Copyright (c) 2002-2005 Takashi Iwai <tiwai@suse.de>
This document is free; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version.
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Abstract
This document describes how to write an ALSA (Advanced Linux Sound Architecture) driver.
Table of Contents
List of Examples
This document describes how to write an ALSA (Advanced Linux Sound Architecture) driver. The document focuses mainly on PCI soundcards. In the case of other device types, the API might be different, too. However, at least the ALSA kernel API is consistent, and therefore it would be still a bit help for writing them.
This document targets people who already have enough C language skills and have basic linux kernel programming knowledge. This document doesn't explain the general topic of linux kernel coding and doesn't cover low-level driver implementation details. It only describes the standard way to write a PCI sound driver on ALSA.
If you are already familiar with the older ALSA ver.0.5.x API, you
can check the drivers such as sound/pci/es1938.c or
sound/pci/maestro3.c which have also almost the same
code-base in the ALSA 0.5.x tree, so you can compare the differences.
This document is still a draft version. Any feedback and corrections, please!!
Table of Contents
The ALSA drivers are provided in two ways.
One is the trees provided as a tarball or via cvs from the ALSA's ftp site, and another is the 2.6 (or later) Linux kernel tree. To synchronize both, the ALSA driver tree is split into two different trees: alsa-kernel and alsa-driver. The former contains purely the source code for the Linux 2.6 (or later) tree. This tree is designed only for compilation on 2.6 or later environment. The latter, alsa-driver, contains many subtle files for compiling ALSA drivers outside of the Linux kernel tree, wrapper functions for older 2.2 and 2.4 kernels, to adapt the latest kernel API, and additional drivers which are still in development or in tests. The drivers in alsa-driver tree will be moved to alsa-kernel (and eventually to the 2.6 kernel tree) when they are finished and confirmed to work fine.
The file tree structure of ALSA driver is depicted below. Both alsa-kernel and alsa-driver have almost the same file structure, except for “core” directory. It's named as “acore” in alsa-driver tree.
Example 1.1. ALSA File Tree Structure
sound
/core
/oss
/seq
/oss
/instr
/ioctl32
/include
/drivers
/mpu401
/opl3
/i2c
/l3
/synth
/emux
/pci
/(cards)
/isa
/(cards)
/arm
/ppc
/sparc
/usb
/pcmcia /(cards)
/oss
This directory contains the middle layer which is the heart of ALSA drivers. In this directory, the native ALSA modules are stored. The sub-directories contain different modules and are dependent upon the kernel config.
The codes for PCM and mixer OSS emulation modules are stored
in this directory. The rawmidi OSS emulation is included in
the ALSA rawmidi code since it's quite small. The sequencer
code is stored in core/seq/oss directory (see
below).
This directory contains the 32bit-ioctl wrappers for 64bit architectures such like x86-64, ppc64 and sparc64. For 32bit and alpha architectures, these are not compiled.
This directory and its sub-directories are for the ALSA
sequencer. This directory contains the sequencer core and
primary sequencer modules such like snd-seq-midi,
snd-seq-virmidi, etc. They are compiled only when
CONFIG_SND_SEQUENCER is set in the kernel
config.
This is the place for the public header files of ALSA drivers, which are to be exported to user-space, or included by several files at different directories. Basically, the private header files should not be placed in this directory, but you may still find files there, due to historical reasons :)
This directory contains code shared among different drivers on different architectures. They are hence supposed not to be architecture-specific. For example, the dummy pcm driver and the serial MIDI driver are found in this directory. In the sub-directories, there is code for components which are independent from bus and cpu architectures.
This contains the ALSA i2c components.
Although there is a standard i2c layer on Linux, ALSA has its own i2c code for some cards, because the soundcard needs only a simple operation and the standard i2c API is too complicated for such a purpose.
This contains the synth middle-level modules.
So far, there is only Emu8000/Emu10k1 synth driver under
the synth/emux sub-directory.
This directory and its sub-directories hold the top-level card modules for PCI soundcards and the code specific to the PCI BUS.
The drivers compiled from a single file are stored directly in the pci directory, while the drivers with several source files are stored on their own sub-directory (e.g. emu10k1, ice1712).
This directory and its sub-directories hold the top-level card modules for ISA soundcards.
They are used for top-level card modules which are specific to one of these architectures.
This directory contains the USB-audio driver. In the latest version, the USB MIDI driver is integrated in the usb-audio driver.
The PCMCIA, especially PCCard drivers will go here. CardBus drivers will be in the pci directory, because their API is identical to that of standard PCI cards.
Table of Contents
The minimum flow for PCI soundcards is as follows:
define the PCI ID table (see the section PCI Entries ).
create probe() callback.
create remove() callback.
create a pci_driver structure containing the three pointers above.
create an init() function just calling
the pci_register_driver() to register the pci_driver table
defined above.
create an exit() function to call
the pci_unregister_driver() function.
The code example is shown below. Some parts are kept
unimplemented at this moment but will be filled in the
next sections. The numbers in the comment lines of the
snd_mychip_probe() function
refer to details explained in the following section.
Example 2.1. Basic Flow for PCI Drivers - Example
#include <linux/init.h>
#include <linux/pci.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/initval.h>
/* module parameters (see "Module Parameters") */
/* SNDRV_CARDS: maximum number of cards supported by this module */
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
/* definition of the chip-specific record */
struct mychip {
struct snd_card *card;
/* the rest of the implementation will be in section
* "PCI Resource Management"
*/
};
/* chip-specific destructor
* (see "PCI Resource Management")
*/
static int snd_mychip_free(struct mychip *chip)
{
.... /* will be implemented later... */
}
/* component-destructor
* (see "Management of Cards and Components")
*/
static int snd_mychip_dev_free(struct snd_device *device)
{
return snd_mychip_free(device->device_data);
}
/* chip-specific constructor
* (see "Management of Cards and Components")
*/
static int __devinit snd_mychip_create(struct snd_card *card,
struct pci_dev *pci,
struct mychip **rchip)
{
struct mychip *chip;
int err;
static struct snd_device_ops ops = {
.dev_free = snd_mychip_dev_free,
};
*rchip = NULL;
/* check PCI availability here
* (see "PCI Resource Management")
*/
....
/* allocate a chip-specific data with zero filled */
chip = kzalloc(sizeof(*chip), GFP_KERNEL);
if (chip == NULL)
return -ENOMEM;
chip->card = card;
/* rest of initialization here; will be implemented
* later, see "PCI Resource Management"
*/
....
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
snd_mychip_free(chip);
return err;
}
snd_card_set_dev(card, &pci->dev);
*rchip = chip;
return 0;
}
/* constructor -- see "Constructor" sub-section */
static int __devinit snd_mychip_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
static int dev;
struct snd_card *card;
struct mychip *chip;
int err;
/* (1) */
if (dev >= SNDRV_CARDS)
return -ENODEV;
if (!enable[dev]) {
dev++;
return -ENOENT;
}
/* (2) */
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
if (err < 0)
return err;
/* (3) */
err = snd_mychip_create(card, pci, &chip);
if (err < 0) {
snd_card_free(card);
return err;
}
/* (4) */
strcpy(card->driver, "My Chip");
strcpy(card->shortname, "My Own Chip 123");
sprintf(card->longname, "%s at 0x%lx irq %i",
card->shortname, chip->ioport, chip->irq);
/* (5) */
.... /* implemented later */
/* (6) */
err = snd_card_register(card);
if (err < 0) {
snd_card_free(card);
return err;
}
/* (7) */
pci_set_drvdata(pci, card);
dev++;
return 0;
}
/* destructor -- see the "Destructor" sub-section */
static void __devexit snd_mychip_remove(struct pci_dev *pci)
{
snd_card_free(pci_get_drvdata(pci));
pci_set_drvdata(pci, NULL);
}
The real constructor of PCI drivers is the probe callback.
The probe callback and other component-constructors which are called
from the probe callback should be defined with
the __devinit prefix. You
cannot use the __init prefix for them,
because any PCI device could be a hotplug device.
In the probe callback, the following scheme is often used.
static int dev;
....
if (dev >= SNDRV_CARDS)
return -ENODEV;
if (!enable[dev]) {
dev++;
return -ENOENT;
}
where enable[dev] is the module option.
Each time the probe callback is called, check the
availability of the device. If not available, simply increment
the device index and returns. dev will be incremented also
later (step
7).
struct snd_card *card;
int err;
....
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
The details will be explained in the section Management of Cards and Components.
In this part, the PCI resources are allocated.
struct mychip *chip;
....
err = snd_mychip_create(card, pci, &chip);
if (err < 0) {
snd_card_free(card);
return err;
}
The details will be explained in the section PCI Resource Management.
strcpy(card->driver, "My Chip");
strcpy(card->shortname, "My Own Chip 123");
sprintf(card->longname, "%s at 0x%lx irq %i",
card->shortname, chip->ioport, chip->irq);
The driver field holds the minimal ID string of the chip. This is used by alsa-lib's configurator, so keep it simple but unique. Even the same driver can have different driver IDs to distinguish the functionality of each chip type.
The shortname field is a string shown as more verbose
name. The longname field contains the information
shown in /proc/asound/cards.
Here you define the basic components such as PCM, mixer (e.g. AC97), MIDI (e.g. MPU-401), and other interfaces. Also, if you want a proc file, define it here, too.
err = snd_card_register(card);
if (err < 0) {
snd_card_free(card);
return err;
}
Will be explained in the section Management of Cards and Components, too.
The destructor, remove callback, simply releases the card instance. Then the ALSA middle layer will release all the attached components automatically.
It would be typically like the following:
static void __devexit snd_mychip_remove(struct pci_dev *pci)
{
snd_card_free(pci_get_drvdata(pci));
pci_set_drvdata(pci, NULL);
}
The above code assumes that the card pointer is set to the PCI driver data.
For the above example, at least the following include files are necessary.
#include <linux/init.h>
#include <linux/pci.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/initval.h>
where the last one is necessary only when module options are defined in the source file. If the code is split into several files, the files without module options don't need them.
In addition to these headers, you'll need
<linux/interrupt.h> for interrupt
handling, and <asm/io.h> for I/O
access. If you use the mdelay() or
udelay() functions, you'll need to include
<linux/delay.h> too.
The ALSA interfaces like the PCM and control APIs are defined in other
<sound/xxx.h> header files.
They have to be included after
<sound/core.h>.
Table of Contents
For each soundcard, a “card” record must be allocated.
A card record is the headquarters of the soundcard. It manages the whole list of devices (components) on the soundcard, such as PCM, mixers, MIDI, synthesizer, and so on. Also, the card record holds the ID and the name strings of the card, manages the root of proc files, and controls the power-management states and hotplug disconnections. The component list on the card record is used to manage the correct release of resources at destruction.
As mentioned above, to create a card instance, call
snd_card_create().
struct snd_card *card;
int err;
err = snd_card_create(index, id, module, extra_size, &card);
The function takes five arguments, the card-index number, the
id string, the module pointer (usually
THIS_MODULE),
the size of extra-data space, and the pointer to return the
card instance. The extra_size argument is used to
allocate card->private_data for the
chip-specific data. Note that these data
are allocated by snd_card_create().
After the card is created, you can attach the components (devices) to the card instance. In an ALSA driver, a component is represented as a struct snd_device object. A component can be a PCM instance, a control interface, a raw MIDI interface, etc. Each such instance has one component entry.
A component can be created via
snd_device_new() function.
snd_device_new(card, SNDRV_DEV_XXX, chip, &ops);
This takes the card pointer, the device-level
(SNDRV_DEV_XXX), the data pointer, and the
callback pointers (&ops). The
device-level defines the type of components and the order of
registration and de-registration. For most components, the
device-level is already defined. For a user-defined component,
you can use SNDRV_DEV_LOWLEVEL.
This function itself doesn't allocate the data space. The data
must be allocated manually beforehand, and its pointer is passed
as the argument. This pointer is used as the
(chip identifier in the above example)
for the instance.
Each pre-defined ALSA component such as ac97 and pcm calls
snd_device_new() inside its
constructor. The destructor for each component is defined in the
callback pointers. Hence, you don't need to take care of
calling a destructor for such a component.
If you wish to create your own component, you need to
set the destructor function to the dev_free callback in
the ops, so that it can be released
automatically via snd_card_free().
The next example will show an implementation of chip-specific
data.
Chip-specific information, e.g. the I/O port address, its resource pointer, or the irq number, is stored in the chip-specific record.
struct mychip {
....
};
In general, there are two ways of allocating the chip record.
