Merge git://git.kernel.org/pub/scm/linux/kernel/git/perex/alsa

    This module supports multiple cards and autoprobe.

    The power-management is supported.

  Module snd-ens1371

  ------------------

    This module supports multiple cards and autoprobe.

    The power-management is supported.

  Module snd-es968

  ----------------

    model	- force the model name

    position_fix - Fix DMA pointer (0 = auto, 1 = none, 2 = POSBUF, 3 = FIFO size)

    single_cmd  - Use single immediate commands to communicate with

		codecs (for debugging only)

    This module supports one card and autoprobe.

	  asus		3-jack

	  uniwill	3-jack

	  F1734		2-jack

	  lg		LG laptop (m1 express dual)

	  test		for testing/debugging purpose, almost all controls can be

			adjusted.  Appearing only when compiled with

			$CONFIG_SND_DEBUG=y

	  auto		auto-config reading BIOS (default)

	ALC260

	  hp		HP machines

	  fujitsu	Fujitsu S7020

	  acer		Acer TravelMate

	  basic		fixed pin assignment (old default model)

	  auto		auto-config reading BIOS (default)

	ALC262

	  fujitsu	Fujitsu Laptop

	  basic		fixed pin assignment w/o SPDIF

	  auto		auto-config reading BIOS (default)

	ALC882/883/885

	  3stack-dig	3-jack with SPDIF I/O

	  6stck-dig	6-jack digital with SPDIF I/O

	  auto		auto-config reading BIOS (default)

	ALC861

	  3stack	3-jack

	  3stack-dig	3-jack with SPDIF I/O

	  6stack-dig	6-jack with SPDIF I/O

	  auto		auto-config reading BIOS (default)

	CMI9880

	  minimal	3-jack in back

	  allout	5-jack in back, 2-jack in front, SPDIF out

	  auto		auto-config reading BIOS (default)

	AD1981

	  basic		3-jack (default)

	  hp		HP nx6320

	AD1986A

	  6stack	6-jack, separate surrounds (default)

	  3stack	3-stack, shared surrounds

	  laptop	2-channel only (FSC V2060, Samsung M50)

	  laptop-eapd	2-channel with EAPD (Samsung R65, ASUS A6J)

	AD1988

	  6stack	6-jack

	  6stack-dig	ditto with SPDIF

	  3stack	3-jack

	  3stack-dig	ditto with SPDIF

	  laptop	3-jack with hp-jack automute

	  laptop-dig	ditto with SPDIF

	  auto		auto-confgi reading BIOS (default)

	STAC7661(?)

	  vaio		Setup for VAIO FE550G/SZ110

    If the default configuration doesn't work and one of the above

    matches with your device, report it together with the PCI

    subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel

	    (Usually SD_LPLIB register is more accurate than the

	    position buffer.)

    NB: If you get many "azx_get_response timeout" messages at

    loading, it's likely a problem of interrupts (e.g. ACPI irq

    routing).  Try to boot with options like "pci=noacpi".  Also, you

    can try "single_cmd=1" module option.  This will switch the

    communication method between HDA controller and codecs to the

    single immediate commands instead of CORB/RIRB.  Basically, the

    single command mode is provided only for BIOS, and you won't get

    unsolicited events, too.  But, at least, this works independently

    from the irq.  Remember this is a last resort, and should be

    avoided as much as possible...

    The power-management is supported.

  Module snd-hdsp

  ------------------

    Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.

