mode + - hw:1,2 is not available + * device_setup=0x03 + - 16bits 48kHz mode with only the Do port enabled + - AC3 with DTS passthru (not tested) + - Caution with this setup the Do port is mapped to the pcm device hw:1,0 + +2.2.2.2 - Setting and switching configurations with the device_setup parameter + +The parameter can be given: + * By manually probing the device (as root): + # modprobe -r snd-usb-audio + # modprobe snd-usb-audio index=1 device_setup=0x09 + * Or while configuring the modules options in your modules configuration file + - For Fedora distributions, edit the /etc/modprobe.conf file: + alias snd-card-1 snd-usb-audio + options snd-usb-audio index=1 device_setup=0x09 + +IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: +------------------------------------------- + * You may need to _first_ intialize the module with the correct device_setup + parameter and _only_after_ turn on the Audiophile USB device + * This is especially true when switching the sample depth: + - first trun off the device + - de-register the snd-usb-audio module + - change the device_setup parameter (by either manually reprobing the module + or changing modprobe.conf) + - turn on the device + +2.2.2.3 - Setting and switching configurations with the device_setup parameter + +If you want to understand the device_setup magic numbers for the Audiophile +USB, you need some very basic understanding of binary computation. However, +this is not required to use the parameter and you may skip thi section. + +The device_setup is one byte long and its structure is the following: + + +---+---+---+---+---+---+---+---+ + | b7| b6| b5| b4| b3| b2| b1| b0| + +---+---+---+---+---+---+---+---+ + | 0 | 0 | 0 | Di|24B|96K|DTS|SET| + +---+---+---+---+---+---+---+---+ + +Where: + * b0 is the "SET" bit + - it MUST be set if device_setup is initialized + * b1 is the "DTS" bit + - it is set only for Digital output with DTS/AC3 + - this setup is not tested + * b2 is the Rate selection flag + - When set to "1" the rate range is 48.1-96kHz + - Otherwise the sample rate range is 8-48kHz + * b3 is the bit depth selection flag + - When set to "1" samples are 24bits long + - Otherwise they are 16bits long + - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits + samples + * b4 is the Digital input flag + - When set to "1" the device assumes that an active digital source is + connected + - You shouldn't enable Di if no source is seen on the port (this leads to + synchronization issues) + - b4 is implied by b2 (since only one port is enabled at a time no synch + error can occur) + * b5 to b7 are reserved for future uses, and must be set to "0" + - might become Ao, Do, Ai, for b7, b6, b4 respectively + +Caution: + * there is no check on the value you will give to device_setup + - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since + b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages + * Hardware constraints due to the USB bus limitation aren't checked + - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll + only be able to use one at the same time + +2.2.3 - Technical Details for Audiophile Usb + +You may safely skip this section if you're not interrested in driver +development. + +This section describes some internals aspect of the device and summarize the +data I got by usb-snooping the windows and linux drivers. + +The M-Audio Audiophile USB has 7 Usb Interfaces: +a "USB interface": + * Usb Interface nb.0 + * Usb Interface nb.1 + - Audio Control function + * Usb Interface nb.2 + - Analog Output + * Usb Interface nb.3 + - Digital Output + * Usb Interface nb.4 + - Analog Input + * Usb Interface nb.5 + - Digital Input + * Usb Interface nb.6 + - MIDI interface compliant with the MIDIMAN quirk + +Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: + * Interface 3 (Digital Out) has an extra Alset nb.6 + * Interface 5 (Digital In) does not have Alset nb.3 and 5 + +Here is a short description of the AltSettings capabilities: + * AltSettings 1 corresponds to + - 24-bit depth, 48.1-96kHz sample mode + - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di) + * AltSettings 2 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 3 corresponds to + - 24-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 4 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Asynch capture and playback (Ao,Ai,Do,Di) + * AltSettings 5 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch capture (Ai) and Adaptive playback (Ao,Do) + * AltSettings 6 corresponds to + - 16-bit depth, 8-48kHz sample mode + - Synch playback (Do), audio format type III IEC1937_AC-3 + +In order to ensure a correct intialization of the device, the driver +_must_know_ how the device will be used: + * if DTS is choosen, only Interface 2 with AltSet nb.6 must be + registered + * if 96KHz only AltSets nb.1 of each interface must be selected + * if samples are using 24bits/48KHz then AltSet 2 must me used if + Digital input is connected, and only AltSet nb.3 if Digital input + is not connected + * if samples are using 16bits/48KHz then AltSet 4 must me used if + Digital input is connected, and only AltSet nb.5 if Digital input + is not connected + +When device_setup is given as a parameter to the snd-usb-audio module, the +parse_audio_enpoint function uses a quirk called +"audiophile_skip_setting_quirk" in order to prevent AltSettings not +corresponding to device_setup from being registered in the driver. + +3 - Audiophile USB and Jack support +=================================== + +This section deals with support of the Audiophile USB device in Jack. +The main issue regarding this support is that the device is Big Endian +compliant. + +3.1 - Using the plug alsa plugin +-------------------------------- + +Jack doesn't directly support big endian devices. Thus, one way to have support +for this device with Alsa is to use the Alsa "plug" converter. + +For instance here is one way to run Jack with 2 playback channels on Ao and 2 +capture channels from Ai: + % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1 + + +However you may see the following warning message: +"You appear to be using the ALSA software "plug" layer, probably a result of +using the "default" ALSA device. This is less efficient than it could be. +Consider using a hardware device instead rather than using the plug layer." + + +3.2 - Patching alsa to use direct pcm device +------------------------------------------- +A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. +However it has not been included in the CVS tree. + +You can find it at the following URL: +http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687& +atid=425939 + +After having applied the patch you can run jackd with the following command +line: +# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 6fad2c40c77cf4..4e614ac39f2194 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -70,6 +70,7 @@ static int vid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Vendor ID for static int pid[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = -1 }; /* Product ID for this card */ static int nrpacks = 4; /* max. number of packets per urb */ static int async_unlink = 1; +static int device_setup[SNDRV_CARDS]; /* device parameter for this card*/ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the USB audio adapter."); @@ -85,6 +86,8 @@ module_param(nrpacks, int, 0644); MODULE_PARM_DESC(nrpacks, "Max. number of packets per URB."); module_param(async_unlink, bool, 0444); MODULE_PARM_DESC(async_unlink, "Use async unlink mode."); +module_param_array(device_setup, int, NULL, 0444); +MODULE_PARM_DESC(device_setup, "Specific device setup (if needed)."); /* @@ -2547,6 +2550,8 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp return 0; } +static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { struct usb_device *dev; @@ -2581,6 +2586,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; + + /* audiophile usb: skip altsets incompatible with device_setup + */ + if (chip->usb_id == USB_ID(0x0763, 0x2003) && + audiophile_skip_setting_quirk(chip, iface_no, altno)) + continue; /* get audio formats */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); @@ -2675,7 +2686,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, i, fp->endpoint); + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint); err = add_audio_endpoint(chip, stream, fp); if (err < 0) { kfree(fp->rate_table); @@ -3083,6 +3094,45 @@ static int snd_usb_audigy2nx_boot_quirk(struct usb_device *dev) return 0; } +/* + * Setup quirks + */ +#define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */ +#define AUDIOPHILE_SET_DTS 0x02 /* if set, enable DTS Digital Output */ +#define AUDIOPHILE_SET_96K 0x04 /* 48-96KHz rate if set, 8-48KHz otherwise */ +#define AUDIOPHILE_SET_24B 0x08 /* 24bits sample if set, 16bits otherwise */ +#define AUDIOPHILE_SET_DI 0x10 /* if set, enable Digital Input */ +#define AUDIOPHILE_SET_MASK 0x1F /* bit mask for setup value */ +#define AUDIOPHILE_SET_24B_48K_DI 0x19 /* value for 24bits+48KHz+Digital Input */ +#define AUDIOPHILE_SET_24B_48K_NOTDI 0x09 /* value for 24bits+48KHz+No Digital Input */ +#define AUDIOPHILE_SET_16B_48K_DI 0x11 /* value for 16bits+48KHz+Digital Input */ +#define AUDIOPHILE_SET_16B_48K_NOTDI 0x01 /* value for 16bits+48KHz+No Digital Input */ + +static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, + int iface, int altno) +{ + if (device_setup[chip->index] & AUDIOPHILE_SET) { + if ((device_setup[chip->index] & AUDIOPHILE_SET_DTS) + && altno != 6) + return 1; /* skip this altsetting */ + if ((device_setup[chip->index] & AUDIOPHILE_SET_96K) + && altno != 1) + return 1; /* skip this altsetting */ + if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_24B_48K_DI && altno != 2) + return 1; /* skip this altsetting */ + if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_24B_48K_NOTDI && altno != 3) + return 1; /* skip this altsetting */ + if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_16B_48K_DI && altno != 4) + return 1; /* skip this altsetting */ + if ((device_setup[chip->index] & AUDIOPHILE_SET_MASK) == + AUDIOPHILE_SET_16B_48K_NOTDI && altno != 5) + return 1; /* skip this altsetting */ + } + return 0; /* keep this altsetting */ +} /* * audio-interface quirks -- cgit 1.2.3-korg From 4ccb4a4387b3ed8c5a03862ef1e6f7be484ade25 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 15 Mar 2006 12:24:19 +0100 Subject: [ALSA] usb-audio: add MDP-5/EZ-J24 support Modules: USB generic driver Add support for the Yamaha MDP-5 and EZ-J24. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 248a11aa8cdfc7..f231de5a586f44 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -82,6 +82,7 @@ YAMAHA_DEVICE(0x1012, "TYROS"), YAMAHA_DEVICE(0x1013, "PF-500"), YAMAHA_DEVICE(0x1014, "S90"), YAMAHA_DEVICE(0x1015, "MOTIF-R"), +YAMAHA_DEVICE(0x1016, "MDP-5"), YAMAHA_DEVICE(0x1017, "CVP-204"), YAMAHA_DEVICE(0x1018, "CVP-206"), YAMAHA_DEVICE(0x1019, "CVP-208"), @@ -90,6 +91,7 @@ YAMAHA_DEVICE(0x101b, "PSR-1100"), YAMAHA_DEVICE(0x101c, "PSR-2100"), YAMAHA_DEVICE(0x101d, "CLP-175"), YAMAHA_DEVICE(0x101e, "PSR-K1"), +YAMAHA_DEVICE(0x101f, "EZ-J24"), YAMAHA_DEVICE(0x1020, "EZ-250i"), YAMAHA_DEVICE(0x1021, "MOTIF ES 6"), YAMAHA_DEVICE(0x1022, "MOTIF ES 7"), -- cgit 1.2.3-korg From 7c5706bb33687ce82f30d9ac06dd1bdf71b2262e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Mar 2006 13:52:54 +0100 Subject: [ALSA] ac97 - Allow drivers to set static volume resolution table Modules: AC97 Codec Add the pointer to a static volume resolution table to ac97 template, so that the drivers can define the volume resolution, too. Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 13 +++++++------ sound/pci/ac97/ac97_codec.c | 1 + 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 9036d25e1baca3..dee766c17e9cdc 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -433,6 +433,12 @@ struct snd_ac97_bus { struct snd_info_entry *proc; }; +/* static resolution table */ +struct snd_ac97_res_table { + unsigned short reg; /* register */ + unsigned short bits; /* resolution bitmask */ +}; + struct snd_ac97_template { void *private_data; void (*private_free) (struct snd_ac97 *ac97); @@ -442,12 +448,7 @@ struct snd_ac97_template { unsigned int scaps; /* driver capabilities */ unsigned int limited_regs; /* allow limited registers only */ DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */ -}; - -/* static resolution table */ -struct snd_ac97_res_table { - unsigned short reg; /* register */ - unsigned short bits; /* resolution bitmask */ + const struct snd_ac97_res_table *res_table; /* static resolution */ }; struct snd_ac97 { diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 38b6c65d40c37d..c5bbdcbf542275 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -1868,6 +1868,7 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, ac97->scaps = template->scaps; ac97->limited_regs = template->limited_regs; memcpy(ac97->reg_accessed, template->reg_accessed, sizeof(ac97->reg_accessed)); + ac97->res_table = template->res_table; bus->codec[ac97->num] = ac97; mutex_init(&ac97->reg_mutex); mutex_init(&ac97->page_mutex); -- cgit 1.2.3-korg From a23446c085faed6c1c90fba5cdd21d6990871750 Mon Sep 17 00:00:00 2001 From: Florian Schlichting Date: Wed, 15 Mar 2006 14:05:19 +0100 Subject: [ALSA] Fix NM256 hard lock up Modules: NM256 driver Treat the nm256 mixer as a write-only device so as to avoid hangs on initialisation. Signed-off-by: Florian Schlichting Signed-off-by: Takashi Iwai --- sound/pci/nm256/nm256.c | 120 ++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 96 insertions(+), 24 deletions(-) diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 3a3ba6f547bc3b..cc297abc9d1137 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -228,6 +228,7 @@ struct nm256 { unsigned int use_cache: 1; /* use one big coef. table */ unsigned int reset_workaround: 1; /* Workaround for some laptops to avoid freeze */ unsigned int reset_workaround_2: 1; /* Extended workaround for some other laptops to avoid freeze */ + unsigned int in_resume: 1; int mixer_base; /* register offset of ac97 mixer */ int mixer_status_offset; /* offset of mixer status reg. */ @@ -242,6 +243,7 @@ struct nm256 { struct nm256_stream streams[2]; struct snd_ac97 *ac97; + unsigned short *ac97_regs; /* register caches, only for valid regs */ struct snd_pcm *pcm; @@ -1153,23 +1155,63 @@ snd_nm256_ac97_ready(struct nm256 *chip) return 0; } +/* + * Initial register values to be written to the AC97 mixer. + * While most of these are identical to the reset values, we do this + * so that we have most of the register contents cached--this avoids + * reading from the mixer directly (which seems to be problematic, + * probably due to ignorance). + */ + +struct initialValues { + unsigned short reg; + unsigned short value; +}; + +static struct initialValues nm256_ac97_init_val[] = +{ + { AC97_MASTER, 0x8000 }, + { AC97_HEADPHONE, 0x8000 }, + { AC97_MASTER_MONO, 0x8000 }, + { AC97_PC_BEEP, 0x8000 }, + { AC97_PHONE, 0x8008 }, + { AC97_MIC, 0x8000 }, + { AC97_LINE, 0x8808 }, + { AC97_CD, 0x8808 }, + { AC97_VIDEO, 0x8808 }, + { AC97_AUX, 0x8808 }, + { AC97_PCM, 0x8808 }, + { AC97_REC_SEL, 0x0000 }, + { AC97_REC_GAIN, 0x0B0B }, + { AC97_GENERAL_PURPOSE, 0x0000 }, + { AC97_3D_CONTROL, 0x8000 }, + { AC97_VENDOR_ID1, 0x8384 }, + { AC97_VENDOR_ID2, 0x7609 }, +}; + +static int nm256_ac97_idx(unsigned short reg) +{ + int i; + for (i = 0; i < ARRAY_SIZE(nm256_ac97_init_val); i++) + if (nm256_ac97_init_val[i].reg == reg) + return i; + return -1; +} + /* + * some nm256 easily crash when reading from mixer registers + * thus we're treating it as a write-only mixer and cache the + * written values */ static unsigned short snd_nm256_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct nm256 *chip = ac97->private_data; - int res; - - if (reg >= 128) - return 0; + int idx = nm256_ac97_idx(reg); - if (! snd_nm256_ac97_ready(chip)) + if (idx < 0) return 0; - res = snd_nm256_readw(chip, chip->mixer_base + reg); - /* Magic delay. Bleah yucky. */ - msleep(1); - return res; + return chip->ac97_regs[idx]; } /* @@ -1180,8 +1222,12 @@ snd_nm256_ac97_write(struct snd_ac97 *ac97, { struct nm256 *chip = ac97->private_data; int tries = 2; + int idx = nm256_ac97_idx(reg); u32 base; + if (idx < 0) + return; + base = chip->mixer_base; snd_nm256_ac97_ready(chip); @@ -1190,12 +1236,32 @@ snd_nm256_ac97_write(struct snd_ac97 *ac97, while (tries-- > 0) { snd_nm256_writew(chip, base + reg, val); msleep(1); /* a little delay here seems better.. */ - if (snd_nm256_ac97_ready(chip)) + if (snd_nm256_ac97_ready(chip)) { + /* successful write: set cache */ + chip->ac97_regs[idx] = val; return; + } } snd_printd("nm256: ac97 codec not ready..\n"); } +/* static resolution table */ +static struct snd_ac97_res_table nm256_res_table[] = { + { AC97_MASTER, 0x1f1f }, + { AC97_HEADPHONE, 0x1f1f }, + { AC97_MASTER_MONO, 0x001f }, + { AC97_PC_BEEP, 0x001f }, + { AC97_PHONE, 0x001f }, + { AC97_MIC, 0x001f }, + { AC97_LINE, 0x1f1f }, + { AC97_CD, 0x1f1f }, + { AC97_VIDEO, 0x1f1f }, + { AC97_AUX, 0x1f1f }, + { AC97_PCM, 0x1f1f }, + { AC97_REC_GAIN, 0x0f0f }, + { } /* terminator */ +}; + /* initialize the ac97 into a known state */ static void snd_nm256_ac97_reset(struct snd_ac97 *ac97) @@ -1213,6 +1279,16 @@ snd_nm256_ac97_reset(struct snd_ac97 *ac97) snd_nm256_writeb(chip, 0x6cc, 0x80); snd_nm256_writeb(chip, 0x6cc, 0x0); } + if (! chip->in_resume) { + int i; + for (i = 0; i < ARRAY_SIZE(nm256_ac97_init_val); i++) { + /* preload the cache, so as to avoid even a single + * read of the mixer regs + */ + snd_nm256_ac97_write(ac97, nm256_ac97_init_val[i].reg, + nm256_ac97_init_val[i].value); + } + } } /* create an ac97 mixer interface */ @@ -1221,32 +1297,25 @@ snd_nm256_mixer(struct nm256 *chip) { struct snd_ac97_bus *pbus; struct snd_ac97_template ac97; - int i, err; + int err; static struct snd_ac97_bus_ops ops = { .reset = snd_nm256_ac97_reset, .write = snd_nm256_ac97_write, .read = snd_nm256_ac97_read, }; - /* looks like nm256 hangs up when unexpected registers are touched... */ - static int mixer_regs[] = { - AC97_MASTER, AC97_HEADPHONE, AC97_MASTER_MONO, - AC97_PC_BEEP, AC97_PHONE, AC97_MIC, AC97_LINE, AC97_CD, - AC97_VIDEO, AC97_AUX, AC97_PCM, AC97_REC_SEL, - AC97_REC_GAIN, AC97_GENERAL_PURPOSE, AC97_3D_CONTROL, - /*AC97_EXTENDED_ID,*/ - AC97_VENDOR_ID1, AC97_VENDOR_ID2, - -1 - }; + + chip->ac97_regs = kcalloc(sizeof(short), + ARRAY_SIZE(nm256_ac97_init_val), GFP_KERNEL); + if (! chip->ac97_regs) + return -ENOMEM; if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) return err; memset(&ac97, 0, sizeof(ac97)); ac97.scaps = AC97_SCAP_AUDIO; /* we support audio! */ - ac97.limited_regs = 1; - for (i = 0; mixer_regs[i] >= 0; i++) - set_bit(mixer_regs[i], ac97.reg_accessed); ac97.private_data = chip; + ac97.res_table = nm256_res_table; pbus->no_vra = 1; err = snd_ac97_mixer(pbus, &ac97, &chip->ac97); if (err < 0) @@ -1331,6 +1400,7 @@ static int nm256_resume(struct pci_dev *pci) int i; /* Perform a full reset on the hardware */ + chip->in_resume = 1; pci_restore_state(pci); pci_enable_device(pci); snd_nm256_init_chip(chip); @@ -1348,6 +1418,7 @@ static int nm256_resume(struct pci_dev *pci) } snd_power_change_state(card, SNDRV_CTL_POWER_D0); + chip->in_resume = 0; return 0; } #endif /* CONFIG_PM */ @@ -1372,6 +1443,7 @@ static int snd_nm256_free(struct nm256 *chip) free_irq(chip->irq, chip); pci_disable_device(chip->pci); + kfree(chip->ac97_regs); kfree(chip); return 0; } -- cgit 1.2.3-korg From 0a589d80191471754e6f6cab3015a879bca2f7d2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Mar 2006 14:08:20 +0100 Subject: [ALSA] ac97 - Clean up obsolete workarounds Modules: AC97 Codec Clean up obsolete workarounds provided only for nm256. Signed-off-by: Takashi Iwai --- include/sound/ac97_codec.h | 3 --- sound/pci/ac97/ac97_codec.c | 8 -------- 2 files changed, 11 deletions(-) diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index dee766c17e9cdc..b45a7371274806 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -446,8 +446,6 @@ struct snd_ac97_template { unsigned short num; /* number of codec: 0 = primary, 1 = secondary */ unsigned short addr; /* physical address of codec [0-3] */ unsigned int scaps; /* driver capabilities */ - unsigned int limited_regs; /* allow limited registers only */ - DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */ const struct snd_ac97_res_table *res_table; /* static resolution */ }; @@ -477,7 +475,6 @@ struct snd_ac97 { unsigned int rates[6]; /* see AC97_RATES_* defines */ unsigned int spdif_status; unsigned short regs[0x80]; /* register cache */ - unsigned int limited_regs; /* allow limited registers only */ DECLARE_BITMAP(reg_accessed, 0x80); /* bit flags */ union { /* vendor specific code */ struct { diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index c5bbdcbf542275..278319bbdea19b 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -192,9 +192,6 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) { - if (ac97->limited_regs && ! test_bit(reg, ac97->reg_accessed)) - return 0; - /* filter some registers for buggy codecs */ switch (ac97->id) { case AC97_ID_AK4540: @@ -1008,9 +1005,6 @@ static int snd_ac97_try_volume_mix(struct snd_ac97 * ac97, int reg) break; } - if (ac97->limited_regs && test_bit(reg, ac97->reg_accessed)) - return 1; /* allow without check */ - val = snd_ac97_read(ac97, reg); if (!