As mentioned above, you can pass the extra-data-length
to the 4th argument of snd_card_create(), i.e.
err = snd_card_create(index[dev], id[dev], THIS_MODULE,
sizeof(struct mychip), &card);
struct mychip is the type of the chip record.
In return, the allocated record can be accessed as
struct mychip *chip = card->private_data;
With this method, you don't have to allocate twice. The record is released together with the card instance.
After allocating a card instance via
snd_card_create() (with
0 on the 4th arg), call
kzalloc().
struct snd_card *card;
struct mychip *chip;
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
.....
chip = kzalloc(sizeof(*chip), GFP_KERNEL);
The chip record should have the field to hold the card pointer at least,
struct mychip {
struct snd_card *card;
....
};
Then, set the card pointer in the returned chip instance.
chip->card = card;
Next, initialize the fields, and register this chip
record as a low-level device with a specified
ops,
static struct snd_device_ops ops = {
.dev_free = snd_mychip_dev_free,
};
....
snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
snd_mychip_dev_free() is the
device-destructor function, which will call the real
destructor.
static int snd_mychip_dev_free(struct snd_device *device)
{
return snd_mychip_free(device->device_data);
}
where snd_mychip_free() is the real destructor.
After all components are assigned, register the card instance
by calling snd_card_register(). Access
to the device files is enabled at this point. That is, before
snd_card_register() is called, the
components are safely inaccessible from external side. If this
call fails, exit the probe function after releasing the card via
snd_card_free().
For releasing the card instance, you can call simply
snd_card_free(). As mentioned earlier, all
components are released automatically by this call.
As further notes, the destructors (both
snd_mychip_dev_free and
snd_mychip_free) cannot be defined with
the __devexit prefix, because they may be
called from the constructor, too, at the false path.
For a device which allows hotplugging, you can use
snd_card_free_when_closed. This one will
postpone the destruction until all devices are closed.
Table of Contents
In this section, we'll complete the chip-specific constructor, destructor and PCI entries. Example code is shown first, below.
Example 4.1. PCI Resource Management Example
struct mychip {
struct snd_card *card;
struct pci_dev *pci;
unsigned long port;
int irq;
};
static int snd_mychip_free(struct mychip *chip)
{
/* disable hardware here if any */
.... /* (not implemented in this document) */
/* release the irq */
if (chip->irq >= 0)
free_irq(chip->irq, chip);
/* release the I/O ports & memory */
pci_release_regions(chip->pci);
/* disable the PCI entry */
pci_disable_device(chip->pci);
/* release the data */
kfree(chip);
return 0;
}
/* chip-specific constructor */
static int __devinit snd_mychip_create(struct snd_card *card,
struct pci_dev *pci,
struct mychip **rchip)
{
struct mychip *chip;
int err;
static struct snd_device_ops ops = {
.dev_free = snd_mychip_dev_free,
};
*rchip = NULL;
/* initialize the PCI entry */
err = pci_enable_device(pci);
if (err < 0)
return err;
/* check PCI availability (28bit DMA) */
if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 ||
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) {
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;
}
/* initialize the stuff */
chip->card = card;
chip->pci = pci;
chip->irq = -1;
/* (1) PCI resource allocation */
err = pci_request_regions(pci, "My Chip");
if (err < 0) {
kfree(chip);
pci_disable_device(pci);
return err;
}
chip->port = pci_resource_start(pci, 0);
if (request_irq(pci->irq, snd_mychip_interrupt,
IRQF_SHARED, "My Chip", chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
}
chip->irq = pci->irq;
/* (2) initialization of the chip hardware */
.... /* (not implemented in this document) */
err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
if (err < 0) {
snd_mychip_free(chip);
return err;
}
snd_card_set_dev(card, &pci->dev);
*rchip = chip;
return 0;
}
/* PCI IDs */
static struct pci_device_id snd_mychip_ids[] = {
{ PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
....
{ 0, }
};
MODULE_DEVICE_TABLE(pci, snd_mychip_ids);
/* pci_driver definition */
static struct pci_driver driver = {
.name = "My Own Chip",
.id_table = snd_mychip_ids,
.probe = snd_mychip_probe,
.remove = __devexit_p(snd_mychip_remove),
};
/* module initialization */
static int __init alsa_card_mychip_init(void)
{
return pci_register_driver(&driver);
}
/* module clean up */
static void __exit alsa_card_mychip_exit(void)
{
pci_unregister_driver(&driver);
}
module_init(alsa_card_mychip_init)
module_exit(alsa_card_mychip_exit)
EXPORT_NO_SYMBOLS; /* for old kernels only */
The allocation of PCI resources is done in the
probe() function, and usually an extra
xxx_create() function is written for this
purpose.
In the case of PCI devices, you first have to call
the pci_enable_device() function before
allocating resources. Also, you need to set the proper PCI DMA
mask to limit the accessed I/O range. In some cases, you might
need to call pci_set_master() function,
too.
Suppose the 28bit mask, and the code to be added would be like:
err = pci_enable_device(pci);
if (err < 0)
return err;
if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 ||
pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) {
printk(KERN_ERR "error to set 28bit mask DMA\n");
pci_disable_device(pci);
return -ENXIO;
}
The allocation of I/O ports and irqs is done via standard kernel functions. Unlike ALSA ver.0.5.x., there are no helpers for that. And these resources must be released in the destructor function (see below). Also, on ALSA 0.9.x, you don't need to allocate (pseudo-)DMA for PCI like in ALSA 0.5.x.
Now assume that the PCI device has an I/O port with 8 bytes and an interrupt. Then struct mychip will have the following fields:
struct mychip {
struct snd_card *card;
unsigned long port;
int irq;
};
For an I/O port (and also a memory region), you need to have
the resource pointer for the standard resource management. For
an irq, you have to keep only the irq number (integer). But you
need to initialize this number as -1 before actual allocation,
since irq 0 is valid. The port address and its resource pointer
can be initialized as null by
kzalloc() automatically, so you
don't have to take care of resetting them.
The allocation of an I/O port is done like this:
err = pci_request_regions(pci, "My Chip");
if (err < 0) {
kfree(chip);
pci_disable_device(pci);
return err;
}
chip->port = pci_resource_start(pci, 0);
It will reserve the I/O port region of 8 bytes of the given
PCI device. The returned value, chip->res_port, is allocated
via kmalloc() by
request_region(). The pointer must be
released via kfree(), but there is a
problem with this. This issue will be explained later.
The allocation of an interrupt source is done like this:
if (request_irq(pci->irq, snd_mychip_interrupt,
IRQF_SHARED, "My Chip", chip)) {
printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
snd_mychip_free(chip);
return -EBUSY;
}
chip->irq = pci->irq;
where snd_mychip_interrupt() is the
interrupt handler defined later.
Note that chip->irq should be defined
only when request_irq() succeeded.
On the PCI bus, interrupts can be shared. Thus,
IRQF_SHARED is used as the interrupt flag of
request_irq().
The last argument of request_irq() is the
data pointer passed to the interrupt handler. Usually, the
chip-specific record is used for that, but you can use what you
like, too.
I won't give details about the interrupt handler at this point, but at least its appearance can be explained now. The interrupt handler looks usually like the following:
static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
{
struct mychip *chip = dev_id;
....
return IRQ_HANDLED;
}
Now let's write the corresponding destructor for the resources above. The role of destructor is simple: disable the hardware (if already activated) and release the resources. So far, we have no hardware part, so the disabling code is not written here.
To release the resources, the “check-and-release” method is a safer way. For the interrupt, do like this:
if (chip->irq >= 0)
free_irq(chip->irq, chip);
Since the irq number can start from 0, you should initialize chip->irq with a negative value (e.g. -1), so that you can check the validity of the irq number as above.
When you requested I/O ports or memory regions via
pci_request_region() or
pci_request_regions() like in this example,
release the resource(s) using the corresponding function,
pci_release_region() or
pci_release_regions().
pci_release_regions(chip->pci);
When you requested manually via request_region()
or request_mem_region, you can release it via
release_resource(). Suppose that you keep
the resource pointer returned from request_region()
in chip->res_port, the release procedure looks like:
release_and_free_resource(chip->res_port);
Don't forget to call pci_disable_device()
before the end.
And finally, release the chip-specific record.
kfree(chip);
Again, remember that you cannot
use the __devexit prefix for this destructor.
We didn't implement the hardware disabling part in the above. If you need to do this, please note that the destructor may be called even before the initialization of the chip is completed. It would be better to have a flag to skip hardware disabling if the hardware was not initialized yet.
When the chip-data is assigned to the card using
snd_device_new() with
SNDRV_DEV_LOWLELVEL , its destructor is
called at the last. That is, it is assured that all other
components like PCMs and controls have already been released.
You don't have to stop PCMs, etc. explicitly, but just
call low-level hardware stopping.
The management of a memory-mapped region is almost as same as the management of an I/O port. You'll need three fields like the following:
struct mychip {
....
unsigned long iobase_phys;
void __iomem *iobase_virt;
};
and the allocation would be like below:
if ((err = pci_request_regions(pci, "My Chip")) < 0) {
kfree(chip);
return err;
}
chip->iobase_phys = pci_resource_start(pci, 0);
chip->iobase_virt = ioremap_nocache(chip->iobase_phys,
pci_resource_len(pci, 0));
and the corresponding destructor would be:
static int snd_mychip_free(struct mychip *chip)
{
....
if (chip->iobase_virt)
iounmap(chip->iobase_virt);
....
pci_release_regions(chip->pci);
....
}
At some point, typically after calling snd_device_new(),
you need to register the struct device of the chip
you're handling for udev and co. ALSA provides a macro for compatibility with
older kernels. Simply call like the following:
snd_card_set_dev(card, &pci->dev);
so that it stores the PCI's device pointer to the card. This will be referred by ALSA core functions later when the devices are registered.
In the case of non-PCI, pass the proper device struct pointer of the BUS instead. (In the case of legacy ISA without PnP, you don't have to do anything.)
So far, so good. Let's finish the missing PCI stuff. At first, we need a pci_device_id table for this chipset. It's a table of PCI vendor/device ID number, and some masks.
For example,
static struct pci_device_id snd_mychip_ids[] = {
{ PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
....
{ 0, }
};
MODULE_DEVICE_TABLE(pci, snd_mychip_ids);
The first and second fields of the pci_device_id structure are the vendor and device IDs. If you have no reason to filter the matching devices, you can leave the remaining fields as above. The last field of the pci_device_id struct contains private data for this entry. You can specify any value here, for example, to define specific operations for supported device IDs. Such an example is found in the intel8x0 driver.
The last entry of this list is the terminator. You must specify this all-zero entry.
Then, prepare the pci_driver record:
static struct pci_driver driver = {
.name = "My Own Chip",
.id_table = snd_mychip_ids,
.probe = snd_mychip_probe,
.remove = __devexit_p(snd_mychip_remove),
};
The probe and
remove functions have already
been defined in the previous sections.
The remove function should
be defined with the
__devexit_p() macro, so that it's not
defined for built-in (and non-hot-pluggable) case. The
name
field is the name string of this device. Note that you must not
use a slash “/” in this string.
And at last, the module entries:
static int __init alsa_card_mychip_init(void)
{
return pci_register_driver(&driver);
}
static void __exit alsa_card_mychip_exit(void)
{
pci_unregister_driver(&driver);
}
module_init(alsa_card_mychip_init)
module_exit(alsa_card_mychip_exit)
Note that these module entries are tagged with
__init and
__exit prefixes, not
__devinit nor
__devexit.
Oh, one thing was forgotten. If you have no exported symbols, you need to declare it in 2.2 or 2.4 kernels (it's not necessary in 2.6 kernels).
EXPORT_NO_SYMBOLS;
That's all!
Table of Contents
The PCM middle layer of ALSA is quite powerful and it is only necessary for each driver to implement the low-level functions to access its hardware.
For accessing to the PCM layer, you need to include
<sound/pcm.h> first. In addition,
<sound/pcm_params.h> might be needed
if you access to some functions related with hw_param.
Each card device can have up to four pcm instances. A pcm instance corresponds to a pcm device file. The limitation of number of instances comes only from the available bit size of the Linux's device numbers. Once when 64bit device number is used, we'll have more pcm instances available.
A pcm instance consists of pcm playback and capture streams,
and each pcm stream consists of one or more pcm substreams. Some
soundcards support multiple playback functions. For example,
emu10k1 has a PCM playback of 32 stereo substreams. In this case, at
each open, a free substream is (usually) automatically chosen
and opened. Meanwhile, when only one substream exists and it was
already opened, the successful open will either block
or error with EAGAIN according to the
file open mode. But you don't have to care about such details in your
driver. The PCM middle layer will take care of such work.