			* MidiMan M Audio Revolution 5.1

			* MidiMan M Audio Revolution 7.1

			* AMP Ltd AUDIO2000

			* TerraTec Aureon 5.1 Sky

			* TerraTec Phase 22

			* TerraTec Phase 28

			* AudioTrak Prodigy 7.1

			* AudioTrak Prodigy 7.1LT

			* AudioTrak Prodigy 192

			* Pontis MS300

			* Albatron K8X800 Pro II 

			* Shuttle SN25P

    model       - Use the given board model, one of the following:

		  revo71, amp2000, prodigy71, prodigy192, aureon51,

		  aureon71, universe, k8x800, phase22, phase28, ms300,

		  av710

		  revo51, revo71, amp2000, prodigy71, prodigy71lt,

		  prodigy192, aureon51, aureon71, universe,

		  k8x800, phase22, phase28, ms300, av710

    This module supports multiple cards and autoprobe.

    vid             - Vendor ID for the device (optional)

    pid             - Product ID for the device (optional)

    device_setup    - Device specific magic number (optional)

                    - Influence depends on the device

                    - Default: 0x0000 

    This module supports multiple devices, autoprobe and hotplugging.

	Guide to using M-Audio Audiophile USB with ALSA and Jack	v1.2

	========================================================

	    Thibault Le Meur <Thibault.LeMeur@supelec.fr>

This document is a guide to using the M-Audio Audiophile USB (tm) device with 

ALSA and JACK.

1 - Audiophile USB Specs and correct usage

==========================================

This part is a reminder of important facts about the functions and limitations 

of the device.

The device has 4 audio interfaces, and 2 MIDI ports:

 * Analog Stereo Input (Ai)

   - This port supports 2 pairs of line-level audio inputs (1/4" TS and RCA) 

   - When the 1/4" TS (jack) connectors are connected, the RCA connectors

     are disabled

 * Analog Stereo Output (Ao)

 * Digital Stereo Input (Di)

 * Digital Stereo Output (Do)

 * Midi In (Mi)

 * Midi Out (Mo)

The internal DAC/ADC has the following caracteristics:

* sample depth of 16 or 24 bits

* sample rate from 8kHz to 96kHz

* Two ports can't use different sample depths at the same time.Moreover, the 

Audiophile USB documentation gives the following Warning: "Please exit any 

audio application running before switching between bit depths"

Due to the USB 1.1 bandwidth limitation, a limited number of interfaces can be 

activated at the same time depending on the audio mode selected:

 * 16-bit/48kHz ==> 4 channels in/ 4 channels out

   - Ai+Ao+Di+Do

 * 24-bit/48kHz ==> 4 channels in/2 channels out, 

                    or 2 channels in/4 channels out

   - Ai+Ao+Do or Ai+Di+Ao or Ai+Di+Do or Di+Ao+Do

 * 24-bit/96kHz ==> 2 channels in, or 2 channels out (half duplex only)

   - Ai or Ao or Di or Do

Important facts about the Digital interface:

--------------------------------------------

 * The Do port additionnaly supports surround-encoded AC-3 and DTS passthrough, 

though I haven't tested it under linux

   - Note that in this setup only the Do interface can be enabled

 * Apart from recording an audio digital stream, enabling the Di port is a way 

to synchronize the device to an external sample clock

   - As a consequence, the Di port must be enable only if an active Digital 

source is connected

   - Enabling Di when no digital source is connected can result in a 

synchronization error (for instance sound played at an odd sample rate)

2 - Audiophile USB support in ALSA

==================================

2.1 - MIDI ports

----------------

The Audiophile USB MIDI ports will be automatically supported once the 

following modules have been loaded:

 * snd-usb-audio

 * snd-seq

 * snd-seq-midi

No additionnal setting is required.

2.2 - Audio ports

-----------------

Audio functions of the Audiophile USB device are handled by the snd-usb-audio 

module. This module can work in a default mode (without any device-specific 

parameter), or in an advanced mode with the device-specific parameter called 

"device_setup".

2.2.1 - Default Alsa driver mode

The default behaviour of the snd-usb-audio driver is to parse the device 

capabilities at startup and enable all functions inside the device (including 

all ports at any sample rates and any sample depths supported). This approach 

has the advantage to let the driver easily switch from sample rates/depths 

automatically according to the need of the application claiming the device.