(val & mask)) { /* nothing seems to be here - mute flag is not set */ @@ -1866,8 +1860,6 @@ int snd_ac97_mixer(struct snd_ac97_bus *bus, struct snd_ac97_template *template, ac97->num = template->num; ac97->addr = template->addr; ac97->scaps = template->scaps; - ac97->limited_regs = template->limited_regs; - memcpy(ac97->reg_accessed, template->reg_accessed, sizeof(ac97->reg_accessed)); ac97->res_table = template->res_table; bus->codec[ac97->num] = ac97; mutex_init(&ac97->reg_mutex); -- cgit 1.2.3-korg From a12606cff106335650f0e9382e87b0fb08733eff Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Mar 2006 18:24:43 +0100 Subject: [ALSA] hda-codec - Add model entry for FIC P4M-915GD1 Modules: HDA Codec driver Add model entry for FIC P4M-915GD1 with ALC880 codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0697c9bd45cc3f..1607a1fab5682e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2051,6 +2051,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1025, .pci_subdevice = 0x0078, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x0087, .config = ALC880_6ST_DIG }, { .pci_subvendor = 0x1297, .pci_subdevice = 0xc790, .config = ALC880_6ST_DIG }, /* Shuttle ST20G5 */ + { .pci_subvendor = 0x1509, .pci_subdevice = 0x925d, .config = ALC880_6ST_DIG }, /* FIC P4M-915GD1 */ { .modelname = "asus", .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1964, .config = ALC880_ASUS_DIG }, -- cgit 1.2.3-korg From 9808dc962f6780ce7eac47c82400fede7f4a08b3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 16 Mar 2006 08:15:25 +0100 Subject: [ALSA] usb-audio: add Casio PL-40R support Modules: USB generic driver Add a quirk entry for the Casio PL-40R. Signed-off-by: Clemens Ladisch --- sound/usb/usbquirks.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index f231de5a586f44..0992a0923f1aa8 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1409,6 +1409,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Casio devices */ +{ + USB_DEVICE(0x07cf, 0x6801), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Casio", + .product_name = "PL-40R", + .ifnum = 0, + .type = QUIRK_MIDI_YAMAHA + } +}, { /* this ID is used by several devices without a product ID */ USB_DEVICE(0x07cf, 0x6802), -- cgit 1.2.3-korg From c7a93b852bf9ffcf466b304fa3cfac8823f9b932 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 16 Mar 2006 08:25:56 +0100 Subject: [ALSA] usb-audio: add Maya44 mixer control names Modules: USB generic driver Add mixer control names for the AudioTrak Maya44 USB. Signed-off-by: Clemens Ladisch --- sound/usb/usbmixer_maps.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index c1264434e50ac0..37accb68652d07 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -195,6 +195,22 @@ static struct usbmix_name_map linex_map[] = { { 0 } /* terminator */ }; +static struct usbmix_name_map maya44_map[] = { + /* 1: IT line */ + { 2, "Line Playback" }, /* FU */ + /* 3: IT line */ + { 4, "Line Playback" }, /* FU */ + /* 5: IT pcm playback */ + /* 6: MU */ + { 7, "Master Playback" }, /* FU */ + /* 8: OT speaker */ + /* 9: IT line */ + { 10, "Line Capture" }, /* FU */ + /* 11: MU */ + /* 12: OT pcm capture */ + { } +}; + /* Section "justlink_map" below added by James Courtier-Dutton * sourced from Maplin Electronics (http://www.maplin.co.uk), part number A56AK * Part has 2 connectors that act as a single output. (TOSLINK Optical for digital out, and 3.5mm Jack for Analogue out.) @@ -252,6 +268,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = linex_map, .ignore_ctl_error = 1, }, + { + .id = USB_ID(0x0a92, 0x0091), + .map = maya44_map, + }, { .id = USB_ID(0x0c45, 0x1158), .map = justlink_map, -- cgit 1.2.3-korg From db064e503419c32df463326a3891a973bb30582e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 16 Mar 2006 16:04:58 +0100 Subject: [ALSA] hda-codec - Add support for VAIO FE550G and SZ110 Modules: Documentation,HDA Codec driver Add support for VAIO FE550G and SZ110 laptops with Sigmatel codec (7661). The new model 'vaio' is added. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 + sound/pci/hda/patch_sigmatel.c | 161 ++++++++++++++++++++++++ 2 files changed, 164 insertions(+) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index f947c4b04ab8b6..dcc9cd9633ddd8 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -755,6 +755,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. laptop-dig ditto with SPDIF auto auto-confgi reading BIOS (default) + STAC7661(?) + vaio Setup for VAIO FE550G/SZ110 + If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 45ddf548d6fb40..240958df26ceaa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1147,6 +1147,166 @@ static int patch_stac927x(struct hda_codec *codec) return 0; } +/* + * STAC 7661(?) hack + */ + +/* static config for Sony VAIO FE550G */ +static hda_nid_t vaio_dacs[] = { 0x2 }; +#define VAIO_HP_DAC 0x5 +static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; +static hda_nid_t vaio_mux_nids[] = { 0x15 }; + +static struct hda_input_mux vaio_mux = { + .num_items = 2, + .items = { + /* { "HP", 0x0 }, + { "Unknown", 0x1 }, */ + { "Mic", 0x2 }, + { "PCM", 0x3 }, + } +}; + +static struct hda_verb vaio_init[] = { + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ + {} +}; + +/* bind volumes of both NID 0x02 and 0x05 */ +static int vaio_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +/* bind volumes of both NID 0x02 and 0x05 */ +static int vaio_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x02, 0, HDA_OUTPUT, 0, + 0x80, valp[0] & 0x80); + change |= snd_hda_codec_amp_update(codec, 0x02, 1, HDA_OUTPUT, 0, + 0x80, valp[1] & 0x80); + snd_hda_codec_amp_update(codec, 0x05, 0, HDA_OUTPUT, 0, + 0x80, valp[0] & 0x80); + snd_hda_codec_amp_update(codec, 0x05, 1, HDA_OUTPUT, 0, + 0x80, valp[1] & 0x80); + return change; +} + +static struct snd_kcontrol_new vaio_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = vaio_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = vaio_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + }, + /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .count = 1, + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, + }, + {} +}; + +static struct hda_codec_ops stac7661_patch_ops = { + .build_controls = stac92xx_build_controls, + .build_pcms = stac92xx_build_pcms, + .init = stac92xx_init, + .free = stac92xx_free, +#ifdef CONFIG_PM + .resume = stac92xx_resume, +#endif +}; + +enum { STAC7661_VAIO }; + +static struct hda_board_config stac7661_cfg_tbl[] = { + { .modelname = "vaio", .config = STAC7661_VAIO }, + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81e6, + .config = STAC7661_VAIO }, + { .pci_subvendor = 0x104d, .pci_subdevice = 0x81ef, + .config = STAC7661_VAIO }, + {} +}; + +static int patch_stac7661(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + int board_config; + + board_config = snd_hda_check_board_config(codec, stac7661_cfg_tbl); + if (board_config < 0) + /* unknown config, let generic-parser do its job... */ + return snd_hda_parse_generic_codec(codec); + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + switch (board_config) { + case STAC7661_VAIO: + spec->mixer = vaio_mixer; + spec->init = vaio_init; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); + spec->multiout.dac_nids = vaio_dacs; + spec->multiout.hp_nid = VAIO_HP_DAC; + spec->num_adcs = ARRAY_SIZE(vaio_adcs); + spec->adc_nids = vaio_adcs; + spec->input_mux = &vaio_mux; + spec->mux_nids = vaio_mux_nids; + break; + } + + codec->patch_ops = stac7661_patch_ops; + return 0; +} + + /* * patch entries */ @@ -1168,5 +1328,6 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x }, { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x }, { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x }, + { .id = 0x83847661, .name = "STAC7661", .patch = patch_stac7661 }, {} /* terminator */ }; -- cgit 1.2.3-korg From 825aa97241b46d2819c1db984c86a1a9df41b8e1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Mar 2006 10:50:49 +0100 Subject: [ALSA] hda-codec - Fix for Samsung R65 and ASUS A6J Modules: Documentation,HDA Codec driver Added a new model 'laptop-eapd' to AD1986A codec for Samsung R65 and ASUS A6J laptops. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 1 + sound/pci/hda/patch_analog.c | 116 +++++++++++++++++++++++- 2 files changed, 116 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index dcc9cd9633ddd8..1def6049784c95 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -745,6 +745,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 6stack 6-jack, separate surrounds (default) 3stack 3-stack, shared surrounds laptop 2-channel only (FSC V2060, Samsung M50) + laptop-eapd 2-channel with EAPD (Samsung R65, ASUS A6J) AD1988 6stack 6-jack diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cdcc815a9320f0..2b14fa74a8fd01 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -581,6 +581,97 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { { } /* end */ }; +/* laptop-eapd model - 2ch only */ + +/* master controls both pins 0x1a and 0x1b */ +static int ad1986a_laptop_master_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, + 0x7f, valp[0] & 0x7f); + snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, + 0x7f, valp[1] & 0x7f); + return change; +} + +static int ad1986a_laptop_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + change |= snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, + 0x80, valp[0] ? 