The example code below does not include any hardware access routines but shows only the skeleton, how to build up the PCM interfaces.
Example 5.1. PCM Example Code
#include <sound/pcm.h>
....
/* hardware definition */
static struct snd_pcm_hardware snd_mychip_playback_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 32768,
.period_bytes_min = 4096,
.period_bytes_max = 32768,
.periods_min = 1,
.periods_max = 1024,
};
/* hardware definition */
static struct snd_pcm_hardware snd_mychip_capture_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 32768,
.period_bytes_min = 4096,
.period_bytes_max = 32768,
.periods_min = 1,
.periods_max = 1024,
};
/* open callback */
static int snd_mychip_playback_open(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_mychip_playback_hw;
/* more hardware-initialization will be done here */
....
return 0;
}
/* close callback */
static int snd_mychip_playback_close(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
/* the hardware-specific codes will be here */
....
return 0;
}
/* open callback */
static int snd_mychip_capture_open(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_mychip_capture_hw;
/* more hardware-initialization will be done here */
....
return 0;
}
/* close callback */
static int snd_mychip_capture_close(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
/* the hardware-specific codes will be here */
....
return 0;
}
/* hw_params callback */
static int snd_mychip_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params)
{
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
}
/* hw_free callback */
static int snd_mychip_pcm_hw_free(struct snd_pcm_substream *substream)
{
return snd_pcm_lib_free_pages(substream);
}
/* prepare callback */
static int snd_mychip_pcm_prepare(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
/* set up the hardware with the current configuration
* for example...
*/
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_addr,
chip->buffer_size,
chip->period_size);
return 0;
}
/* trigger callback */
static int snd_mychip_pcm_trigger(struct snd_pcm_substream *substream,
int cmd)
{
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* do something to start the PCM engine */
....
break;
case SNDRV_PCM_TRIGGER_STOP:
/* do something to stop the PCM engine */
....
break;
default:
return -EINVAL;
}
}
/* pointer callback */
static snd_pcm_uframes_t
snd_mychip_pcm_pointer(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
unsigned int current_ptr;
/* get the current hardware pointer */
current_ptr = mychip_get_hw_pointer(chip);
return current_ptr;
}
/* operators */
static struct snd_pcm_ops snd_mychip_playback_ops = {
.open = snd_mychip_playback_open,
.close = snd_mychip_playback_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_mychip_pcm_hw_params,
.hw_free = snd_mychip_pcm_hw_free,
.prepare = snd_mychip_pcm_prepare,
.trigger = snd_mychip_pcm_trigger,
.pointer = snd_mychip_pcm_pointer,
};
/* operators */
static struct snd_pcm_ops snd_mychip_capture_ops = {
.open = snd_mychip_capture_open,
.close = snd_mychip_capture_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_mychip_pcm_hw_params,
.hw_free = snd_mychip_pcm_hw_free,
.prepare = snd_mychip_pcm_prepare,
.trigger = snd_mychip_pcm_trigger,
.pointer = snd_mychip_pcm_pointer,
};
/*
* definitions of capture are omitted here...
*/
/* create a pcm device */
static int __devinit snd_mychip_new_pcm(struct mychip *chip)
{
struct snd_pcm *pcm;
int err;
err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
strcpy(pcm->name, "My Chip");
chip->pcm = pcm;
/* set operators */
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
&snd_mychip_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
&snd_mychip_capture_ops);
/* pre-allocation of buffers */
/* NOTE: this may fail */
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(chip->pci),
64*1024, 64*1024);
return 0;
}
A pcm instance is allocated by the snd_pcm_new()
function. It would be better to create a constructor for pcm,
namely,
static int __devinit snd_mychip_new_pcm(struct mychip *chip)
{
struct snd_pcm *pcm;
int err;
err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1, &pcm);
if (err < 0)
return err;
pcm->private_data = chip;
strcpy(pcm->name, "My Chip");
chip->pcm = pcm;
....
return 0;
}
The snd_pcm_new() function takes four
arguments. The first argument is the card pointer to which this
pcm is assigned, and the second is the ID string.
The third argument (index, 0 in the
above) is the index of this new pcm. It begins from zero. If
you create more than one pcm instances, specify the
different numbers in this argument. For example,
index = 1 for the second PCM device.
The fourth and fifth arguments are the number of substreams for playback and capture, respectively. Here 1 is used for both arguments. When no playback or capture substreams are available, pass 0 to the corresponding argument.
If a chip supports multiple playbacks or captures, you can specify more numbers, but they must be handled properly in open/close, etc. callbacks. When you need to know which substream you are referring to, then it can be obtained from struct snd_pcm_substream data passed to each callback as follows:
struct snd_pcm_substream *substream;
int index = substream->number;
After the pcm is created, you need to set operators for each pcm stream.
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
&snd_mychip_playback_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
&snd_mychip_capture_ops);
The operators are defined typically like this:
static struct snd_pcm_ops snd_mychip_playback_ops = {
.open = snd_mychip_pcm_open,
.close = snd_mychip_pcm_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_mychip_pcm_hw_params,
.hw_free = snd_mychip_pcm_hw_free,
.prepare = snd_mychip_pcm_prepare,
.trigger = snd_mychip_pcm_trigger,
.pointer = snd_mychip_pcm_pointer,
};
All the callbacks are described in the Operators subsection.
After setting the operators, you probably will want to pre-allocate the buffer. For the pre-allocation, simply call the following:
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(chip->pci),
64*1024, 64*1024);
It will allocate a buffer up to 64kB as default. Buffer management details will be described in the later section Buffer and Memory Management.
Additionally, you can set some extra information for this pcm
in pcm->info_flags.
The available values are defined as
SNDRV_PCM_INFO_XXX in
<sound/asound.h>, which is used for
the hardware definition (described later). When your soundchip
supports only half-duplex, specify like this:
pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
The destructor for a pcm instance is not always necessary. Since the pcm device will be released by the middle layer code automatically, you don't have to call the destructor explicitly.
The destructor would be necessary if you created special records internally and needed to release them. In such a case, set the destructor function to pcm->private_free:
Example 5.2. PCM Instance with a Destructor
static void mychip_pcm_free(struct snd_pcm *pcm)
{
struct mychip *chip = snd_pcm_chip(pcm);
/* free your own data */
kfree(chip->my_private_pcm_data);
/* do what you like else */
....
}
static int __devinit snd_mychip_new_pcm(struct mychip *chip)
{
struct snd_pcm *pcm;
....
/* allocate your own data */
chip->my_private_pcm_data = kmalloc(...);
/* set the destructor */
pcm->private_data = chip;
pcm->private_free = mychip_pcm_free;
....
}
When the PCM substream is opened, a PCM runtime instance is
allocated and assigned to the substream. This pointer is
accessible via substream->runtime.
This runtime pointer holds most information you need
to control the PCM: the copy of hw_params and sw_params configurations, the buffer
pointers, mmap records, spinlocks, etc.
The definition of runtime instance is found in
<sound/pcm.h>. Here are
the contents of this file:
struct _snd_pcm_runtime {
/* -- Status -- */
struct snd_pcm_substream *trigger_master;
snd_timestamp_t trigger_tstamp; /* trigger timestamp */
int overrange;
snd_pcm_uframes_t avail_max;
snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */
snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/
/* -- HW params -- */
snd_pcm_access_t access; /* access mode */
snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */
snd_pcm_subformat_t subformat; /* subformat */
unsigned int rate; /* rate in Hz */
unsigned int channels; /* channels */
snd_pcm_uframes_t period_size; /* period size */
unsigned int periods; /* periods */
snd_pcm_uframes_t buffer_size; /* buffer size */
unsigned int tick_time; /* tick time */
snd_pcm_uframes_t min_align; /* Min alignment for the format */
size_t byte_align;
unsigned int frame_bits;
unsigned int sample_bits;
unsigned int info;
unsigned int rate_num;
unsigned int rate_den;
/* -- SW params -- */
struct timespec tstamp_mode; /* mmap timestamp is updated */
unsigned int period_step;
unsigned int sleep_min; /* min ticks to sleep */
snd_pcm_uframes_t start_threshold;
snd_pcm_uframes_t stop_threshold;
snd_pcm_uframes_t silence_threshold; /* Silence filling happens when
noise is nearest than this */
snd_pcm_uframes_t silence_size; /* Silence filling size */
snd_pcm_uframes_t boundary; /* pointers wrap point */
snd_pcm_uframes_t silenced_start;
snd_pcm_uframes_t silenced_size;
snd_pcm_sync_id_t sync; /* hardware synchronization ID */
/* -- mmap -- */
volatile struct snd_pcm_mmap_status *status;
volatile struct snd_pcm_mmap_control *control;
atomic_t mmap_count;
/* -- locking / scheduling -- */
spinlock_t lock;
wait_queue_head_t sleep;
struct timer_list tick_timer;
struct fasync_struct *fasync;
/* -- private section -- */
void *private_data;
void (*private_free)(struct snd_pcm_runtime *runtime);
/* -- hardware description -- */
struct snd_pcm_hardware hw;
struct snd_pcm_hw_constraints hw_constraints;
/* -- interrupt callbacks -- */
void (*transfer_ack_begin)(struct snd_pcm_substream *substream);
void (*transfer_ack_end)(struct snd_pcm_substream *substream);
/* -- timer -- */
unsigned int timer_resolution; /* timer resolution */
/* -- DMA -- */
unsigned char *dma_area; /* DMA area */
dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */
size_t dma_bytes; /* size of DMA area */
struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */
#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
/* -- OSS things -- */
struct snd_pcm_oss_runtime oss;
#endif
};
For the operators (callbacks) of each sound driver, most of
these records are supposed to be read-only. Only the PCM
middle-layer changes / updates them. The exceptions are
the hardware description (hw), interrupt callbacks
(transfer_ack_xxx), DMA buffer information, and the private
data. Besides, if you use the standard buffer allocation
method via snd_pcm_lib_malloc_pages(),
you don't need to set the DMA buffer information by yourself.
In the sections below, important records are explained.
The hardware descriptor (struct snd_pcm_hardware)
contains the definitions of the fundamental hardware
configuration. Above all, you'll need to define this in
the open callback.
Note that the runtime instance holds the copy of the
descriptor, not the pointer to the existing descriptor. That
is, in the open callback, you can modify the copied descriptor
(runtime->hw) as you need. For example, if the maximum
number of channels is 1 only on some chip models, you can
still use the same hardware descriptor and change the
channels_max later:
struct snd_pcm_runtime *runtime = substream->runtime;
...
runtime->hw = snd_mychip_playback_hw; /* common definition */
if (chip->model == VERY_OLD_ONE)
runtime->hw.channels_max = 1;
Typically, you'll have a hardware descriptor as below:
static struct snd_pcm_hardware snd_mychip_playback_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_8000_48000,
.rate_min = 8000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = 32768,
.period_bytes_min = 4096,
.period_bytes_max = 32768,
.periods_min = 1,
.periods_max = 1024,
};
The info field contains the type and
capabilities of this pcm. The bit flags are defined in
<sound/asound.h> as
SNDRV_PCM_INFO_XXX. Here, at least, you
have to specify whether the mmap is supported and which
interleaved format is supported.
When the is supported, add the
SNDRV_PCM_INFO_MMAP flag here. When the
hardware supports the interleaved or the non-interleaved
formats, SNDRV_PCM_INFO_INTERLEAVED or
SNDRV_PCM_INFO_NONINTERLEAVED flag must
be set, respectively. If both are supported, you can set both,
too.
In the above example, MMAP_VALID and
BLOCK_TRANSFER are specified for the OSS mmap
mode. Usually both are set. Of course,
MMAP_VALID is set only if the mmap is
really supported.
The other possible flags are
SNDRV_PCM_INFO_PAUSE and
SNDRV_PCM_INFO_RESUME. The
PAUSE bit means that the pcm supports the
“pause” operation, while the
RESUME bit means that the pcm supports
the full “suspend/resume” operation.
If the PAUSE flag is set,
the trigger callback below
must handle the corresponding (pause push/release) commands.
The suspend/resume trigger commands can be defined even without
the RESUME flag. See
Power Management section for details.
When the PCM substreams can be synchronized (typically,
synchronized start/stop of a playback and a capture streams),
you can give SNDRV_PCM_INFO_SYNC_START,
too. In this case, you'll need to check the linked-list of
PCM substreams in the trigger callback. This will be
described in the later section.
formats field contains the bit-flags
of supported formats (SNDRV_PCM_FMTBIT_XXX).