In this case the Audiophile ports are mapped to alsa pcm devices in the 

following way (I suppose the device's index is 1):

 * hw:1,0 is Ao in playback and Di in capture

 * hw:1,1 is Do in playback and Ai in capture

 * hw:1,2 is Do in AC3/DTS passthrough mode

You must note as well that the device uses Big Endian byte encoding so that 

supported audio format are S16_BE  for 16-bit depth modes and S24_3BE for 

24-bits depth mode. One exception is the hw:1,2 port which is Little Endian 

compliant and thus uses S16_LE.

Examples:

 * playing a S24_3BE encoded raw file to the Ao port

   % aplay -D hw:1,0 -c2 -t raw -r48000 -fS24_3BE test.raw

 * recording a  S24_3BE encoded raw file from the Ai port

   % arecord -D hw:1,1 -c2  -t raw -r48000 -fS24_3BE test.raw

 * playing a S16_BE encoded raw file to the Do port

   % aplay -D hw:1,1 -c2 -t raw -r48000 -fS16_BE test.raw

If you're happy with the default Alsa driver setup and don't experience any 

issue with this mode, then you can skip the following chapter.

2.2.2 - Advanced module setup

Due to the hardware constraints described above, the device initialization made 

by the Alsa driver in default mode may result in a corrupted state of the 

device. For instance, a particularly annoying issue is that the sound captured 

from the Ai port sounds distorted (as if boosted with an excessive high volume 

gain).

For people having this problem, the snd-usb-audio module has a new module 

parameter called "device_setup".

2.2.2.1 - Initializing the working mode of the Audiohile USB

As far as the Audiohile USB device is concerned, this value let the user 

specify:

 * the sample depth

 * the sample rate

 * whether the Di port is used or not 

Here is a list of supported device_setup values for this device:

 * device_setup=0x00 (or omitted)

   - Alsa driver default mode

   - maintains backward compatibility with setups that do not use this 

     parameter by not introducing any change

   - results sometimes in corrupted sound as decribed earlier

 * device_setup=0x01

   - 16bits 48kHz mode with Di disabled

   - Ai,Ao,Do can be used at the same time

   - hw:1,0 is not available in capture mode

   - hw:1,2 is not available

 * device_setup=0x11

   - 16bits 48kHz mode with Di enabled

   - Ai,Ao,Di,Do can be used at the same time

   - hw:1,0 is available in capture mode

   - hw:1,2 is not available

 * device_setup=0x09

   - 24bits 48kHz mode with Di disabled

   - Ai,Ao,Do can be used at the same time

   - hw:1,0 is not available in capture mode

   - hw:1,2 is not available

 * device_setup=0x19

   - 24bits 48kHz mode with Di enabled

   - 3 ports from {Ai,Ao,Di,Do} can be used at the same time

   - hw:1,0 is available in capture mode and an active digital source must be 

     connected to Di

   - hw:1,2 is not available

 * device_setup=0x0D or 0x10

   - 24bits 96kHz mode

   - Di is enabled by default for this mode but does not need to be connected 

     to an active source

   - Only 1 port from {Ai,Ao,Di,Do} can be used at the same time

   - hw:1,0 is available in captured mode

   - hw:1,2 is not available

 * device_setup=0x03

   - 16bits 48kHz mode with only the Do port enabled 

   - AC3 with DTS passthru (not tested)

   - Caution with this setup the Do port is mapped to the pcm device hw:1,0

2.2.2.2 - Setting and switching configurations with the device_setup parameter

The parameter can be given:

 * By manually probing the device (as root):

   # modprobe -r snd-usb-audio

   # modprobe snd-usb-audio index=1 device_setup=0x09

 * Or while configuring the modules options in your modules configuration file

   - For Fedora distributions, edit the /etc/modprobe.conf file:

       alias snd-card-1 snd-usb-audio

       options snd-usb-audio index=1 device_setup=0x09

IMPORTANT NOTE WHEN SWITCHING CONFIGURATION:

-------------------------------------------

 * You may need to _first_ intialize the module with the correct device_setup 

   parameter and _only_after_ turn on the Audiophile USB device

 * This is especially true when switching the sample depth:

   - first trun off the device

   - de-register the snd-usb-audio module

   - change the device_setup parameter (by either manually reprobing the module 

     or changing modprobe.conf)

   - turn on the device

2.2.2.3 - Audiophile USB's device_setup structure

If you want to understand the device_setup magic numbers for the Audiophile 

USB, you need some very basic understanding of binary computation. However, 

this is not required to use the parameter and you may skip thi section.