0 : 0x80); + snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, + 0x80, valp[1] ? 0 : 0x80); + return change; +} + +static struct hda_input_mux ad1986a_laptop_eapd_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x4 }, + { "Mix", 0x5 }, + }, +}; + +static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = ad1986a_laptop_master_vol_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = ad1986a_laptop_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "External Amplifier", + .info = ad198x_eapd_info, + .get = ad198x_eapd_get, + .put = ad198x_eapd_put, + .private_value = 0x1b | (1 << 8), /* port-D, inversed */ + }, + { } /* end */ +}; + /* * initialization verbs */ @@ -683,8 +774,14 @@ static struct hda_channel_mode ad1986a_modes[3] = { { 6, ad1986a_ch6_init }, }; +/* eapd initialization */ +static struct hda_verb ad1986a_eapd_init_verbs[] = { + {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, + {} +}; + /* models */ -enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP }; +enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD }; static struct hda_board_config ad1986a_cfg_tbl[] = { { .modelname = "6stack", .config = AD1986A_6STACK }, @@ -694,6 +791,13 @@ static struct hda_board_config ad1986a_cfg_tbl[] = { .config = AD1986A_LAPTOP }, /* FSC V2060 */ { .pci_subvendor = 0x17c0, .pci_subdevice = 0x2017, .config = AD1986A_LAPTOP }, /* Samsung M50 */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x818f, + .config = AD1986A_LAPTOP }, /* ASUS P5GV-MX */ + { .modelname = "laptop-eapd", .config = AD1986A_LAPTOP_EAPD }, + { .pci_subvendor = 0x144d, .pci_subdevice = 0xc024, + .config = AD1986A_LAPTOP_EAPD }, /* Samsung R65-T2300 Charis */ + { .pci_subvendor = 0x1043, .pci_subdevice = 0x1213, + .config = AD1986A_LAPTOP_EAPD }, /* ASUS A6J */ {} }; @@ -741,6 +845,16 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.num_dacs = 1; spec->multiout.dac_nids = ad1986a_laptop_dac_nids; break; + case AD1986A_LAPTOP_EAPD: + spec->mixers[0] = ad1986a_laptop_eapd_mixers; + spec->num_init_verbs = 2; + spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1986a_laptop_eapd_capture_source; + break; } return 0; -- cgit 1.2.3-korg From e64d2e36236175a56fd1d8b2acd784e21bdc7de6 Mon Sep 17 00:00:00 2001 From: Eugene Teo Date: Fri, 17 Mar 2006 16:32:17 +0100 Subject: [ALSA] Fix seq_clientmgr dereferences before NULL check Modules: ALSA sequencer cptr->pool must be non-NULL there, so just the if (cptr->pool) is superfluous. Thanks Takashi. Signed-off-by: Eugene Teo Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index aae6420f594870..bb15d9ee8842a0 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1866,8 +1866,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, info.output_pool = cptr->pool->size; info.output_room = cptr->pool->room; info.output_free = info.output_pool; - if (cptr->pool) - info.output_free = snd_seq_unused_cells(cptr->pool); + info.output_free = snd_seq_unused_cells(cptr->pool); if (cptr->type == USER_CLIENT) { info.input_pool = cptr->data.user.fifo_pool_size; info.input_free = info.input_pool; -- cgit 1.2.3-korg From d61975fc6b40dadd2cd61fc9535499c7d7b806de Mon Sep 17 00:00:00 2001 From: Eugene Teo Date: Fri, 17 Mar 2006 16:32:52 +0100 Subject: [ALSA] Fix gus_pcm dereference before NULL Modules: GUS Library The NULL check of substream is simply superfluous. It is guaranteed to receive non-NULL substream. Thanks Takashi. Coverity bug #861 Signed-off-by: Eugene Teo Signed-off-by: Takashi Iwai --- sound/isa/gus/gus_pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index d0829393ec8a21..c7f95e7aa01851 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -114,8 +114,6 @@ static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream) unsigned char pan; unsigned int voice; - if (substream == NULL) - return; spin_lock_irqsave(&pcmp->lock, flags); if (pcmp->flags & SNDRV_GF1_PCM_PFLG_ACTIVE) { spin_unlock_irqrestore(&pcmp->lock, flags); -- cgit 1.2.3-korg From 878b4789196e3cd470f843854b6b09d963214659 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 20 Mar 2006 11:27:13 +0100 Subject: [ALSA] fix resource leak in usbmixer Modules: USB generic driver We may leak 'namelist' in sound/usb/usbmixer.c::parse_audio_selector_unit() Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 0bfea72373957e..8d08b34a1cb569 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1468,6 +1468,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi kctl = snd_ctl_new1(&mixer_selectunit_ctl, cval); if (! kctl) { snd_printk(KERN_ERR "cannot malloc kcontrol\n"); + kfree(namelist); kfree(cval); return -ENOMEM; } -- cgit 1.2.3-korg From 6682025e1220a1ab0b6b855b7805928c3163e043 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 20 Mar 2006 18:31:57 +0100 Subject: [ALSA] ice1712 - Fix wrong value types for enum items Modules: ICE1712 driver Fix the access to wrong type values for enum items in aureon.c (ALSA bug#1527). Signed-off-by: Takashi Iwai --- sound/pci/ice1712/aureon.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 0f7f4d8263c0eb..7e6608b14abce3 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -955,8 +955,8 @@ static int wm_adc_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val mutex_lock(&ice->gpio_mutex); val = wm_get(ice, WM_ADC_MUX); - ucontrol->value.integer.value[0] = val & 7; - ucontrol->value.integer.value[1] = (val >> 4) & 7; + ucontrol->value.enumerated.item[0] = val & 7; + ucontrol->value.enumerated.item[1] = (val >> 4) & 7; mutex_unlock(&ice->gpio_mutex); return 0; } @@ -970,8 +970,8 @@ static int wm_adc_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_val snd_ice1712_save_gpio_status(ice); oval = wm_get(ice, WM_ADC_MUX); nval = oval & ~0x77; - nval |= ucontrol->value.integer.value[0] & 7; - nval |= (ucontrol->value.integer.value[1] & 7) << 4; + nval |= ucontrol->value.enumerated.item[0] & 7; + nval |= (ucontrol->value.enumerated.item[1] & 7) << 4; change = (oval != nval); if (change) wm_put(ice, WM_ADC_MUX, nval); @@ -1011,7 +1011,7 @@ static int aureon_cs8415_mux_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e //snd_ice1712_save_gpio_status(ice); //val = aureon_cs8415_get(ice, CS8415_CTRL2); - ucontrol->value.integer.value[0] = ice->spec.aureon.cs8415_mux; + ucontrol->value.enumerated.item[0] = ice->spec.aureon.cs8415_mux; //snd_ice1712_restore_gpio_status(ice); return 0; } @@ -1025,12 +1025,12 @@ static int aureon_cs8415_mux_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e snd_ice1712_save_gpio_status(ice); oval = aureon_cs8415_get(ice, CS8415_CTRL2); nval = oval & ~0x07; - nval |= ucontrol->value.integer.value[0] & 7; + nval |= ucontrol->value.enumerated.item[0] & 7; change = (oval != nval); if (change) aureon_cs8415_put(ice, CS8415_CTRL2, nval); snd_ice1712_restore_gpio_status(ice); - ice->spec.aureon.cs8415_mux = ucontrol->value.integer.value[0]; + ice->spec.aureon.cs8415_mux = ucontrol->value.enumerated.item[0]; return change; } -- cgit 1.2.3-korg From d8327c784b51b57dac2c26cfad87dce0d68dfd98 Mon Sep 17 00:00:00 2001 From: Sergei Shtylylov Date: Mon, 20 Mar 2006 18:38:21 +0100 Subject: [ALSA] AMD Au1x00: make driver build after cleanup Modules: MIPS AU1x00 driver AMD Au1x00 ALSA driver doesn't build after the recent code cleanup: sound/mips/au1x00.c: In function 'au1000_setup_dma_link': sound/mips/au1x00.c:173: error: 'pointer' undeclared (first use in this function) sound/mips/au1x00.c:173: error: (Each undeclared identifier is reported only once sound/mips/au1x00.c:173: error: for each function it appears in.) sound/mips/au1x00.c: In function 'snd_au1000_hw_params': sound/mips/au1x00.c:339: warning: implicit declaration of function 'snd_mask_min' Signed-off-by: Takashi Iwai --- sound/mips/au1x00.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 6d8f8b3eabd0ef..e66b0144e3b561 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -43,6 +43,7 @@ #include #include #include +#include #include #include #include @@ -153,6 +154,7 @@ au1000_setup_dma_link(struct audio_stream *stream, unsigned int period_bytes, { struct snd_pcm_substream *substream = stream->substream; struct snd_pcm_runtime *runtime = substream->runtime; + struct au1000_period *pointer; unsigned long dma_start; int i; -- cgit 1.2.3-korg From 01a00e5e421b3c40240781d67bb9fd5fd9b1f0f1 Mon Sep 17 00:00:00 2001 From: Alan Horstmann Date: Tue, 21 Mar 2006 09:57:36 +0100 Subject: [ALSA] ice1712 - typo fixes for dxr_enable module option Modules: ICE1712 driver Signed-off-by: Jaroslav Kysela --- sound/pci/ice1712/ice1712.