If the hardware supports more than one format, give all or'ed
bits. In the example above, the signed 16bit little-endian
format is specified.
rates field contains the bit-flags of
supported rates (SNDRV_PCM_RATE_XXX).
When the chip supports continuous rates, pass
CONTINUOUS bit additionally.
The pre-defined rate bits are provided only for typical
rates. If your chip supports unconventional rates, you need to add
the KNOT bit and set up the hardware
constraint manually (explained later).
rate_min and
rate_max define the minimum and
maximum sample rate. This should correspond somehow to
rates bits.
channel_min and
channel_max
define, as you might already expected, the minimum and maximum
number of channels.
buffer_bytes_max defines the
maximum buffer size in bytes. There is no
buffer_bytes_min field, since
it can be calculated from the minimum period size and the
minimum number of periods.
Meanwhile, period_bytes_min and
define the minimum and maximum size of the period in bytes.
periods_max and
periods_min define the maximum and
minimum number of periods in the buffer.
The “period” is a term that corresponds to a fragment in the OSS world. The period defines the size at which a PCM interrupt is generated. This size strongly depends on the hardware. Generally, the smaller period size will give you more interrupts, that is, more controls. In the case of capture, this size defines the input latency. On the other hand, the whole buffer size defines the output latency for the playback direction.
There is also a field fifo_size.
This specifies the size of the hardware FIFO, but currently it
is neither used in the driver nor in the alsa-lib. So, you
can ignore this field.
Ok, let's go back again to the PCM runtime records.
The most frequently referred records in the runtime instance are
the PCM configurations.
The PCM configurations are stored in the runtime instance
after the application sends hw_params data via
alsa-lib. There are many fields copied from hw_params and
sw_params structs. For example,
format holds the format type
chosen by the application. This field contains the enum value
SNDRV_PCM_FORMAT_XXX.
One thing to be noted is that the configured buffer and period
sizes are stored in “frames” in the runtime.
In the ALSA world, 1 frame = channels * samples-size.
For conversion between frames and bytes, you can use the
frames_to_bytes() and
bytes_to_frames() helper functions.
period_bytes = frames_to_bytes(runtime, runtime->period_size);
Also, many software parameters (sw_params) are stored in frames, too. Please check the type of the field. snd_pcm_uframes_t is for the frames as unsigned integer while snd_pcm_sframes_t is for the frames as signed integer.
The DMA buffer is defined by the following four fields,
dma_area,
dma_addr,
dma_bytes and
dma_private.
The dma_area holds the buffer
pointer (the logical address). You can call
memcpy from/to
this pointer. Meanwhile, dma_addr
holds the physical address of the buffer. This field is
specified only when the buffer is a linear buffer.
dma_bytes holds the size of buffer
in bytes. dma_private is used for
the ALSA DMA allocator.
If you use a standard ALSA function,
snd_pcm_lib_malloc_pages(), for
allocating the buffer, these fields are set by the ALSA middle
layer, and you should not change them by
yourself. You can read them but not write them.
On the other hand, if you want to allocate the buffer by
yourself, you'll need to manage it in hw_params callback.
At least, dma_bytes is mandatory.
dma_area is necessary when the
buffer is mmapped. If your driver doesn't support mmap, this
field is not necessary. dma_addr
is also optional. You can use
dma_private as you like, too.
The running status can be referred via runtime->status.
This is the pointer to the struct snd_pcm_mmap_status
record. For example, you can get the current DMA hardware
pointer via runtime->status->hw_ptr.
The DMA application pointer can be referred via
runtime->control, which points to the
struct snd_pcm_mmap_control record.
However, accessing directly to this value is not recommended.
You can allocate a record for the substream and store it in
runtime->private_data. Usually, this
is done in
the open callback.
Don't mix this with pcm->private_data.
The pcm->private_data usually points to the
chip instance assigned statically at the creation of PCM, while the
runtime->private_data points to a dynamic
data structure created at the PCM open callback.
static int snd_xxx_open(struct snd_pcm_substream *substream)
{
struct my_pcm_data *data;
....
data = kmalloc(sizeof(*data), GFP_KERNEL);
substream->runtime->private_data = data;
....
}
The allocated object must be released in the close callback.
OK, now let me give details about each pcm callback
(ops). In general, every callback must
return 0 if successful, or a negative error number
such as -EINVAL. To choose an appropriate
error number, it is advised to check what value other parts of
the kernel return when the same kind of request fails.
The callback function takes at least the argument with snd_pcm_substream pointer. To retrieve the chip record from the given substream instance, you can use the following macro.
int xxx() {
struct mychip *chip = snd_pcm_substream_chip(substream);
....
}
The macro reads substream->private_data,
which is a copy of pcm->private_data.
You can override the former if you need to assign different data
records per PCM substream. For example, the cmi8330 driver assigns
different private_data for playback and capture directions,
because it uses two different codecs (SB- and AD-compatible) for
different directions.
static int snd_xxx_open(struct snd_pcm_substream *substream);
This is called when a pcm substream is opened.
At least, here you have to initialize the runtime->hw record. Typically, this is done by like this:
static int snd_xxx_open(struct snd_pcm_substream *substream)
{
struct mychip *chip = snd_pcm_substream_chip(substream);
struct snd_pcm_runtime *runtime = substream->runtime;
runtime->hw = snd_mychip_playback_hw;
return 0;
}
where snd_mychip_playback_hw is the
pre-defined hardware description.
You can allocate a private data in this callback, as described in Private Data section.
If the hardware configuration needs more constraints, set the hardware constraints here, too. See Constraints for more details.
static int snd_xxx_close(struct snd_pcm_substream *substream);
Obviously, this is called when a pcm substream is closed.
Any private instance for a pcm substream allocated in the open callback will be released here.
static int snd_xxx_close(struct snd_pcm_substream *substream)
{
....
kfree(substream->runtime->private_data);
....
}
This is used for any special call to pcm ioctls. But
usually you can pass a generic ioctl callback,
snd_pcm_lib_ioctl.
static int snd_xxx_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *hw_params);
This is called when the hardware parameter
(hw_params) is set
up by the application,
that is, once when the buffer size, the period size, the
format, etc. are defined for the pcm substream.
Many hardware setups should be done in this callback, including the allocation of buffers.
Parameters to be initialized are retrieved by
params_xxx() macros. To allocate
buffer, you can call a helper function,
snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
snd_pcm_lib_malloc_pages() is available
only when the DMA buffers have been pre-allocated.
See the section
Buffer Types for more details.
Note that this and prepare callbacks
may be called multiple times per initialization.
For example, the OSS emulation may
call these callbacks at each change via its ioctl.
Thus, you need to be careful not to allocate the same buffers many times, which will lead to memory leaks! Calling the helper function above many times is OK. It will release the previous buffer automatically when it was already allocated.
Another note is that this callback is non-atomic
(schedulable). This is important, because the
trigger callback
is atomic (non-schedulable). That is, mutexes or any
schedule-related functions are not available in
trigger callback.
Please see the subsection
Atomicity for details.
static int snd_xxx_hw_free(struct snd_pcm_substream *substream);
This is called to release the resources allocated via
hw_params. For example, releasing the
buffer via
snd_pcm_lib_malloc_pages() is done by
calling the following:
snd_pcm_lib_free_pages(substream);
This function is always called before the close callback is called. Also, the callback may be called multiple times, too. Keep track whether the resource was already released.
static int snd_xxx_prepare(struct snd_pcm_substream *substream);
This callback is called when the pcm is
“prepared”. You can set the format type, sample
rate, etc. here. The difference from
hw_params is that the
prepare callback will be called each
time
snd_pcm_prepare() is called, i.e. when
recovering after underruns, etc.
Note that this callback is now non-atomic. You can use schedule-related functions safely in this callback.
In this and the following callbacks, you can refer to the values via the runtime record, substream->runtime. For example, to get the current rate, format or channels, access to runtime->rate, runtime->format or runtime->channels, respectively. The physical address of the allocated buffer is set to runtime->dma_area. The buffer and period sizes are in runtime->buffer_size and runtime->period_size, respectively.
Be careful that this callback will be called many times at each setup, too.
static int snd_xxx_trigger(struct snd_pcm_substream *substream, int cmd);
This is called when the pcm is started, stopped or paused.
Which action is specified in the second argument,
SNDRV_PCM_TRIGGER_XXX in
<sound/pcm.h>. At least,
the START and STOP
commands must be defined in this callback.
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
/* do something to start the PCM engine */
break;
case SNDRV_PCM_TRIGGER_STOP:
/* do something to stop the PCM engine */
break;
default:
return -EINVAL;
}
When the pcm supports the pause operation (given in the info
field of the hardware table), the PAUSE_PUSE
and PAUSE_RELEASE commands must be
handled here, too. The former is the command to pause the pcm,
and the latter to restart the pcm again.
When the pcm supports the suspend/resume operation,
regardless of full or partial suspend/resume support,
the SUSPEND and RESUME
commands must be handled, too.
These commands are issued when the power-management status is
changed. Obviously, the SUSPEND and
RESUME commands
suspend and resume the pcm substream, and usually, they
are identical to the STOP and
START commands, respectively.
See the
Power Management section for details.
As mentioned, this callback is atomic. You cannot call functions which may sleep. The trigger callback should be as minimal as possible, just really triggering the DMA. The other stuff should be initialized hw_params and prepare callbacks properly beforehand.
static snd_pcm_uframes_t snd_xxx_pointer(struct snd_pcm_substream *substream)
This callback is called when the PCM middle layer inquires the current hardware position on the buffer. The position must be returned in frames, ranging from 0 to buffer_size - 1.
This is called usually from the buffer-update routine in the
pcm middle layer, which is invoked when
snd_pcm_period_elapsed() is called in the
interrupt routine. Then the pcm middle layer updates the
position and calculates the available space, and wakes up the
sleeping poll threads, etc.
This callback is also atomic.
These callbacks are not mandatory, and can be omitted in most cases. These callbacks are used when the hardware buffer cannot be in the normal memory space. Some chips have their own buffer on the hardware which is not mappable. In such a case, you have to transfer the data manually from the memory buffer to the hardware buffer. Or, if the buffer is non-contiguous on both physical and virtual memory spaces, these callbacks must be defined, too.
If these two callbacks are defined, copy and set-silence operations are done by them. The detailed will be described in the later section Buffer and Memory Management.
This callback is also not mandatory. This callback is called when the appl_ptr is updated in read or write operations. Some drivers like emu10k1-fx and cs46xx need to track the current appl_ptr for the internal buffer, and this callback is useful only for such a purpose.
This callback is atomic.
This callback is optional too. This callback is used mainly for non-contiguous buffers. The mmap calls this callback to get the page address. Some examples will be explained in the later section Buffer and Memory Management, too.
The rest of pcm stuff is the PCM interrupt handler. The
role of PCM interrupt handler in the sound driver is to update
the buffer position and to tell the PCM middle layer when the
buffer position goes across the prescribed period size. To
inform this, call the snd_pcm_period_elapsed()
function.
There are several types of sound chips to generate the interrupts.
This is the most frequently found type: the hardware
generates an interrupt at each period boundary.
In this case, you can call
snd_pcm_period_elapsed() at each
interrupt.
snd_pcm_period_elapsed() takes the
substream pointer as its argument. Thus, you need to keep the
substream pointer accessible from the chip instance. For
example, define substream field in the chip record to hold the
current running substream pointer, and set the pointer value
at open callback (and reset at close callback).
If you acquire a spinlock in the interrupt handler, and the
lock is used in other pcm callbacks, too, then you have to
release the lock before calling
snd_pcm_period_elapsed(), because
snd_pcm_period_elapsed() calls other pcm
callbacks inside.
Typical code would be like:
Example 5.3. Interrupt Handler Case #1
static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
{
struct mychip *chip = dev_id;
spin_lock(&chip->lock);
....
if (pcm_irq_invoked(chip)) {
/* call updater, unlock before it */
spin_unlock(&chip->lock);
snd_pcm_period_elapsed(chip->substream);
spin_lock(&chip->lock);
/* acknowledge the interrupt if necessary */
}
....
spin_unlock(&chip->lock);
return IRQ_HANDLED;
}
This happense when the hardware doesn't generate interrupts
at the period boundary but issues timer interrupts at a fixed
timer rate (e.g. es1968 or ymfpci drivers).
In this case, you need to check the current hardware
position and accumulate the processed sample length at each
interrupt. When the accumulated size exceeds the period
size, call
snd_pcm_period_elapsed() and reset the
accumulator.