The device_setup is one byte long and its structure is the following:

       +---+---+---+---+---+---+---+---+

       | b7| b6| b5| b4| b3| b2| b1| b0|

       +---+---+---+---+---+---+---+---+

       | 0 | 0 | 0 | Di|24B|96K|DTS|SET|

       +---+---+---+---+---+---+---+---+

Where:

 * b0 is the "SET" bit

   - it MUST be set if device_setup is initialized 

 * b1 is the "DTS" bit

   - it is set only for Digital output with DTS/AC3

   - this setup is not tested

 * b2 is the Rate selection flag

   - When set to "1" the rate range is 48.1-96kHz

   - Otherwise the sample rate range is 8-48kHz

 * b3 is the bit depth selection flag

   - When set to "1" samples are 24bits long

   - Otherwise they are 16bits long

   - Note that b2 implies b3 as the 96kHz mode is only supported for 24 bits 

     samples

 * b4 is the Digital input flag

   - When set to "1" the device assumes that an active digital source is 

     connected 

   - You shouldn't enable Di if no source is seen on the port (this leads to 

     synchronization issues)

   - b4 is implied by b2 (since only one port is enabled at a time no synch 

     error can occur) 

 * b5 to b7 are reserved for future uses, and must be set to "0"

   - might become Ao, Do, Ai, for b7, b6, b4 respectively

Caution:

 * there is no check on the value you will give to device_setup

   - for instance choosing 0x05 (16bits 96kHz) will fail back to 0x09 since 

     b2 implies b3. But _there_will_be_no_warning_ in /var/log/messages

 * Hardware constraints due to the USB bus limitation aren't checked

   - choosing b2 will prepare all interfaces for 24bits/96kHz but you'll

     only be able to use one at the same time

2.2.3 -  USB implementation details for this device

You may safely skip this section if you're not interrested in driver 

development.

This section describes some internals aspect of the device and summarize the 

data I got by usb-snooping the windows and linux drivers.

The M-Audio Audiophile USB has 7 USB Interfaces:

a "USB interface":

 * USB Interface nb.0

 * USB Interface nb.1

   - Audio Control function

 * USB Interface nb.2

   - Analog Output

 * USB Interface nb.3

   - Digital Output

 * USB Interface nb.4

   - Analog Input

 * USB Interface nb.5

   - Digital Input

 * USB Interface nb.6

   - MIDI interface compliant with the MIDIMAN quirk 

Each interface has 5 altsettings (AltSet 1,2,3,4,5) except:

 * Interface 3 (Digital Out) has an extra Alset nb.6 

 * Interface 5 (Digital In) does not have Alset nb.3 and 5 

Here is a short description of the AltSettings capabilities:

 * AltSettings 1 corresponds to

  - 24-bit depth, 48.1-96kHz sample mode

  - Adaptive playback (Ao and Do), Synch capture (Ai), or Asynch capture (Di)

 * AltSettings 2 corresponds to

  - 24-bit depth, 8-48kHz sample mode

  - Asynch capture and playback  (Ao,Ai,Do,Di)

 * AltSettings 3 corresponds to

  - 24-bit depth, 8-48kHz sample mode

  - Synch capture (Ai) and Adaptive playback (Ao,Do)

 * AltSettings 4 corresponds to

  - 16-bit depth, 8-48kHz sample mode

  - Asynch capture and playback  (Ao,Ai,Do,Di)

 * AltSettings 5 corresponds to

  - 16-bit depth, 8-48kHz sample mode

  - Synch capture (Ai) and Adaptive playback (Ao,Do)

 * AltSettings 6 corresponds to

  - 16-bit depth, 8-48kHz sample mode

  - Synch playback (Do), audio format type III IEC1937_AC-3

In order to ensure a correct intialization of the device, the driver 

_must_know_ how the device will be used:

 * if DTS is choosen, only Interface 2 with AltSet nb.6 must be

   registered

 * if 96KHz only AltSets nb.1 of each interface must be selected

 * if samples are using 24bits/48KHz then AltSet 2 must me used if

   Digital input is connected, and only AltSet nb.3 if Digital input

   is not connected

 * if samples are using 16bits/48KHz then AltSet 4 must me used if

   Digital input is connected, and only AltSet nb.5 if Digital input

   is not connected

When device_setup is given as a parameter to the snd-usb-audio module, the 

parse_audio_enpoint function uses a quirk called 

"audiophile_skip_setting_quirk" in order to prevent AltSettings not 

corresponding to device_setup from being registered in the driver.

3 - Audiophile USB and Jack support

===================================

This section deals with support of the Audiophile USB device in Jack.

The main issue regarding this support is that the device is Big Endian 

compliant.

3.1 - Using the plug alsa plugin

--------------------------------

Jack doesn't directly support big endian devices. Thus, one way to have support 

for this device with Alsa is to use the Alsa "plug" converter.

For instance here is one way to run Jack with 2 playback channels on Ao and 2 

capture channels from Ai:

  % jackd -R -dalsa -dplughw:1 -r48000 -p256 -n2 -D -Cplughw:1,1

However you may see the following warning message:

"You appear to be using the ALSA software "plug" layer, probably a result of 

using the "default" ALSA device. This is less efficient than it could be. 

Consider using a hardware device instead rather than using the plug layer."

3.2 - Patching alsa to use direct pcm device

-------------------------------------------

A patch for Jack by Andreas Steinmetz adds support for Big Endian devices. 

However it has not been included in the CVS tree.

You can find it at the following URL:

http://sourceforge.net/tracker/index.php?func=detail&aid=1289682&group_id=39687&

atid=425939

After having applied the patch you can run jackd with the following command 

line:

  % jackd -R -dalsa -Phw:1,0 -r48000 -p128 -n2 -D -Chw:1,1

          mychip_set_sample_format(chip, runtime->format);

          mychip_set_sample_rate(chip, runtime->rate);

          mychip_set_channels(chip, runtime->channels);

          mychip_set_dma_setup(chip, runtime->dma_area,

          mychip_set_dma_setup(chip, runtime->dma_addr,

                               chip->buffer_size,

                               chip->period_size);

          return 0;

          .name = "PCM Playback Switch",

          .index = 0,

          .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,

          .private_values = 0xffff,

          .private_value = 0xffff,

          .info = my_control_info,

          .get = my_control_get,

          .put = my_control_put

      </para>

      <para>

        The <structfield>private_values</structfield> field contains

        The <structfield>private_value</structfield> field contains

      an arbitrary long integer value for this record. When using

      generic <structfield>info</structfield>,

      <structfield>get</structfield> and

	.resource	= smc91x_resources,

};

static int mst_audio_startup(snd_pcm_substream_t *substream, void *priv)

static int mst_audio_startup(struct snd_pcm_substream *substream, void *priv)

{

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

		MST_MSCWR2 &= ~MST_MSCWR2_AC97_SPKROFF;

	return 0;

}

static void mst_audio_shutdown(snd_pcm_substream_t *substream, void *priv)

static void mst_audio_shutdown(struct snd_pcm_substream *substream, void *priv)

{

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)

		MST_MSCWR2 |= MST_MSCWR2_AC97_SPKROFF;

	/* audio controls */

	int                        irq;

	snd_card_t                 *card;

	struct snd_card            *card;

	spinlock_t                 reg_lock;

	struct cx88_buffer   *buf;

	long opened;

	snd_pcm_substream_t *substream;

	struct snd_pcm_substream *substream;