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 78098319209cf5..b96b5d6efc5d23 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -83,11 +83,11 @@ MODULE_SUPPORTED_DEVICE("{" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ -static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;/* Enable this card */ static char *model[SNDRV_CARDS]; -static int omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ +static int omni[SNDRV_CARDS]; /* Delta44 & 66 Omni I/O support */ static int cs8427_timeout[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* CS8427 S/PDIF transciever reset timeout value in msec */ -static int dxr_enable[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = 500}; /* DXR enable for DMX6FIRE */ +static int dxr_enable[SNDRV_CARDS]; /* DXR enable for DMX6FIRE */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for ICE1712 soundcard."); @@ -102,7 +102,7 @@ MODULE_PARM_DESC(cs8427_timeout, "Define reset timeout for cs8427 chip in msec r module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); module_param_array(dxr_enable, int, NULL, 0444); -MODULE_PARM_DESC(dsr_enable, "Enable DXR support for Terratec DMX6FIRE."); +MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); static struct pci_device_id snd_ice1712_ids[] = { -- cgit 1.2.3-korg From 19739fef0203d2f3eecc9c4b1ef25b57d85f2b30 Mon Sep 17 00:00:00 2001 From: Thibault LE MEUR Date: Tue, 21 Mar 2006 11:06:40 +0100 Subject: [ALSA] Fixes typos in Audiophile-USB.txt Modules: Documentation Fixes typos in Audiophile-USB.txt. Signed-off-by: Thibault LE MEUR Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/Audiophile-Usb.txt | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/Documentation/sound/alsa/Audiophile-Usb.txt b/Documentation/sound/alsa/Audiophile-Usb.txt index 3ba45adbf04052..4692c8e77dc17d 100644 --- a/Documentation/sound/alsa/Audiophile-Usb.txt +++ b/Documentation/sound/alsa/Audiophile-Usb.txt @@ -1,4 +1,4 @@ - Guide to using M-Audio Audiophile USB with ALSA and Jack v1.1 + Guide to using M-Audio Audiophile USB with ALSA and Jack v1.2 ======================================================== Thibault Le Meur @@ -13,6 +13,9 @@ of the device. The device has 4 audio interfaces, and 2 MIDI ports: * Analog Stereo Input (Ai) + - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) + - When the 1/4" TS (jack) connectors are connected, the RCA connectors + are disabled * Analog Stereo Output (Ao) * Digital Stereo Input (Di) * Digital Stereo Output (Do) @@ -42,7 +45,7 @@ Important facts about the Digital interface: though I haven't tested it under linux - Note that in this setup only the Do interface can be enabled * Apart from recording an audio digital stream, enabling the Di port is a way -to syncrhonize the device to an external sample clock +to synchronize the device to an external sample clock - As a consequence, the Di port must be enable only if an active Digital source is connected - Enabling Di when no digital source is connected can result in a @@ -180,7 +183,7 @@ IMPORTANT NOTE WHEN SWITCHING CONFIGURATION: or changing modprobe.conf) - turn on the device -2.2.2.3 - Setting and switching configurations with the device_setup parameter +2.2.2.3 - Audiophile USB's device_setup structure If you want to understand the device_setup magic numbers for the Audiophile USB, you need some very basic understanding of binary computation. However, @@ -226,7 +229,7 @@ Caution: - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll only be able to use one at the same time -2.2.3 - Technical Details for Audiophile Usb +2.2.3 - USB implementation details for this device You may safely skip this section if you're not interrested in driver development. @@ -234,20 +237,20 @@ development. This section describes some internals aspect of the device and summarize the data I got by usb-snooping the windows and linux drivers. -The M-Audio Audiophile USB has 7 Usb Interfaces: +The M-Audio Audiophile USB has 7 USB Interfaces: a "USB interface": - * Usb Interface nb.0 - * Usb Interface nb.1 + * USB Interface nb.0 + * USB Interface nb.1 - Audio Control function - * Usb Interface nb.2 + * USB Interface nb.2 - Analog Output - * Usb Interface nb.3 + * USB Interface nb.3 - Digital Output - * Usb Interface nb.4 + * USB Interface nb.4 - Analog Input - * Usb Interface nb.5 + * USB Interface nb.5 - Digital Input - * Usb Interface nb.6 + * USB Interface nb.6 - MIDI interface compliant with the MIDIMAN quirk Each interface has 5 altsettings (AltSet 1,2,3,4,5) except: @@ -326,5 +329,5 @@ atid=425939 After having applied the patch you can run jackd with the following command line: -# /usr/local/bin/jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 + % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1 -- cgit 1.2.3-korg From 82bc955f6379135e6ce35ff90c7ac411fd412c4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2006 11:24:42 +0100 Subject: [ALSA] hda-codec - Fix BIOS auto-configuration Modules: HDA Codec driver,HDA generic driver - Fix autoconfig speaker/hp detection Now it allows multiple speaker pins (e.g. Dell laptops have such config) - Use speaker or hp pins if no line-outs are available This fixes the silence output on recent Dell laptops with STAC9200 (ALSA bug#1843) - Fix analog/realtek/sigmatel autoconfig parser Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 77 ++++++++++++++++++++++++++++++++++++-- sound/pci/hda/hda_local.h | 4 +- sound/pci/hda/patch_analog.c | 23 ++++++------ sound/pci/hda/patch_realtek.c | 47 +++++++---------------- sound/pci/hda/patch_sigmatel.c | 84 +++++++++++++++++++++++++++++------------- 5 files changed, 160 insertions(+), 75 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fc91256e42eb6d..b42dff7ceed033 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1899,6 +1899,13 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_o if (mout->hp_nid) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); + /* extra outputs copied from front */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) + if (mout->extra_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->extra_out_nid[i], + stream_tag, 0, format); + /* surrounds */ for (i = 1; i < mout->num_dacs; i++) { if (chs >= (i + 1) * 2) /* independent out */ @@ -1923,6 +1930,11 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_o snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0); if (mout->hp_nid) snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0); + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) + if (mout->extra_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->extra_out_nid[i], + 0, 0, 0); mutex_lock(&codec->spdif_mutex); if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) { snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); @@ -1944,13 +1956,29 @@ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) return 0; } -/* parse all pin widgets and store the useful pin nids to cfg */ +/* + * Parse all pin widgets and store the useful pin nids to cfg + * + * The number of line-outs or any primary output is stored in line_outs, + * and the corresponding output pins are assigned to line_out_pins[], + * in the order of front, rear, CLFE, side, ... + * + * If more extra outputs (speaker and headphone) are found, the pins are + * assisnged to hp_pin and speaker_pins[], respectively. If no line-out jack + * is detected, one of speaker of HP pins is assigned as the primary + * output, i.e. to line_out_pins[0]. So, line_outs is always positive + * if any analog output exists. + * + * The analog input pins are assigned to input_pins array. + * The digital input/output pins are assigned to dig_in_pin and dig_out_pin, + * respectively. + */ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *cfg, hda_nid_t *ignore_nids) { hda_nid_t nid, nid_start; int i, j, nodes; - short seq, sequences[4], assoc_line_out; + short seq, assoc_line_out, sequences[ARRAY_SIZE(cfg->line_out_pins)]; memset(cfg, 0, sizeof(*cfg)); @@ -1992,7 +2020,10 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c cfg->line_outs++; break; case AC_JACK_SPEAKER: - cfg->speaker_pin = nid; + if (cfg->speaker_outs >= ARRAY_SIZE(cfg->speaker_pins)) + continue; + cfg->speaker_pins[cfg->speaker_outs] = nid; + cfg->speaker_outs++; break; case AC_JACK_HP_OUT: cfg->hp_pin = nid; @@ -2057,6 +2088,46 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, struct auto_pin_cfg *c break; } + /* + * debug prints of the parsed results + */ + snd_printd("autoconfig: line_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->line_outs, cfg->line_out_pins[0], cfg->line_out_pins[1], + cfg->line_out_pins[2], cfg->line_out_pins[3], + cfg->line_out_pins[4]); + snd_printd(" speaker_outs=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", + cfg->speaker_outs, cfg->speaker_pins[0], + cfg->speaker_pins[1], cfg->speaker_pins[2], + cfg->speaker_pins[3], cfg->speaker_pins[4]); + snd_printd(" hp=0x%x, dig_out=0x%x, din_in=0x%x\n", + cfg->hp_pin, cfg->dig_out_pin, cfg->dig_in_pin); + snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," + " cd=0x%x, aux=0x%x\n", + cfg->input_pins[AUTO_PIN_MIC], + cfg->input_pins[AUTO_PIN_FRONT_MIC], + cfg->input_pins[AUTO_PIN_LINE], + cfg->input_pins[AUTO_PIN_FRONT_LINE], + cfg->input_pins[AUTO_PIN_CD], + cfg->input_pins[AUTO_PIN_AUX]); + + /* + * FIX-UP: if no line-outs are detected, try to use speaker or HP pin + * as a primary output + */ + if (! cfg->line_outs) { + if (cfg->speaker_outs) { + cfg->line_outs = cfg->speaker_outs; + memcpy(cfg->line_out_pins, cfg->speaker_pins, + sizeof(cfg->speaker_pins)); + cfg->speaker_outs = 0; + memset(cfg->speaker_pins, 0, sizeof(cfg->speaker_pins)); + } else if (cfg->hp_pin) { + cfg->line_outs = 1; + cfg->line_out_pins[0] = cfg->hp_pin; + cfg->hp_pin = 0; + } + } + return 0; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 548a8b6984871a..14e8aa2806edaf 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -135,6 +135,7 @@ struct hda_multi_out { int num_dacs; /* # of DACs, must be more than 1 */ hda_nid_t *dac_nids; /* DAC list */ hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ + hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ hda_nid_t dig_out_nid; /* digital out audio widget */ int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ @@ -221,7 +222,8 @@ extern const char *auto_pin_cfg_labels[AUTO_PIN_LAST]; struct auto_pin_cfg { int line_outs; hda_nid_t line_out_pins[5]; /* sorted in the order of Front/Surr/CLFE/Side */ - hda_nid_t speaker_pin; + int speaker_outs; + hda_nid_t speaker_pins[5]; hda_nid_t hp_pin; hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2b14fa74a8fd01..7fbe71e69cfced 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -786,6 +786,8 @@ enum { AD1986A_6STACK, AD1986A_3STACK, AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD }; static struct hda_board_config ad1986a_cfg_tbl[] = { { .modelname = "6stack", .config = AD1986A_6STACK }, { .modelname = "3stack", .config = AD1986A_3STACK }, + { .pci_subvendor = 0x10de, .pci_subdevice = 0xcb84, + .config = AD1986A_3STACK }, /* ASUS A8N-VM CSM */ { .modelname = "laptop", .config = AD1986A_LAPTOP }, { .pci_subvendor = 0x144d, .pci_subdevice = 0xc01e, .config = AD1986A_LAPTOP }, /* FSC V2060 */ @@ -2253,14 +2255,11 @@ static int ad1988_auto_create_extra_out(struct hda_codec *codec, hda_nid_t pin, idx = ad1988_pin_idx(pin); nid = ad1988_idx_to_dac(codec, idx); - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = nid; - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } else - /* specify the DAC as the extra output */ + /* specify the DAC as the extra output */ + if (! spec->multiout.hp_nid) spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ sprintf(name, "%s Playback Volume", pfx); if ((err = add_control(spec, AD_CTL_WIDGET_VOL, name, @@ -2379,7 +2378,7 @@ static void ad1988_auto_init_extra_out(struct hda_codec *codec) struct ad198x_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.speaker_pin; + pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ ad1988_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); pin = spec->autocfg.hp_pin; @@ -2428,13 +2427,13 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) return err; if ((err = ad1988_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = ad1988_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pin, + (err = ad1988_auto_create_extra_out(codec, + spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pin, + (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pin, "Headphone")) < 0 || (err = ad1988_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1607a1fab5682e..de145d102d9ffc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2431,14 +2431,11 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = nid; - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } else - /* specify the DAC as the extra output */ + /* specify the DAC as the extra output */ + if (! spec->multiout.hp_nid) spec->multiout.hp_nid = nid; + else + spec->multiout.extra_out_nid[0] = nid; /* control HP volume/switch on the output mixer amp */ nid = alc880_idx_to_mixer(alc880_fixed_pin_idx(pin)); sprintf(name, "%s Playback Volume", pfx); @@ -2451,12 +2448,6 @@ static int alc880_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return err; } else if (alc880_is_multi_pin(pin)) { /* set manual connection */ - if (! spec->multiout.dac_nids[0]) { - /* use this as the primary output */ - spec->multiout.dac_nids[0] = alc880_idx_to_dac(alc880_multi_pin_idx(pin)); - if (! spec->multiout.num_dacs) - spec->multiout.num_dacs = 1; - } /* we have only a switch on HP-out PIN */ sprintf(name, "%s Playback Switch", pfx); if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, @@ -2540,7 +2531,7 @@ static void alc880_auto_init_extra_out(struct hda_codec *codec) struct alc_spec *spec = codec->spec; hda_nid_t pin; - pin = spec->autocfg.speaker_pin; + pin = spec->autocfg.speaker_pins[0]; if (pin) /* connect to front */ alc880_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); pin = spec->autocfg.hp_pin; @@ -2576,15 +2567,15 @@ static int alc880_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc880_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc880_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || (err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + (err = alc880_auto_create_extra_out(spec, + spec->autocfg.speaker_pins[0], "Speaker")) < 0 || - (err = alc880_auto_create_extra_out(spec, spec->autocfg.speaker_pin, + (err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pin, "Headphone")) < 0 || (err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) return err; @@ -3445,7 +3436,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, return err; } - nid = cfg->speaker_pin; + nid = cfg->speaker_pins[0]; if (nid) { err = alc260_add_playback_controls(spec, nid, "Speaker"); if (err < 0) @@ -3518,7 +3509,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); - nid = spec->autocfg.speaker_pin; + nid = spec->autocfg.speaker_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); @@ -4602,7 +4593,7 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct return err; } - nid = cfg->speaker_pin; + nid = cfg->speaker_pins[0]; if (nid) { if (nid == 0x16) { if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", @@ -4612,10 +4603,6 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) return err; } else { - if (! cfg->line_out_pins[0]) - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Speaker Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) - return err; if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Speaker Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; @@ -4632,10 +4619,6 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT))) < 0) return err; } else { - if (! cfg->line_out_pins[0]) - if ((err = add_control(spec, ALC_CTL_WIDGET_VOL, "Headphone Playback Volume", - HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT))) < 0) - return err; if ((err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Headphone Playback Switch", HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT))) < 0) return err; @@ -4729,8 +4712,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc262_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || (err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg)) < 0) @@ -5404,8 +5386,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc861_ignore)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.speaker_pin && - ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid BIOS pin config */ if ((err = alc861_auto_fill_dac_nids(spec, &spec->autocfg)) < 0 || diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 240958df26ceaa..b56ca401939225 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -51,6 +51,7 @@ struct sigmatel_spec { unsigned int line_switch: 1; unsigned int mic_switch: 1; unsigned int alt_switch: 1; + unsigned int hp_detect: 1; /* playback */ struct hda_multi_out multiout; @@ -697,13 +698,7 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, const struct aut AC_VERB_GET_CONNECT_LIST, 0) & 0xff; } - if (cfg->line_outs) - spec->multiout.num_dacs = cfg->line_outs; - else if (cfg->hp_pin) { - spec->multiout.dac_nids[0] = snd_hda_codec_read(codec, cfg->hp_pin, 0, - AC_VERB_GET_CONNECT_LIST, 0) & 0xff; - spec->multiout.num_dacs = 1; - } + spec->multiout.num_dacs = cfg->line_outs; return 0; } @@ -772,11 +767,13 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin return 0; wid_caps = get_wcaps(codec, pin); - if (wid_caps & AC_WCAP_UNSOL_CAP) + if (wid_caps & AC_WCAP_UNSOL_CAP) { /* Enable unsolicited responses on the HP widget */ snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_UNSOLICITED_ENABLE, STAC_UNSOL_ENABLE); + spec->hp_detect = 1; + } nid = snd_hda_codec_read(codec, pin, 0, AC_VERB_GET_CONNECT_LIST, 0) & 0xff; for (i = 0; i < cfg->line_outs; i++) { @@ -810,9 +807,6 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const for (i = 0; i < AUTO_PIN_LAST; i++) { int index = -1; if (cfg->input_pins[i]) { - /* Enable active pin widget as an input */ - stac92xx_auto_set_pinctl(codec, cfg->input_pins[i], AC_PINCTL_IN_EN); - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; for (j=0; jnum_muxes; j++) { @@ -861,10 +855,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if ((err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL)) < 0) return err; - if (! spec->autocfg.line_outs && ! spec->autocfg.hp_pin) + if (! spec->autocfg.line_outs) return 0; /* can't find valid pin config */ - stac92xx_auto_init_multi_out(codec); - stac92xx_auto_init_hp_out(codec); if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) return err; if ((err = stac92xx_auto_fill_dac_nids(codec, &spec->autocfg)) < 0) @@ -879,14 +871,10 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - if (spec->autocfg.dig_out_pin) { + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = dig_out; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN); - } - if (spec->autocfg.dig_in_pin) { + if (spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN); - } if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; @@ -896,6 +884,29 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return 1; } +/* add playback controls for HP output */ +static int stac9200_auto_create_hp_ctls(struct hda_codec *codec, + struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + hda_nid_t pin = cfg->hp_pin; + unsigned int wid_caps; + + if (! pin) + return 0; + + wid_caps = get_wcaps(codec, pin); + if (wid_caps & AC_WCAP_UNSOL_CAP) { + /* Enable unsolicited responses on the HP widget */ + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + STAC_UNSOL_ENABLE); + spec->hp_detect = 1; + } + + return 0; +} + static int stac9200_parse_auto_config(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -907,14 +918,13 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) if ((err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) return err; - if (spec->autocfg.dig_out_pin) { + if ((err = stac9200_auto_create_hp_ctls(codec, &spec->autocfg)) < 0) + return err; + + if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = 0x05; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_out_pin, AC_PINCTL_OUT_EN); - } - if (spec->autocfg.dig_in_pin) { + if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; - stac92xx_auto_set_pinctl(codec, spec->autocfg.dig_in_pin, AC_PINCTL_IN_EN); - } if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; @@ -927,9 +937,31 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; snd_hda_sequence_write(codec, spec->init); + /* set up pins */ + if (spec->hp_detect) { + /* fake event to set up pins */ + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + } else { + stac92xx_auto_init_multi_out(codec); + stac92xx_auto_init_hp_out(codec); + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (cfg->input_pins[i]) + stac92xx_auto_set_pinctl(codec, cfg->input_pins[i], + AC_PINCTL_IN_EN); + } + if (cfg->dig_out_pin) + stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, + AC_PINCTL_OUT_EN); + if (cfg->dig_in_pin) + stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, + AC_PINCTL_IN_EN); + return 0; } -- cgit 1.2.3-korg From 97ec558a88fb03fa23aee0832a88964e76e4a4db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2006 11:29:07 +0100 Subject: [ALSA] hda-codec - Fix generic auto-configurator Modules: HDA generic driver Fixed the generic auto-configurator to check speaker pins in addition. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 128 +++++++++++++++++++++++++++++++------------- 1 file changed, 92 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 39edfcfd3abdf0..85ad164ada59e1 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -47,10 +47,10 @@ struct hda_gnode { /* patch-specific record */ struct hda_gspec { - struct hda_gnode *dac_node; /* DAC node */ - struct hda_gnode *out_pin_node; /* Output pin (Line-Out) node */ - struct hda_gnode *pcm_vol_node; /* Node for PCM volume */ - unsigned int pcm_vol_index; /* connection of PCM volume */ + struct hda_gnode *dac_node[2]; /* DAC node */ + struct hda_gnode *out_pin_node[2]; /* Output pin (Line-Out) node */ + struct hda_gnode *pcm_vol_node[2]; /* Node for PCM volume */ + unsigned int pcm_vol_index[2]; /* connection of PCM volume */ struct hda_gnode *adc_node; /* ADC node */ struct hda_gnode *cap_vol_node; /* Node for capture volume */ @@ -69,8 +69,12 @@ struct hda_gspec { /* * retrieve the default device type from the default config value */ -#define defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT) -#define defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT) +#define defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> \ + AC_DEFCFG_DEVICE_SHIFT) +#define defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> \ + AC_DEFCFG_LOCATION_SHIFT) +#define defcfg_port_conn(node) (((node)->def_cfg & AC_DEFCFG_PORT_CONN) >> \ + AC_DEFCFG_PORT_CONN_SHIFT) /* * destructor @@ -261,7 +265,7 @@ static void clear_check_flags(struct hda_gspec *spec) * returns 0 if not found, 1 if found, or a negative error code. */ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, - struct hda_gnode *node) + struct hda_gnode *node, int dac_idx) { int i, err; struct hda_gnode *child; @@ -276,14 +280,14 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, return 0; } snd_printdd("AUD_OUT found %x\n", node->nid); - if (spec->dac_node) { + if (spec->dac_node[dac_idx]) { /* already DAC node is assigned, just unmute & connect */ - return node == spec->dac_node; + return node == spec->dac_node[dac_idx]; } - spec->dac_node = node; + spec->dac_node[dac_idx] = node; if (node->wid_caps & AC_WCAP_OUT_AMP) { - spec->pcm_vol_node = node; - spec->pcm_vol_index = 0; + spec->pcm_vol_node[dac_idx] = node; + spec->pcm_vol_index[dac_idx] = 0; } return 1; /* found */ } @@ -292,7 +296,7 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, child = hda_get_node(spec, node->conn_list[i]); if (! child) continue; - err = parse_output_path(codec, spec, child); + err = parse_output_path(codec, spec, child, dac_idx); if (err < 0) return err; else if (err > 0) { @@ -303,13 +307,13 @@ static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec, select_input_connection(codec, node, i); unmute_input(codec, node, i); unmute_output(codec, node); - if (! spec->pcm_vol_node) { + if (! spec->pcm_vol_node[dac_idx]) { if (node->wid_caps & AC_WCAP_IN_AMP) { - spec->pcm_vol_node = node; - spec->pcm_vol_index = i; + spec->pcm_vol_node[dac_idx] = node; + spec->pcm_vol_index[dac_idx] = i; } else if (node->wid_caps & AC_WCAP_OUT_AMP) { - spec->pcm_vol_node = node; - spec->pcm_vol_index = 0; + spec->pcm_vol_node[dac_idx] = node; + spec->pcm_vol_index[dac_idx] = 0; } } return 1; @@ -339,6 +343,8 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, /* output capable? */ if (! (node->pin_caps & AC_PINCAP_OUT)) continue; + if (defcfg_port_conn(node) == AC_JACK_PORT_NONE) + continue; /* unconnected */ if (jack_type >= 0) { if (jack_type != defcfg_type(node)) continue; @@ -350,10 +356,15 @@ static struct hda_gnode *parse_output_jack(struct hda_codec *codec, continue; } clear_check_flags(spec); - err = parse_output_path(codec, spec, node); + err = parse_output_path(codec, spec, node, 0); if (err < 0) return NULL; - else if (err > 0) { + if (! err && spec->out_pin_node[0]) { + err = parse_output_path(codec, spec, node, 1); + if (err < 0) + return NULL; + } + if (err > 0) { /* unmute the PIN output */ unmute_output(codec, node); /* set PIN-Out enable */ @@ -381,20 +392,28 @@ static int parse_output(struct hda_codec *codec) /* first, look for the line-out pin */ node = parse_output_jack(codec, spec, AC_JACK_LINE_OUT); if (node) /* found, remember the PIN node */ - spec->out_pin_node = node; + spec->out_pin_node[0] = node; + else { + /* if no line-out is found, try speaker out */ + node = parse_output_jack(codec, spec, AC_JACK_SPEAKER); + if (node) + spec->out_pin_node[0] = node; + } /* look for the HP-out pin */ node = parse_output_jack(codec, spec, AC_JACK_HP_OUT); if (node) { - if (! spec->out_pin_node) - spec->out_pin_node = node; + if (! spec->out_pin_node[0]) + spec->out_pin_node[0] = node; + else + spec->out_pin_node[1] = node; } - if (! spec->out_pin_node) { + if (! spec->out_pin_node[0]) { /* no line-out or HP pins found, * then choose for the first output pin */ - spec->out_pin_node = parse_output_jack(codec, spec, -1); - if (! spec->out_pin_node) + spec->out_pin_node[0] = parse_output_jack(codec, spec, -1); + if (! spec->out_pin_node[0]) snd_printd("hda_generic: no proper output path found\n"); } @@ -505,6 +524,9 @@ static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec, if (! (node->pin_caps & AC_PINCAP_IN)) return 0; + if (defcfg_port_conn(node) == AC_JACK_PORT_NONE) + return 0; /* unconnected */ + if (node->wid_caps & AC_WCAP_DIGITAL) return 0; /* skip SPDIF */ @@ -703,12 +725,16 @@ static int check_existing_control(struct hda_codec *codec, const char *type, con static int build_output_controls(struct hda_codec *codec) { struct hda_gspec *spec = codec->spec; - int err; + static const char *types[2] = { "Master", "Headphone" }; + int i, err; - err = create_mixer(codec, spec->pcm_vol_node, spec->pcm_vol_index, - "PCM", "Playback"); - if (err < 0) - return err; + for (i = 0; i < 2 && spec->pcm_vol_node[i]; i++) { + err = create_mixer(codec, spec->pcm_vol_node[i], + spec->pcm_vol_index[i], + types[i], "Playback"); + if (err < 0) + return err; + } return 0; } @@ -805,7 +831,7 @@ static int build_loopback_controls(struct hda_codec *codec) int err; const char *type; - if (! spec->out_pin_node) + if (! spec->out_pin_node[0]) return 0; list_for_each(p, &spec->nid_list) { @@ -820,7 +846,8 @@ static int build_loopback_controls(struct hda_codec *codec) if (check_existing_control(codec, type, "Playback")) continue; clear_check_flags(spec); - err = parse_loopback_path(codec, spec, spec->out_pin_node, + err = parse_loopback_path(codec, spec, + spec->out_pin_node[0], node, type); if (err < 0) return err; @@ -855,12 +882,37 @@ static struct hda_pcm_stream generic_pcm_playback = { .channels_max = 2, }; +static int generic_pcm2_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct hda_gspec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, + stream_tag, 0, format); + return 0; +} + +static int generic_pcm2_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hda_gspec *spec = codec->spec; + + snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0); + snd_hda_codec_setup_stream(codec, spec->dac_node[1]->nid, 0, 0, 0); + return 0; +} + static int build_generic_pcms(struct hda_codec *codec) { struct hda_gspec *spec = codec->spec; struct hda_pcm *info = &spec->pcm_rec; - if (! spec->dac_node && ! spec->adc_node) { + if (! spec->dac_node[0] && ! spec->adc_node) { snd_printd("hda_generic: no PCM found\n"); return 0; } @@ -869,9 +921,13 @@ static int build_generic_pcms(struct hda_codec *codec) codec->pcm_info = info; info->name = "HDA Generic"; - if (spec->dac_node) { + if (spec->dac_node[0]) { info->stream[0] = generic_pcm_playback; - info->stream[0].nid = spec->dac_node->nid; + info->stream[0].nid = spec->dac_node[0]->nid; + if (spec->dac_node[1]) { + info->stream[0].ops.prepare = generic_pcm2_prepare; + info->stream[0].ops.cleanup = generic_pcm2_cleanup; + } } if (spec->adc_node) { info->stream[1] = generic_pcm_playback; -- cgit 1.2.3-korg From bb160b850d7e285f8b15906d9c0d1916acfdb953 Mon Sep 17 00:00:00 2001 From: Sergei Shtylyov Date: Tue, 21 Mar 2006 11:58:48 +0100 Subject: [ALSA] AMD Au1x00: fix DMA init/cleanup Modules: MIPS AU1x00 driver AMD Au1x00 ALSA driver causes kernel oops in au1000_init() by trying to set DMA channel to -1 in yet unallocated audio streams. Here's the patch that staightens up DMA init/cleanup code. Signed-off-by: Sergei Shtylyov Signed-off-by: Takashi Iwai --- sound/mips/au1x00.c | 33 ++++++++++++++++++++------------- 1 file changed, 20 insertions(+), 13 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index e66b0144e3b561..961453b77fc935 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -612,14 +612,17 @@ snd_au1000_free(struct snd_card *card) release_and_free_resource(au1000->ac97_res_port); } - if (au1000->stream[PLAYBACK]->dma >= 0) - free_au1000_dma(au1000->stream[PLAYBACK]->dma); - - if (au1000->stream[CAPTURE]->dma >= 0) - free_au1000_dma(au1000->stream[CAPTURE]->dma); + if (au1000->stream[PLAYBACK]) { + if (au1000->stream[PLAYBACK]->dma >= 0) + free_au1000_dma(au1000->stream[PLAYBACK]->dma); + kfree(au1000->stream[PLAYBACK]); + } - kfree(au1000->stream[PLAYBACK]); - kfree(au1000->stream[CAPTURE]); + if (au1000->stream[CAPTURE]) { + if (au1000->stream[CAPTURE]->dma >= 0) + free_au1000_dma(au1000->stream[CAPTURE]->dma); + kfree(au1000->stream[CAPTURE]); + } } @@ -638,15 +641,19 @@ au1000_init(void) card->private_free = snd_au1000_free; au1000 = card->private_data; - /* so that snd_au1000_free will work as intended */ au1000->card = card; - au1000->stream[PLAYBACK]->dma = -1; - au1000->stream[CAPTURE]->dma = -1; - au1000->ac97_res_port = NULL; + au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); - au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); + au1000->stream[CAPTURE ] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL); + /* so that snd_au1000_free will work as intended */ + au1000->ac97_res_port = NULL; + if (au1000->stream[PLAYBACK]) + au1000->stream[PLAYBACK]->dma = -1; + if (au1000->stream[CAPTURE ]) + au1000->stream[CAPTURE ]->dma = -1; + if (au1000->stream[PLAYBACK] == NULL || - au1000->stream[CAPTURE] == NULL) { + au1000->stream[CAPTURE ] == NULL) { snd_card_free(card); return -ENOMEM; } -- cgit 1.2.3-korg From c5c079e31cba3e6f93ef098911e216b79d0a84e8 Mon Sep 17 00:00:00 2001 From: Sergei Shtylyov Date: Tue, 21 Mar 2006 12:01:17 +0100 Subject: [ALSA] AMD Au1x00: AC'97 controller is memory mapped Modules: MIPS AU1x00 driver AMD Au1x00 ALSA driver erroneously calls request_region() for AC'97 controller registers -- the controller is actually memory mapped at addresses 0x10000000 thru 0x100FFFFF. Signed-off-by: Sergei Shtylyov Signed-off-by: Takashi Iwai --- sound/mips/au1x00.