Typical code would be like the following.
Example 5.4. Interrupt Handler Case #2
static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id)
{
struct mychip *chip = dev_id;
spin_lock(&chip->lock);
....
if (pcm_irq_invoked(chip)) {
unsigned int last_ptr, size;
/* get the current hardware pointer (in frames) */
last_ptr = get_hw_ptr(chip);
/* calculate the processed frames since the
* last update
*/
if (last_ptr < chip->last_ptr)
size = runtime->buffer_size + last_ptr
- chip->last_ptr;
else
size = last_ptr - chip->last_ptr;
/* remember the last updated point */
chip->last_ptr = last_ptr;
/* accumulate the size */
chip->size += size;
/* over the period boundary? */
if (chip->size >= runtime->period_size) {
/* reset the accumulator */
chip->size %= runtime->period_size;
/* call updater */
spin_unlock(&chip->lock);
snd_pcm_period_elapsed(substream);
spin_lock(&chip->lock);
}
/* acknowledge the interrupt if necessary */
}
....
spin_unlock(&chip->lock);
return IRQ_HANDLED;
}
One of the most important (and thus difficult to debug) problems in kernel programming are race conditions. In the Linux kernel, they are usually avoided via spin-locks, mutexes or semaphores. In general, if a race condition can happen in an interrupt handler, it has to be managed atomically, and you have to use a spinlock to protect the critical session. If the critical section is not in interrupt handler code and if taking a relatively long time to execute is acceptable, you should use mutexes or semaphores instead.
As already seen, some pcm callbacks are atomic and some are
not. For example, the hw_params callback is
non-atomic, while trigger callback is
atomic. This means, the latter is called already in a spinlock
held by the PCM middle layer. Please take this atomicity into
account when you choose a locking scheme in the callbacks.
In the atomic callbacks, you cannot use functions which may call
schedule or go to
sleep. Semaphores and mutexes can sleep,
and hence they cannot be used inside the atomic callbacks
(e.g. trigger callback).
To implement some delay in such a callback, please use
udelay() or mdelay().
All three atomic callbacks (trigger, pointer, and ack) are called with local interrupts disabled.
If your chip supports unconventional sample rates, or only the limited samples, you need to set a constraint for the condition.
For example, in order to restrict the sample rates in the some
supported values, use
snd_pcm_hw_constraint_list().
You need to call this function in the open callback.
Example 5.5. Example of Hardware Constraints
static unsigned int rates[] =
{4000, 10000, 22050, 44100};
static struct snd_pcm_hw_constraint_list constraints_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
static int snd_mychip_pcm_open(struct snd_pcm_substream *substream)
{
int err;
....
err = snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_rates);
if (err < 0)
return err;
....
}
There are many different constraints.
Look at sound/pcm.h for a complete list.
You can even define your own constraint rules.
For example, let's suppose my_chip can manage a substream of 1 channel
if and only if the format is S16_LE, otherwise it supports any format
specified in the snd_pcm_hardware structure (or in any
other constraint_list). You can build a rule like this:
Example 5.6. Example of Hardware Constraints for Channels
static int hw_rule_format_by_channels(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *c = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_mask fmt;
snd_mask_any(&fmt); /* Init the struct */
if (c->min < 2) {
fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE;
return snd_mask_refine(f, &fmt);
}
return 0;
}
Then you need to call this function to add your rule:
snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT,
-1);
The rule function is called when an application sets the number of channels. But an application can set the format before the number of channels. Thus you also need to define the inverse rule:
Example 5.7. Example of Hardware Constraints for Channels
static int hw_rule_channels_by_format(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
struct snd_interval *c = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_interval ch;
snd_interval_any(&ch);
if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) {
ch.min = ch.max = 1;
ch.integer = 1;
return snd_interval_refine(c, &ch);
}
return 0;
}
...and in the open callback:
snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
-1);
I won't give more details here, rather I would like to say, “Luke, use the source.”
Table of Contents
The control interface is used widely for many switches, sliders, etc. which are accessed from user-space. Its most important use is the mixer interface. In other words, since ALSA 0.9.x, all the mixer stuff is implemented on the control kernel API.
ALSA has a well-defined AC97 control module. If your chip supports only the AC97 and nothing else, you can skip this section.
The control API is defined in
<sound/control.h>.
Include this file if you want to add your own controls.
To create a new control, you need to define the
following three
callbacks: info,
get and
put. Then, define a
struct snd_kcontrol_new record, such as:
Example 6.1. Definition of a Control
static struct snd_kcontrol_new my_control __devinitdata = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "PCM Playback Switch",
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.private_value = 0xffff,
.info = my_control_info,
.get = my_control_get,
.put = my_control_put
};
Most likely the control is created via
snd_ctl_new1(), and in such a case, you can
add the __devinitdata prefix to the
definition as above.
The iface field specifies the control
type, SNDRV_CTL_ELEM_IFACE_XXX, which
is usually MIXER.
Use CARD for global controls that are not
logically part of the mixer.
If the control is closely associated with some specific device on
the sound card, use HWDEP,
PCM, RAWMIDI,
TIMER, or SEQUENCER, and
specify the device number with the
device and
subdevice fields.
The name is the name identifier
string. Since ALSA 0.9.x, the control name is very important,
because its role is classified from its name. There are
pre-defined standard control names. The details are described in
the
Control Names subsection.
The index field holds the index number
of this control. If there are several different controls with
the same name, they can be distinguished by the index
number. This is the case when
several codecs exist on the card. If the index is zero, you can
omit the definition above.
The access field contains the access
type of this control. Give the combination of bit masks,
SNDRV_CTL_ELEM_ACCESS_XXX, there.
The details will be explained in
the
Access Flags subsection.
The private_value field contains
an arbitrary long integer value for this record. When using
the generic info,
get and
put callbacks, you can pass a value
through this field. If several small numbers are necessary, you can
combine them in bitwise. Or, it's possible to give a pointer
(casted to unsigned long) of some record to this field, too.
The tlv field can be used to provide
metadata about the control; see the
Metadata subsection.
The other three are callback functions.
There are some standards to define the control names. A control is usually defined from the three parts as “SOURCE DIRECTION FUNCTION”.
The first, SOURCE, specifies the source
of the control, and is a string such as “Master”,
“PCM”, “CD” and
“Line”. There are many pre-defined sources.
The second, DIRECTION, is one of the
following strings according to the direction of the control:
“Playback”, “Capture”, “Bypass
Playback” and “Bypass Capture”. Or, it can
be omitted, meaning both playback and capture directions.
The third, FUNCTION, is one of the
following strings according to the function of the control:
“Switch”, “Volume” and
“Route”.
The example of control names are, thus, “Master Capture Switch” or “PCM Playback Volume”.
There are some exceptions:
“Capture Source”, “Capture Switch” and “Capture Volume” are used for the global capture (input) source, switch and volume. Similarly, “Playback Switch” and “Playback Volume” are used for the global output gain switch and volume.
tone-control switch and volumes are specified like “Tone Control - XXX”, e.g. “Tone Control - Switch”, “Tone Control - Bass”, “Tone Control - Center”.
3D-control switches and volumes are specified like “3D Control - XXX”, e.g. “3D Control - Switch”, “3D Control - Center”, “3D Control - Space”.
The access flag is the bitmask which specifies the access type
of the given control. The default access type is
SNDRV_CTL_ELEM_ACCESS_READWRITE,
which means both read and write are allowed to this control.
When the access flag is omitted (i.e. = 0), it is
considered as READWRITE access as default.
When the control is read-only, pass
SNDRV_CTL_ELEM_ACCESS_READ instead.
In this case, you don't have to define
the put callback.
Similarly, when the control is write-only (although it's a rare
case), you can use the WRITE flag instead, and
you don't need the get callback.
If the control value changes frequently (e.g. the VU meter),
VOLATILE flag should be given. This means
that the control may be changed without
notification. Applications should poll such
a control constantly.
When the control is inactive, set
the INACTIVE flag, too.
There are LOCK and
OWNER flags to change the write
permissions.
The info callback is used to get
detailed information on this control. This must store the
values of the given struct snd_ctl_elem_info
object. For example, for a boolean control with a single
element:
Example 6.2. Example of info callback
static int snd_myctl_mono_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;
}
The type field specifies the type
of the control. There are BOOLEAN,
INTEGER, ENUMERATED,
BYTES, IEC958 and
INTEGER64. The
count field specifies the
number of elements in this control. For example, a stereo
volume would have count = 2. The
value field is a union, and
the values stored are depending on the type. The boolean and
integer types are identical.
The enumerated type is a bit different from others. You'll need to set the string for the currently given item index.
static int snd_myctl_enum_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[4] = {
"First", "Second", "Third", "Fourth"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 4;
if (uinfo->value.enumerated.item > 3)
uinfo->value.enumerated.item = 3;
strcpy(uinfo->value.enumerated.name,
texts[uinfo->value.enumerated.item]);
return 0;
}
Some common info callbacks are available for your convenience:
snd_ctl_boolean_mono_info() and
snd_ctl_boolean_stereo_info().
Obviously, the former is an info callback for a mono channel
boolean item, just like snd_myctl_mono_info
above, and the latter is for a stereo channel boolean item.
This callback is used to read the current value of the control and to return to user-space.
For example,
Example 6.3. Example of get callback
static int snd_myctl_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct mychip *chip = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] = get_some_value(chip);
return 0;
}
The value field depends on
the type of control as well as on the info callback. For example,
the sb driver uses this field to store the register offset,
the bit-shift and the bit-mask. The
private_value field is set as follows:
.private_value = reg | (shift << 16) | (mask << 24)
and is retrieved in callbacks like
static int snd_sbmixer_get_single(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int reg = kcontrol->private_value & 0xff;
int shift = (kcontrol->private_value >> 16) & 0xff;
int mask = (kcontrol->private_value >> 24) & 0xff;
....
}
In the get callback,
you have to fill all the elements if the
control has more than one elements,
i.e. count > 1.
In the example above, we filled only one element
(value.integer.value[0]) since it's
assumed as count = 1.
This callback is used to write a value from user-space.
For example,
Example 6.4. Example of put callback
static int snd_myctl_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct mychip *chip = snd_kcontrol_chip(kcontrol);
int changed = 0;
if (chip->current_value !=
ucontrol->value.integer.value[0]) {
change_current_value(chip,
ucontrol->value.integer.value[0]);
changed = 1;
}
return changed;
}
As seen above, you have to return 1 if the value is
changed. If the value is not changed, return 0 instead.
If any fatal error happens, return a negative error code as
usual.
As in the get callback,
when the control has more than one elements,
all elements must be evaluated in this callback, too.
When everything is ready, finally we can create a new
control. To create a control, there are two functions to be
called, snd_ctl_new1() and
snd_ctl_add().
In the simplest way, you can do like this:
err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip));
if (err < 0)
return err;
where my_control is the
struct snd_kcontrol_new object defined above, and chip
is the object pointer to be passed to
kcontrol->private_data
which can be referred to in callbacks.
snd_ctl_new1() allocates a new
snd_kcontrol instance (that's why the definition
of my_control can be with
the __devinitdata
prefix), and snd_ctl_add assigns the given
control component to the card.
If you need to change and update a control in the interrupt
routine, you can call snd_ctl_notify(). For
example,
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer);
This function takes the card pointer, the event-mask, and the
control id pointer for the notification. The event-mask
specifies the types of notification, for example, in the above
example, the change of control values is notified.
The id pointer is the pointer of struct snd_ctl_elem_id
to be notified.
You can find some examples in es1938.c or
es1968.c for hardware volume interrupts.
To provide information about the dB values of a mixer control, use
on of the DECLARE_TLV_xxx macros from
<sound/tlv.h> to define a variable
containing this information, set thetlv.p
field to point to this variable, and include the
SNDRV_CTL_ELEM_ACCESS_TLV_READ flag in the
access field; like this:
static DECLARE_TLV_DB_SCALE(db_scale_my_control, -4050, 150, 0);
static struct snd_kcontrol_new my_control __devinitdata = {
...
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
SNDRV_CTL_ELEM_ACCESS_TLV_READ,
...
.tlv.p = db_scale_my_control,
};
The DECLARE_TLV_DB_SCALE macro defines
information about a mixer control where each step in the control's
value changes the dB value by a constant dB amount.
The first parameter is the name of the variable to be defined.
The second parameter is the minimum value, in units of 0.01 dB.
The third parameter is the step size, in units of 0.01 dB.
Set the fourth parameter to 1 if the minimum value actually mutes
the control.
The DECLARE_TLV_DB_LINEAR macro defines
information about a mixer control where the control's value affects
the output linearly.
The first parameter is the name of the variable to be defined.
The second parameter is the minimum value, in units of 0.01 dB.
The third parameter is the maximum value, in units of 0.01 dB.
If the minimum value mutes the control, set the second parameter to
TLV_DB_GAIN_MUTE.
Table of Contents
The ALSA AC97 codec layer is a well-defined one, and you don't
have to write much code to control it. Only low-level control
routines are necessary. The AC97 codec API is defined in
<sound/ac97_codec.h>.
Example 7.1. Example of AC97 Interface
struct mychip {
....
struct snd_ac97 *ac97;
....
};
static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97,
unsigned short reg)
{
struct mychip *chip = ac97->private_data;
....
/* read a register value here from the codec */
return the_register_value;
}
static void snd_mychip_ac97_write(struct snd_ac97 *ac97,
unsigned short reg, unsigned short val)
{
struct mychip *chip = ac97->private_data;
....
/* write the given register value to the codec */
}
static int snd_mychip_ac97(struct mychip *chip)
{
struct snd_ac97_bus *bus;
struct snd_ac97_template ac97;
int err;
static struct snd_ac97_bus_ops ops = {
.write = snd_mychip_ac97_write,
.read = snd_mychip_ac97_read,
};
err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus);
if (err < 0)
return err;
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = chip;
return snd_ac97_mixer(bus, &ac97, &chip->ac97);
}
To create an ac97 instance, first call snd_ac97_bus
with an ac97_bus_ops_t record with callback functions.
struct snd_ac97_bus *bus;
static struct snd_ac97_bus_ops ops = {
.write = snd_mychip_ac97_write,
.read = snd_mychip_ac97_read,
};
snd_ac97_bus(card, 0, &ops, NULL, &pbus);
The bus record is shared among all belonging ac97 instances.
And then call snd_ac97_mixer() with an
struct snd_ac97_template
record together with the bus pointer created above.
struct snd_ac97_template ac97;
int err;
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = chip;
snd_ac97_mixer(bus, &ac97, &chip->ac97);
where chip->ac97 is a pointer to a newly created ac97_t instance. In this case, the chip pointer is set as the private data, so that the read/write callback functions can refer to this chip instance. This instance is not necessarily stored in the chip record. If you need to change the register values from the driver, or need the suspend/resume of ac97 codecs, keep this pointer to pass to the corresponding functions.
The standard callbacks are read and
write. Obviously they
correspond to the functions for read and write accesses to the
hardware low-level codes.
The read callback returns the
register value specified in the argument.
static unsigned short snd_mychip_ac97_read(struct snd_ac97 *ac97,
unsigned short reg)
{
struct mychip *chip = ac97->private_data;
....
return the_register_value;
}
Here, the chip can be cast from ac97->private_data.
Meanwhile, the write callback is
used to set the register value.
static void snd_mychip_ac97_write(struct snd_ac97 *ac97,
unsigned short reg, unsigned short val)
These callbacks are non-atomic like the control API callbacks.
There are also other callbacks:
reset,
wait and
init.
The reset callback is used to reset
the codec. If the chip requires a special kind of reset, you can
define this callback.
The wait callback is used to
add some waiting time in the standard initialization of the codec. If the
chip requires the extra waiting time, define this callback.
The init callback is used for
additional initialization of the codec.
If you need to access to the codec from the driver, you can
call the following functions:
snd_ac97_write(),
snd_ac97_read(),
snd_ac97_update() and
snd_ac97_update_bits().
Both snd_ac97_write() and
snd_ac97_update() functions are used to
set a value to the given register
(AC97_XXX). The difference between them is
that snd_ac97_update() doesn't write a
value if the given value has been already set, while
snd_ac97_write() always rewrites the
value.
snd_ac97_write(ac97, AC97_MASTER, 0x8080);
snd_ac97_update(ac97, AC97_MASTER, 0x8080);
snd_ac97_read() is used to read the value
of the given register. For example,
value = snd_ac97_read(ac97, AC97_MASTER);
snd_ac97_update_bits() is used to update
some bits in the given register.
snd_ac97_update_bits(ac97, reg, mask, value);
Also, there is a function to change the sample rate (of a
given register such as
AC97_PCM_FRONT_DAC_RATE) when VRA or
DRA is supported by the codec:
snd_ac97_set_rate().
snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100);
The following registers are available to set the rate:
AC97_PCM_MIC_ADC_RATE,
AC97_PCM_FRONT_DAC_RATE,
AC97_PCM_LR_ADC_RATE,
AC97_SPDIF. When
AC97_SPDIF is specified, the register is
not really changed but the corresponding IEC958 status bits will
be updated.
In some chips, the clock of the codec isn't 48000 but using a PCI clock (to save a quartz!). In this case, change the field bus->clock to the corresponding value. For example, intel8x0 and es1968 drivers have their own function to read from the clock.
The ALSA AC97 interface will create a proc file such as
/proc/asound/card0/codec97#0/ac97#0-0 and
ac97#0-0+regs. You can refer to these files to
see the current status and registers of the codec.
When there are several codecs on the same card, you need to
call snd_ac97_mixer() multiple times with
ac97.num=1 or greater. The num field
specifies the codec number.
If you set up multiple codecs, you either need to write different callbacks for each codec or check ac97->num in the callback routines.
Table of Contents
Many soundcards have built-in MIDI (MPU401-UART)
interfaces. When the soundcard supports the standard MPU401-UART
interface, most likely you can use the ALSA MPU401-UART API. The
MPU401-UART API is defined in
<sound/mpu401.h>.
Some soundchips have a similar but slightly different implementation of mpu401 stuff. For example, emu10k1 has its own mpu401 routines.
To create a rawmidi object, call
snd_mpu401_uart_new().
struct snd_rawmidi *rmidi;
snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, info_flags,
irq, irq_flags, &rmidi);
The first argument is the card pointer, and the second is the index of this component. You can create up to 8 rawmidi devices.
The third argument is the type of the hardware,
MPU401_HW_XXX. If it's not a special one,
you can use MPU401_HW_MPU401.
The 4th argument is the I/O port address. Many backward-compatible MPU401 have an I/O port such as 0x330. Or, it might be a part of its own PCI I/O region. It depends on the chip design.
The 5th argument is a bitflag for additional information.
When the I/O port address above is part of the PCI I/O
region, the MPU401 I/O port might have been already allocated
(reserved) by the driver itself. In such a case, pass a bit flag
MPU401_INFO_INTEGRATED,
and the mpu401-uart layer will allocate the I/O ports by itself.
When the controller supports only the input or output MIDI stream,
pass the MPU401_INFO_INPUT or
MPU401_INFO_OUTPUT bitflag, respectively.
Then the rawmidi instance is created as a single stream.
MPU401_INFO_MMIO bitflag is used to change
the access method to MMIO (via readb and writeb) instead of
iob and outb. In this case, you have to pass the iomapped address
to snd_mpu401_uart_new().
When MPU401_INFO_TX_IRQ is set, the output
stream isn't checked in the default interrupt handler. The driver
needs to call snd_mpu401_uart_interrupt_tx()
by itself to start processing the output stream in the irq handler.
Usually, the port address corresponds to the command port and
port + 1 corresponds to the data port. If not, you may change
the cport field of
struct snd_mpu401 manually
afterward. However, snd_mpu401 pointer is not
returned explicitly by
snd_mpu401_uart_new(). You need to cast
rmidi->private_data to
snd_mpu401 explicitly,
struct snd_mpu401 *mpu;
mpu = rmidi->private_data;
and reset the cport as you like:
mpu->cport = my_own_control_port;
The 6th argument specifies the irq number for UART. If the irq
is already allocated, pass 0 to the 7th argument
(irq_flags). Otherwise, pass the flags
for irq allocation
(SA_XXX bits) to it, and the irq will be
reserved by the mpu401-uart layer. If the card doesn't generate
UART interrupts, pass -1 as the irq number. Then a timer
interrupt will be invoked for polling.
When the interrupt is allocated in
snd_mpu401_uart_new(), the private
interrupt handler is used, hence you don't have anything else to do
than creating the mpu401 stuff. Otherwise, you have to call
snd_mpu401_uart_interrupt() explicitly when
a UART interrupt is invoked and checked in your own interrupt
handler.
In this case, you need to pass the private_data of the
returned rawmidi object from
snd_mpu401_uart_new() as the second
argument of snd_mpu401_uart_interrupt().
snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs);
Table of Contents
The raw MIDI interface is used for hardware MIDI ports that can be accessed as a byte stream. It is not used for synthesizer chips that do not directly understand MIDI.
ALSA handles file and buffer management. All you have to do is to write some code to move data between the buffer and the hardware.
The rawmidi API is defined in
<sound/rawmidi.h>.
To create a rawmidi device, call the
snd_rawmidi_new function:
struct snd_rawmidi *rmidi;
err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi);
if (err < 0)
return err;
rmidi->private_data = chip;
strcpy(rmidi->name, "My MIDI");
rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
SNDRV_RAWMIDI_INFO_INPUT |
SNDRV_RAWMIDI_INFO_DUPLEX;
The first argument is the card pointer, the second argument is the ID string.
The third argument is the index of this component. You can create up to 8 rawmidi devices.
The fourth and fifth arguments are the number of output and input substreams, respectively, of this device (a substream is the equivalent of a MIDI port).
Set the info_flags field to specify
the capabilities of the device.
Set SNDRV_RAWMIDI_INFO_OUTPUT if there is
at least one output port,
SNDRV_RAWMIDI_INFO_INPUT if there is at
least one input port,
and SNDRV_RAWMIDI_INFO_DUPLEX if the device
can handle output and input at the same time.
After the rawmidi device is created, you need to set the operators (callbacks) for each substream. There are helper functions to set the operators for all the substreams of a device:
snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops);
snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops);
The operators are usually defined like this:
static struct snd_rawmidi_ops snd_mymidi_output_ops = {
.open = snd_mymidi_output_open,
.close = snd_mymidi_output_close,
.trigger = snd_mymidi_output_trigger,
};
These callbacks are explained in the Callbacks section.
If there are more than one substream, you should give a unique name to each of them:
struct snd_rawmidi_substream *substream;
list_for_each_entry(substream,
&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams,
list {
sprintf(substream->name, "My MIDI Port %d", substream->number + 1);
}
/* same for SNDRV_RAWMIDI_STREAM_INPUT */
In all the callbacks, the private data that you've set for the rawmidi device can be accessed as substream->rmidi->private_data.
If there is more than one port, your callbacks can determine the port index from the struct snd_rawmidi_substream data passed to each callback:
struct snd_rawmidi_substream *substream;
int index = substream->number;
static int snd_xxx_open(struct snd_rawmidi_substream *substream);
This is called when a substream is opened. You can initialize the hardware here, but you shouldn't start transmitting/receiving data yet.
static int snd_xxx_close(struct snd_rawmidi_substream *substream);
Guess what.
The open and close
callbacks of a rawmidi device are serialized with a mutex,
and can sleep.
static void snd_xxx_output_trigger(struct snd_rawmidi_substream *substream, int up);
This is called with a nonzero up
parameter when there is some data in the substream buffer that
must be transmitted.
To read data from the buffer, call
snd_rawmidi_transmit_peek. It will
return the number of bytes that have been read; this will be
less than the number of bytes requested when there are no more
data in the buffer.
After the data have been transmitted successfully, call
snd_rawmidi_transmit_ack to remove the
data from the substream buffer:
unsigned char data;
while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) {
if (snd_mychip_try_to_transmit(data))
snd_rawmidi_transmit_ack(substream, 1);
else
break; /* hardware FIFO full */
}
If you know beforehand that the hardware will accept data, you
can use the snd_rawmidi_transmit function
which reads some data and removes them from the buffer at once:
while (snd_mychip_transmit_possible()) {
unsigned char data;
if (snd_rawmidi_transmit(substream, &data, 1) != 1)
break; /* no more data */
snd_mychip_transmit(data);
}
If you know beforehand how many bytes you can accept, you can
use a buffer size greater than one with the
snd_rawmidi_transmit* functions.
The trigger callback must not sleep. If
the hardware FIFO is full before the substream buffer has been
emptied, you have to continue transmitting data later, either
in an interrupt handler, or with a timer if the hardware
doesn't have a MIDI transmit interrupt.
The trigger callback is called with a
zero up parameter when the transmission
of data should be aborted.
static void snd_xxx_input_trigger(struct snd_rawmidi_substream *substream, int up);
This is called with a nonzero up
parameter to enable receiving data, or with a zero
up parameter do disable receiving data.