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 961453b77fc935..cf476fe1ac5616 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -561,12 +561,13 @@ snd_au1000_ac97_new(struct snd_au1000 *au1000) .read = snd_au1000_ac97_read, }; - if ((au1000->ac97_res_port = request_region(AC97C_CONFIG, - sizeof(struct au1000_ac97_reg), "Au1x00 AC97")) == NULL) { + if ((au1000->ac97_res_port = request_mem_region(CPHYSADDR(AC97C_CONFIG), + 0x100000, "Au1x00 AC97")) == NULL) { snd_printk(KERN_ERR "ALSA AC97: can't grap AC97 port\n"); return -EBUSY; } - au1000->ac97_ioport = (struct au1000_ac97_reg *) au1000->ac97_res_port->start; + au1000->ac97_ioport = (struct au1000_ac97_reg *) + KSEG1ADDR(au1000->ac97_res_port->start); spin_lock_init(&au1000->ac97_lock); -- cgit 1.2.3-korg From c6077b3000184c7f69c4798b9025e5fbd69c8c62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2006 16:07:13 +0100 Subject: [ALSA] Fix memory leaks in error path of control.c Modules: Control Midlevel Fix memory leaks in error path of control.c (only with CONFIG_SND_DEBUG=y). Signed-off-by: Takashi Iwai --- sound/core/control.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 9742bdba0de12c..574745314e7045 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -309,28 +309,29 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) { struct snd_ctl_elem_id id; unsigned int idx; + int err = -EINVAL; - snd_assert(card != NULL, return -EINVAL); if (! kcontrol) - return -EINVAL; - snd_assert(kcontrol->info != NULL, return -EINVAL); + return err; + snd_assert(card != NULL, goto error); + snd_assert(kcontrol->info != NULL, goto error); id = kcontrol->id; down_write(&card->controls_rwsem); if (snd_ctl_find_id(card, &id)) { up_write(&card->controls_rwsem); - snd_ctl_free_one(kcontrol); snd_printd(KERN_ERR "control %i:%i:%i:%s:%i is already present\n", id.iface, id.device, id.subdevice, id.name, id.index); - return -EBUSY; + err = -EBUSY; + goto error; } if (snd_ctl_find_hole(card, kcontrol->count) < 0) { up_write(&card->controls_rwsem); - snd_ctl_free_one(kcontrol); - return -ENOMEM; + err = -ENOMEM; + goto error; } list_add_tail(&kcontrol->list, &card->controls); card->controls_count += kcontrol->count; @@ -340,6 +341,10 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) for (idx = 0; idx < kcontrol->count; idx++, id.index++, id.numid++) snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_ADD, &id); return 0; + + error: + snd_ctl_free_one(kcontrol); + return err; } /** -- cgit 1.2.3-korg From b2c2844c5a486020e5d564870d114a7f4cd5ea4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2006 19:11:50 +0100 Subject: [ALSA] hda-codec - Add support for HP nx9420 laptop Modules: HDA Codec driver Added the support for HP nx9420 (AngelFire) laptop with AD1981HD codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7fbe71e69cfced..32401bd8c22940 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1322,7 +1322,9 @@ enum { AD1981_BASIC, AD1981_HP }; static struct hda_board_config ad1981_cfg_tbl[] = { { .modelname = "hp", .config = AD1981_HP }, { .pci_subvendor = 0x103c, .pci_subdevice = 0x30aa, - .config = AD1981_HP }, + .config = AD1981_HP }, /* HP nx6320 */ + { .pci_subvendor = 0x103c, .pci_subdevice = 0x309f, + .config = AD1981_HP }, /* HP nx9420 AngelFire */ { .modelname = "basic", .config = AD1981_BASIC }, {} }; -- cgit 1.2.3-korg From 84f3430c7255668a0298d166605d27e3c96b5de4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 21 Mar 2006 19:12:53 +0100 Subject: [ALSA] hda-codec - Add support for ASUS P4GPL-X Modules: HDA Codec driver Added the support ASUS P4GPL-X with ALC880 codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index de145d102d9ffc..4c6c9ec8ea5bb6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2065,6 +2065,7 @@ static struct hda_board_config alc880_cfg_tbl[] = { { .pci_subvendor = 0x1043, .pci_subdevice = 0x1123, .config = ALC880_ASUS_DIG }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x1143, .config = ALC880_ASUS }, { .pci_subvendor = 0x1043, .pci_subdevice = 0x10b3, .config = ALC880_ASUS_W1V }, + { .pci_subvendor = 0x1043, .pci_subdevice = 0x8181, .config = ALC880_ASUS_DIG }, /* ASUS P4GPL-X */ { .pci_subvendor = 0x1558, .pci_subdevice = 0x5401, .config = ALC880_ASUS_DIG2 }, { .modelname = "uniwill", .config = ALC880_UNIWILL_DIG }, -- cgit 1.2.3-korg From 9d2f928ddf64ca0361562e30faf584cd33055c60 Mon Sep 17 00:00:00 2001 From: Tobias Klauser Date: Wed, 22 Mar 2006 10:53:19 +0100 Subject: [PATCH] Intruduce DMA_28BIT_MASK This patch introduces the DMA_28BIT_MASK constant in dma-mapping.h ALSA drivers using this mask are changed to use the new constant. Signed-off-by: Tobias Klauser Acked-by: Takashi Iwai Acked-by: Jaroslav Kysela --- include/linux/dma-mapping.h | 1 + sound/pci/ad1889.c | 7 ++++--- sound/pci/emu10k1/emu10k1x.c | 13 +++++++------ sound/pci/es1968.c | 5 +++-- sound/pci/ice1712/ice1712.c | 5 +++-- sound/pci/maestro3.c | 5 +++-- sound/pci/mixart/mixart.c | 3 ++- sound/pci/pcxhr/pcxhr.c | 3 ++- 8 files changed, 25 insertions(+), 17 deletions(-) diff --git a/include/linux/dma-mapping.h b/include/linux/dma-mapping.h index 2d80cc761a1594..a8731062a74cc1 100644 --- a/include/linux/dma-mapping.h +++ b/include/linux/dma-mapping.h @@ -20,6 +20,7 @@ enum dma_data_direction { #define DMA_31BIT_MASK 0x000000007fffffffULL #define DMA_30BIT_MASK 0x000000003fffffffULL #define DMA_29BIT_MASK 0x000000001fffffffULL +#define DMA_28BIT_MASK 0x000000000fffffffULL #include diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index a208075cdc1e33..2aa5a7fdb6e067 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -34,6 +34,7 @@ #include #include +#include #include #include #include @@ -909,10 +910,10 @@ snd_ad1889_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - + /* check PCI availability (32bit DMA) */ - if (pci_set_dma_mask(pci, 0xffffffff) < 0 || - pci_set_consistent_dma_mask(pci, 0xffffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { printk(KERN_ERR PFX "error setting 32-bit DMA mask.\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 1107c8ec7f78ab..2208dbd48be97b 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include @@ -893,24 +894,24 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, static struct snd_device_ops ops = { .dev_free = snd_emu10k1x_dev_free, }; - + *rchip = NULL; - + if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, 0x0fffffff) < 0 || - pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { snd_printk(KERN_ERR "error to set 28bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; } - + chip = kzalloc(sizeof(*chip), GFP_KERNEL); if (chip == NULL) { pci_disable_device(pci); return -ENOMEM; } - + chip->card = card; chip->pci = pci; chip->irq = -1; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 6a265ab3894ee9..dd465a186e119e 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -100,6 +100,7 @@ #include #include #include +#include #include #include #include @@ -2561,8 +2562,8 @@ static int __devinit snd_es1968_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, 0x0fffffff) < 0 || - pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index b96b5d6efc5d23..672e198317e1c4 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -53,6 +53,7 @@ #include #include #include +#include #include #include #include @@ -2553,8 +2554,8 @@ static int __devinit snd_ice1712_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, 0x0fffffff) < 0 || - pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index d3ef0cc6c4f940..8bc084956c2849 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -37,6 +37,7 @@ #include #include #include +#include #include #include #include @@ -2657,8 +2658,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, return -EIO; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, 0x0fffffff) < 0 || - pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || + pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index e79fb264532b60..43ee3b2b948f95 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -1289,7 +1290,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, pci_set_master(pci); /* check if we can restrict PCI DMA transfers to 32 bits */ - if (pci_set_dma_mask(pci, 0xffffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 31a3e8e1b2343e..f679779d96e3b6 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include @@ -1217,7 +1218,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, const struct pci_device_id pci_set_master(pci); /* check if we can restrict PCI DMA transfers to 32 bits */ - if (pci_set_dma_mask(pci, 0xffffffff) < 0) { + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; -- cgit 1.2.3-korg From a3c44854a59f7e983c867060aa906bbf5befb1ef Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 22 Mar 2006 14:37:15 +0100 Subject: [ALSA] version 1.0.11rc4 --- include/sound/version.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/version.h b/include/sound/version.h index 919da0dd001cb8..4f0e65808cf1a2 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h. Generated by configure. */ -#define CONFIG_SND_VERSION "1.0.11rc2" -#define CONFIG_SND_DATE " (Wed Jan 04 08:57:20 2006 UTC)" +#define CONFIG_SND_VERSION "1.0.11rc4" +#define CONFIG_SND_DATE " (Wed Mar 22 10:27:24 2006 UTC)" -- cgit 1.2.3-korg