The trigger callback must not sleep; the
actual reading of data from the device is usually done in an
interrupt handler.
When data reception is enabled, your interrupt handler should
call snd_rawmidi_receive for all received
data:
void snd_mychip_midi_interrupt(...)
{
while (mychip_midi_available()) {
unsigned char data;
data = mychip_midi_read();
snd_rawmidi_receive(substream, &data, 1);
}
}
static void snd_xxx_drain(struct snd_rawmidi_substream *substream);
This is only used with output substreams. This function should wait until all data read from the substream buffer have been transmitted. This ensures that the device can be closed and the driver unloaded without losing data.
This callback is optional. If you do not set
drain in the struct snd_rawmidi_ops
structure, ALSA will simply wait for 50 milliseconds
instead.
Table of Contents
The FM OPL3 is still used in many chips (mainly for backward
compatibility). ALSA has a nice OPL3 FM control layer, too. The
OPL3 API is defined in
<sound/opl3.h>.
FM registers can be directly accessed through the direct-FM API,
defined in <sound/asound_fm.h>. In
ALSA native mode, FM registers are accessed through
the Hardware-Dependant Device direct-FM extension API, whereas in
OSS compatible mode, FM registers can be accessed with the OSS
direct-FM compatible API in /dev/dmfmX device.
To create the OPL3 component, you have two functions to call. The first one is a constructor for the opl3_t instance.
struct snd_opl3 *opl3;
snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX,
integrated, &opl3);
The first argument is the card pointer, the second one is the left port address, and the third is the right port address. In most cases, the right port is placed at the left port + 2.
The fourth argument is the hardware type.
When the left and right ports have been already allocated by
the card driver, pass non-zero to the fifth argument
(integrated). Otherwise, the opl3 module will
allocate the specified ports by itself.
When the accessing the hardware requires special method
instead of the standard I/O access, you can create opl3 instance
separately with snd_opl3_new().
struct snd_opl3 *opl3;
snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3);
Then set command,
private_data and
private_free for the private
access function, the private data and the destructor.
The l_port and r_port are not necessarily set. Only the
command must be set properly. You can retrieve the data
from the opl3->private_data field.
After creating the opl3 instance via snd_opl3_new(),
call snd_opl3_init() to initialize the chip to the
proper state. Note that snd_opl3_create() always
calls it internally.
If the opl3 instance is created successfully, then create a hwdep device for this opl3.
struct snd_hwdep *opl3hwdep;
snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep);
The first argument is the opl3_t instance you created, and the second is the index number, usually 0.
The third argument is the index-offset for the sequencer client assigned to the OPL3 port. When there is an MPU401-UART, give 1 for here (UART always takes 0).
Some chips need user-space access for special
controls or for loading the micro code. In such a case, you can
create a hwdep (hardware-dependent) device. The hwdep API is
defined in <sound/hwdep.h>. You can
find examples in opl3 driver or
isa/sb/sb16_csp.c.
The creation of the hwdep instance is done via
snd_hwdep_new().
struct snd_hwdep *hw;
snd_hwdep_new(card, "My HWDEP", 0, &hw);
where the third argument is the index number.
You can then pass any pointer value to the
private_data.
If you assign a private data, you should define the
destructor, too. The destructor function is set in
the private_free field.
struct mydata *p = kmalloc(sizeof(*p), GFP_KERNEL);
hw->private_data = p;
hw->private_free = mydata_free;
and the implementation of the destructor would be:
static void mydata_free(struct snd_hwdep *hw)
{
struct mydata *p = hw->private_data;
kfree(p);
}
The arbitrary file operations can be defined for this
instance. The file operators are defined in
the ops table. For example, assume that
this chip needs an ioctl.
hw->ops.open = mydata_open;
hw->ops.ioctl = mydata_ioctl;
hw->ops.release = mydata_release;
And implement the callback functions as you like.
Usually the controls for IEC958 devices are implemented via
the control interface. There is a macro to compose a name string for
IEC958 controls, SNDRV_CTL_NAME_IEC958()
defined in <include/asound.h>.
There are some standard controls for IEC958 status bits. These
controls use the type SNDRV_CTL_ELEM_TYPE_IEC958,
and the size of element is fixed as 4 bytes array
(value.iec958.status[x]). For the info
callback, you don't specify
the value field for this type (the count field must be set,
though).
“IEC958 Playback Con Mask” is used to return the
bit-mask for the IEC958 status bits of consumer mode. Similarly,
“IEC958 Playback Pro Mask” returns the bitmask for
professional mode. They are read-only controls, and are defined
as MIXER controls (iface =
SNDRV_CTL_ELEM_IFACE_MIXER).
Meanwhile, “IEC958 Playback Default” control is
defined for getting and setting the current default IEC958
bits. Note that this one is usually defined as a PCM control
(iface = SNDRV_CTL_ELEM_IFACE_PCM),
although in some places it's defined as a MIXER control.
In addition, you can define the control switches to
enable/disable or to set the raw bit mode. The implementation
will depend on the chip, but the control should be named as
“IEC958 xxx”, preferably using
the SNDRV_CTL_NAME_IEC958() macro.
You can find several cases, for example,
pci/emu10k1,
pci/ice1712, or
pci/cmipci.c.
ALSA provides several different buffer allocation functions
depending on the bus and the architecture. All these have a
consistent API. The allocation of physically-contiguous pages is
done via
snd_malloc_xxx_pages() function, where xxx
is the bus type.
The allocation of pages with fallback is
snd_malloc_xxx_pages_fallback(). This
function tries to allocate the specified pages but if the pages
are not available, it tries to reduce the page sizes until
enough space is found.
The release the pages, call
snd_free_xxx_pages() function.
Usually, ALSA drivers try to allocate and reserve a large contiguous physical space at the time the module is loaded for the later use. This is called “pre-allocation”. As already written, you can call the following function at pcm instance construction time (in the case of PCI bus).
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(pci), size, max);
where size is the byte size to be
pre-allocated and the max is the maximum
size to be changed via the prealloc proc file.
The allocator will try to get an area as large as possible
within the given size.
The second argument (type) and the third argument (device pointer)
are dependent on the bus.
In the case of the ISA bus, pass snd_dma_isa_data()
as the third argument with SNDRV_DMA_TYPE_DEV type.
For the continuous buffer unrelated to the bus can be pre-allocated
with SNDRV_DMA_TYPE_CONTINUOUS type and the
snd_dma_continuous_data(GFP_KERNEL) device pointer,
where GFP_KERNEL is the kernel allocation flag to
use.
For the PCI scatter-gather buffers, use
SNDRV_DMA_TYPE_DEV_SG with
snd_dma_pci_data(pci)
(see the
Non-Contiguous Buffers
section).
Once the buffer is pre-allocated, you can use the
allocator in the hw_params callback:
snd_pcm_lib_malloc_pages(substream, size);
Note that you have to pre-allocate to use this function.
Some chips have their own hardware buffers and the DMA transfer from the host memory is not available. In such a case, you need to either 1) copy/set the audio data directly to the external hardware buffer, or 2) make an intermediate buffer and copy/set the data from it to the external hardware buffer in interrupts (or in tasklets, preferably).
The first case works fine if the external hardware buffer is large
enough. This method doesn't need any extra buffers and thus is
more effective. You need to define the
copy and
silence callbacks for
the data transfer. However, there is a drawback: it cannot
be mmapped. The examples are GUS's GF1 PCM or emu8000's
wavetable PCM.
The second case allows for mmap on the buffer, although you have to handle an interrupt or a tasklet to transfer the data from the intermediate buffer to the hardware buffer. You can find an example in the vxpocket driver.
Another case is when the chip uses a PCI memory-map
region for the buffer instead of the host memory. In this case,
mmap is available only on certain architectures like the Intel one.
In non-mmap mode, the data cannot be transferred as in the normal
way. Thus you need to define the copy and
silence callbacks as well,
as in the cases above. The examples are found in
rme32.c and rme96.c.
The implementation of the copy and
silence callbacks depends upon
whether the hardware supports interleaved or non-interleaved
samples. The copy callback is
defined like below, a bit
differently depending whether the direction is playback or
capture:
static int playback_copy(struct snd_pcm_substream *substream, int channel,
snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count);
static int capture_copy(struct snd_pcm_substream *substream, int channel,
snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count);
In the case of interleaved samples, the second argument
(channel) is not used. The third argument
(pos) points the
current position offset in frames.
The meaning of the fourth argument is different between playback and capture. For playback, it holds the source data pointer, and for capture, it's the destination data pointer.
The last argument is the number of frames to be copied.
What you have to do in this callback is again different
between playback and capture directions. In the
playback case, you copy the given amount of data
(count) at the specified pointer
(src) to the specified offset
(pos) on the hardware buffer. When
coded like memcpy-like way, the copy would be like:
my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src,
frames_to_bytes(runtime, count));
For the capture direction, you copy the given amount of
data (count) at the specified offset
(pos) on the hardware buffer to the
specified pointer (dst).
my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos),
frames_to_bytes(runtime, count));
Note that both the position and the amount of data are given in frames.
In the case of non-interleaved samples, the implementation will be a bit more complicated.
You need to check the channel argument, and if it's -1, copy
the whole channels. Otherwise, you have to copy only the
specified channel. Please check
isa/gus/gus_pcm.c as an example.
The silence callback is also
implemented in a similar way.
static int silence(struct snd_pcm_substream *substream, int channel,
snd_pcm_uframes_t pos, snd_pcm_uframes_t count);
The meanings of arguments are the same as in the
copy
callback, although there is no src/dst
argument. In the case of interleaved samples, the channel
argument has no meaning, as well as on
copy callback.
The role of silence callback is to
set the given amount
(count) of silence data at the
specified offset (pos) on the hardware
buffer. Suppose that the data format is signed (that is, the
silent-data is 0), and the implementation using a memset-like
function would be like:
my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0,
frames_to_bytes(runtime, count));
In the case of non-interleaved samples, again, the
implementation becomes a bit more complicated. See, for example,
isa/gus/gus_pcm.c.
If your hardware supports the page table as in emu10k1 or the
buffer descriptors as in via82xx, you can use the scatter-gather
(SG) DMA. ALSA provides an interface for handling SG-buffers.
The API is provided in <sound/pcm.h>.
For creating the SG-buffer handler, call
snd_pcm_lib_preallocate_pages() or
snd_pcm_lib_preallocate_pages_for_all()
with SNDRV_DMA_TYPE_DEV_SG
in the PCM constructor like other PCI pre-allocator.
You need to pass snd_dma_pci_data(pci),
where pci is the struct pci_dev pointer
of the chip as well.
The struct snd_sg_buf instance is created as
substream->dma_private. You can cast
the pointer like:
struct snd_sg_buf *sgbuf = (struct snd_sg_buf *)substream->dma_private;
Then call snd_pcm_lib_malloc_pages()
in the hw_params callback
as well as in the case of normal PCI buffer.
The SG-buffer handler will allocate the non-contiguous kernel
pages of the given size and map them onto the virtually contiguous
memory. The virtual pointer is addressed in runtime->dma_area.
The physical address (runtime->dma_addr) is set to zero,
because the buffer is physically non-contigous.
The physical address table is set up in sgbuf->table.
You can get the physical address at a certain offset via
snd_pcm_sgbuf_get_addr().
When a SG-handler is used, you need to set
snd_pcm_sgbuf_ops_page as
the page callback.
(See
page callback section.)
To release the data, call
snd_pcm_lib_free_pages() in the
hw_free callback as usual.
It's possible to use a buffer allocated via
vmalloc, for example, for an intermediate
buffer. Since the allocated pages are not contiguous, you need
to set the page callback to obtain
the physical address at every offset.
The implementation of page callback
would be like this:
#include <linux/vmalloc.h>
/* get the physical page pointer on the given offset */
static struct page *mychip_page(struct snd_pcm_substream *substream,
unsigned long offset)
{
void *pageptr = substream->runtime->dma_area + offset;
return vmalloc_to_page(pageptr);
}
ALSA provides an easy interface for procfs. The proc files are
very useful for debugging. I recommend you set up proc files if
you write a driver and want to get a running status or register
dumps. The API is found in
<sound/info.h>.
To create a proc file, call
snd_card_proc_new().
struct snd_info_entry *entry;
int err = snd_card_proc_new(card, "my-file", &entry);
where the second argument specifies the name of the proc file to be
created. The above example will create a file
my-file under the card directory,
e.g. /proc/asound/card0/my-file.
Like other components, the proc entry created via
snd_card_proc_new() will be registered and
released automatically in the card registration and release
functions.
When the creation is successful, the function stores a new
instance in the pointer given in the third argument.
It is initialized as a text proc file for read only. To use
this proc file as a read-only text file as it is, set the read
callback with a private data via
snd_info_set_text_ops().
snd_info_set_text_ops(entry, chip, my_proc_read);
where the second argument (chip) is the
private data to be used in the callbacks. The third parameter
specifies the read buffer size and the fourth
(my_proc_read) is the callback function, which
is defined like
static void my_proc_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer);
In the read callback, use snd_iprintf() for
output strings, which works just like normal
printf(). For example,
static void my_proc_read(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct my_chip *chip = entry->private_data;
snd_iprintf(buffer, "This is my chip!\n");
snd_iprintf(buffer, "Port = %ld\n", chip->port);
}
The file permissions can be changed afterwards. As default, it's set as read only for all users. If you want to add write permission for the user (root as default), do as follows:
entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
and set the write buffer size and the callback
entry->c.text.write = my_proc_write;
For the write callback, you can use
snd_info_get_line() to get a text line, and
snd_info_get_str() to retrieve a string from
the line. Some examples are found in
core/oss/mixer_oss.c, core/oss/and
pcm_oss.c.
For a raw-data proc-file, set the attributes as follows:
static struct snd_info_entry_ops my_file_io_ops = {
.read = my_file_io_read,
};
entry->content = SNDRV_INFO_CONTENT_DATA;
entry->private_data = chip;
entry->c.ops = &my_file_io_ops;
entry->size = 4096;
entry->mode = S_IFREG | S_IRUGO;
The callback is much more complicated than the text-file
version. You need to use a low-level I/O functions such as
copy_from/to_user() to transfer the
data.
static long my_file_io_read(struct snd_info_entry *entry,
void *file_private_data,
struct file *file,
char *buf,
unsigned long count,
unsigned long pos)
{
long size = count;
if (pos + size > local_max_size)
size = local_max_size - pos;
if (copy_to_user(buf, local_data + pos, size))
return -EFAULT;
return size;
}
If the chip is supposed to work with suspend/resume
functions, you need to add power-management code to the
driver. The additional code for power-management should be
ifdef'ed with
CONFIG_PM.
If the driver fully supports suspend/resume
that is, the device can be
properly resumed to its state when suspend was called,
you can set the SNDRV_PCM_INFO_RESUME flag
in the pcm info field. Usually, this is possible when the
registers of the chip can be safely saved and restored to
RAM. If this is set, the trigger callback is called with
SNDRV_PCM_TRIGGER_RESUME after the resume
callback completes.
Even if the driver doesn't support PM fully but
partial suspend/resume is still possible, it's still worthy to
implement suspend/resume callbacks. In such a case, applications
would reset the status by calling
snd_pcm_prepare() and restart the stream
appropriately. Hence, you can define suspend/resume callbacks
below but don't set SNDRV_PCM_INFO_RESUME
info flag to the PCM.
Note that the trigger with SUSPEND can always be called when
snd_pcm_suspend_all is called,
regardless of the SNDRV_PCM_INFO_RESUME flag.
The RESUME flag affects only the behavior
of snd_pcm_resume().
(Thus, in theory,
SNDRV_PCM_TRIGGER_RESUME isn't needed
to be handled in the trigger callback when no
SNDRV_PCM_INFO_RESUME flag is set. But,
it's better to keep it for compatibility reasons.)
In the earlier version of ALSA drivers, a common power-management layer was provided, but it has been removed. The driver needs to define the suspend/resume hooks according to the bus the device is connected to. In the case of PCI drivers, the callbacks look like below:
#ifdef CONFIG_PM
static int snd_my_suspend(struct pci_dev *pci, pm_message_t state)
{
.... /* do things for suspend */
return 0;
}
static int snd_my_resume(struct pci_dev *pci)
{
.... /* do things for suspend */
return 0;
}
#endif
The scheme of the real suspend job is as follows.
Retrieve the card and the chip data.
Call snd_power_change_state() with
SNDRV_CTL_POWER_D3hot to change the
power status.
Call snd_pcm_suspend_all() to suspend the running PCM streams.
If AC97 codecs are used, call
snd_ac97_suspend() for each codec.
Save the register values if necessary.
Stop the hardware if necessary.
Disable the PCI device by calling
pci_disable_device(). Then, call
pci_save_state() at last.
A typical code would be like:
static int mychip_suspend(struct pci_dev *pci, pm_message_t state)
{
/* (1) */
struct snd_card *card = pci_get_drvdata(pci);
struct mychip *chip = card->private_data;
/* (2) */
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
/* (3) */
snd_pcm_suspend_all(chip->pcm);
/* (4) */
snd_ac97_suspend(chip->ac97);
/* (5) */
snd_mychip_save_registers(chip);
/* (6) */
snd_mychip_stop_hardware(chip);
/* (7) */
pci_disable_device(pci);
pci_save_state(pci);
return 0;
}
The scheme of the real resume job is as follows.
Retrieve the card and the chip data.
Set up PCI. First, call pci_restore_state().
Then enable the pci device again by calling pci_enable_device().
Call pci_set_master() if necessary, too.
Re-initialize the chip.
Restore the saved registers if necessary.
Resume the mixer, e.g. calling
snd_ac97_resume().
Restart the hardware (if any).
Call snd_power_change_state() with
SNDRV_CTL_POWER_D0 to notify the processes.
A typical code would be like:
static int mychip_resume(struct pci_dev *pci)
{
/* (1) */
struct snd_card *card = pci_get_drvdata(pci);
struct mychip *chip = card->private_data;
/* (2) */
pci_restore_state(pci);
pci_enable_device(pci);
pci_set_master(pci);
/* (3) */
snd_mychip_reinit_chip(chip);
/* (4) */
snd_mychip_restore_registers(chip);
/* (5) */
snd_ac97_resume(chip->ac97);
/* (6) */
snd_mychip_restart_chip(chip);
/* (7) */
snd_power_change_state(card, SNDRV_CTL_POWER_D0);
return 0;
}
As shown in the above, it's better to save registers after
suspending the PCM operations via
snd_pcm_suspend_all() or
snd_pcm_suspend(). It means that the PCM
streams are already stoppped when the register snapshot is
taken. But, remember that you don't have to restart the PCM
stream in the resume callback. It'll be restarted via
trigger call with SNDRV_PCM_TRIGGER_RESUME
when necessary.
OK, we have all callbacks now. Let's set them up. In the
initialization of the card, make sure that you can get the chip
data from the card instance, typically via
private_data field, in case you
created the chip data individually.
static int __devinit snd_mychip_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
....
struct snd_card *card;
struct mychip *chip;
int err;
....
err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
....
chip = kzalloc(sizeof(*chip), GFP_KERNEL);
....
card->private_data = chip;
....
}
When you created the chip data with
snd_card_create(), it's anyway accessible
via private_data field.
static int __devinit snd_mychip_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
....
struct snd_card *card;
struct mychip *chip;
int err;
....
err = snd_card_create(index[dev], id[dev], THIS_MODULE,
sizeof(struct mychip), &card);
....
chip = card->private_data;
....
}
If you need a space to save the registers, allocate the buffer for it here, too, since it would be fatal if you cannot allocate a memory in the suspend phase. The allocated buffer should be released in the corresponding destructor.
And next, set suspend/resume callbacks to the pci_driver.
static struct pci_driver driver = {
.name = "My Chip",
.id_table = snd_my_ids,
.probe = snd_my_probe,
.remove = __devexit_p(snd_my_remove),
#ifdef CONFIG_PM
.suspend = snd_my_suspend,
.resume = snd_my_resume,
#endif
};
There are standard module options for ALSA. At least, each
module should have the index,
id and enable
options.
If the module supports multiple cards (usually up to
8 = SNDRV_CARDS cards), they should be
arrays. The default initial values are defined already as
constants for easier programming:
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
If the module supports only a single card, they could be single
variables, instead. enable option is not
always necessary in this case, but it would be better to have a
dummy option for compatibility.
The module parameters must be declared with the standard
module_param()(),
module_param_array()() and
MODULE_PARM_DESC() macros.
The typical coding would be like below:
#define CARD_NAME "My Chip"
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
module_param_array(id, charp, NULL, 0444);
MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
Also, don't forget to define the module description, classes, license and devices. Especially, the recent modprobe requires to define the module license as GPL, etc., otherwise the system is shown as “tainted”.
MODULE_DESCRIPTION("My Chip");
MODULE_LICENSE("GPL");
MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}");
So far, you've learned how to write the driver codes. And you might have a question now: how to put my own driver into the ALSA driver tree? Here (finally :) the standard procedure is described briefly.
Suppose that you create a new PCI driver for the card
“xyz”. The card module name would be
snd-xyz. The new driver is usually put into the alsa-driver
tree, alsa-driver/pci directory in
the case of PCI cards.
Then the driver is evaluated, audited and tested
by developers and users. After a certain time, the driver
will go to the alsa-kernel tree (to the corresponding directory,
such as alsa-kernel/pci) and eventually
will be integrated into the Linux 2.6 tree (the directory would be
linux/sound/pci).
In the following sections, the driver code is supposed to be put into alsa-driver tree. The two cases are covered: a driver consisting of a single source file and one consisting of several source files.
Modify alsa-driver/pci/Makefile
Suppose you have a file xyz.c. Add the following two lines
snd-xyz-objs := xyz.o
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
Create the Kconfig entry
Add the new entry of Kconfig for your xyz driver.
config SND_XYZ
tristate "Foobar XYZ"
depends on SND
select SND_PCM
help
Say Y here to include support for Foobar XYZ soundcard.
To compile this driver as a module, choose M here: the module
will be called snd-xyz.
the line, select SND_PCM, specifies that the driver xyz supports PCM. In addition to SND_PCM, the following components are supported for select command: SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART, SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC. Add the select command for each supported component.
Note that some selections imply the lowlevel selections. For example, PCM includes TIMER, MPU401_UART includes RAWMIDI, AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP. You don't need to give the lowlevel selections again.
For the details of Kconfig script, refer to the kbuild documentation.
Run cvscompile script to re-generate the configure script and build the whole stuff again.
Suppose that the driver snd-xyz have several source files. They are located in the new subdirectory, pci/xyz.
Add a new directory (xyz) in
alsa-driver/pci/Makefile as below
obj-$(CONFIG_SND) += xyz/
Under the directory xyz, create a Makefile
Example 15.1. Sample Makefile for a driver xyz
ifndef SND_TOPDIR
SND_TOPDIR=../..
endif
include $(SND_TOPDIR)/toplevel.config
include $(SND_TOPDIR)/Makefile.conf
snd-xyz-objs := xyz.o abc.o def.o
obj-$(CONFIG_SND_XYZ) += snd-xyz.o
include $(SND_TOPDIR)/Rules.make
Create the Kconfig entry
This procedure is as same as in the last section.
Run cvscompile script to re-generate the configure script and build the whole stuff again.
Table of Contents
ALSA provides a verbose version of the
printk() function. If a kernel config
CONFIG_SND_VERBOSE_PRINTK is set, this
function prints the given message together with the file name
and the line of the caller. The KERN_XXX
prefix is processed as
well as the original printk() does, so it's
recommended to add this prefix, e.g.
snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\n");
There are also printk()'s for
debugging. snd_printd() can be used for
general debugging purposes. If
CONFIG_SND_DEBUG is set, this function is
compiled, and works just like
snd_printk(). If the ALSA is compiled
without the debugging flag, it's ignored.
snd_printdd() is compiled in only when
CONFIG_SND_DEBUG_VERBOSE is set. Please note
that CONFIG_SND_DEBUG_VERBOSE is not set as default
even if you configure the alsa-driver with
--with-debug=full option. You need to give
explicitly --with-debug=detect option instead.
It shows the BUG? message and
stack trace as well as snd_BUG_ON at the point.
It's useful to show that a fatal error happens there.
When no debug flag is set, this macro is ignored.
snd_BUG_ON() macro is similar with
WARN_ON() macro. For example,
snd_BUG_ON(!pointer);
or it can be used as the condition,
if (snd_BUG_ON(non_zero_is_bug))
return -EINVAL;
The macro takes an conditional expression to evaluate.
When CONFIG_SND_DEBUG, is set, the
expression is actually evaluated. If it's non-zero, it shows
the warning message such as
BUG? (xxx)
normally followed by stack trace. It returns the evaluated
value.
When no CONFIG_SND_DEBUG is set, this
macro always returns zero.