};

typedef struct cx88_audio_dev snd_cx88_card_t;

static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */

static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */

static int enable[SNDRV_CARDS] = {1, [1 ... (SNDRV_CARDS - 1)] = 1};

static snd_card_t *snd_cx88_cards[SNDRV_CARDS];

static struct snd_card *snd_cx88_cards[SNDRV_CARDS];

module_param_array(enable, bool, NULL, 0444);

MODULE_PARM_DESC(enable, "Enable cx88x soundcard. default enabled.");

/*

 * Digital hardware definition

 */

static snd_pcm_hardware_t snd_cx88_digital_hw = {

static struct snd_pcm_hardware snd_cx88_digital_hw = {

	.info = SNDRV_PCM_INFO_MMAP |

		SNDRV_PCM_INFO_INTERLEAVED |

		SNDRV_PCM_INFO_BLOCK_TRANSFER |

/*

 * audio pcm capture runtime free

 */

static void snd_card_cx88_runtime_free(snd_pcm_runtime_t *runtime)

static void snd_card_cx88_runtime_free(struct snd_pcm_runtime *runtime)

{

}

/*

 * audio pcm capture open callback

 */

static int snd_cx88_pcm_open(snd_pcm_substream_t *substream)

static int snd_cx88_pcm_open(struct snd_pcm_substream *substream)

{

	snd_cx88_card_t *chip = snd_pcm_substream_chip(substream);

	snd_pcm_runtime_t *runtime = substream->runtime;

	struct snd_pcm_runtime *runtime = substream->runtime;

	int err;

	if (test_and_set_bit(0, &chip->opened))

/*

 * audio close callback

 */

static int snd_cx88_close(snd_pcm_substream_t *substream)

static int snd_cx88_close(struct snd_pcm_substream *substream)

{

	snd_cx88_card_t *chip = snd_pcm_substream_chip(substream);

/*

 * hw_params callback

 */

static int snd_cx88_hw_params(snd_pcm_substream_t * substream,

				 snd_pcm_hw_params_t * hw_params)

static int snd_cx88_hw_params(struct snd_pcm_substream * substream,

			      struct snd_pcm_hw_params * hw_params)

{

	snd_cx88_card_t *chip = snd_pcm_substream_chip(substream);

	struct cx88_buffer *buf;

/*

 * hw free callback

 */

static int snd_cx88_hw_free(snd_pcm_substream_t * substream)

static int snd_cx88_hw_free(struct snd_pcm_substream * substream)

{

	snd_cx88_card_t *chip = snd_pcm_substream_chip(substream);

/*

 * prepare callback

 */

static int snd_cx88_prepare(snd_pcm_substream_t *substream)

static int snd_cx88_prepare(struct snd_pcm_substream *substream)

{

	return 0;

}

/*

 * trigger callback

 */

static int snd_cx88_card_trigger(snd_pcm_substream_t *substream, int cmd)

static int snd_cx88_card_trigger(struct snd_pcm_substream *substream, int cmd)

{

	snd_cx88_card_t *chip = snd_pcm_substream_chip(substream);

	int err;

/*

 * pointer callback

 */

static snd_pcm_uframes_t snd_cx88_pointer(snd_pcm_substream_t *substream)

static snd_pcm_uframes_t snd_cx88_pointer(struct snd_pcm_substream *substream)

{

	snd_cx88_card_t *chip = snd_pcm_substream_chip(substream);

	snd_pcm_runtime_t *runtime = substream->runtime;

	struct snd_pcm_runtime *runtime = substream->runtime;

	if (chip->read_count) {

		chip->read_count -= snd_pcm_lib_period_bytes(substream);

/*

 * operators

 */

static snd_pcm_ops_t snd_cx88_pcm_ops = {

static struct snd_pcm_ops snd_cx88_pcm_ops = {

	.open = snd_cx88_pcm_open,

	.close = snd_cx88_close,

	.ioctl = snd_pcm_lib_ioctl,

static int __devinit snd_cx88_pcm(snd_cx88_card_t *chip, int device, char *name)

{

	int err;

	snd_pcm_t *pcm;

	struct snd_pcm *pcm;

	err = snd_pcm_new(chip->card, name, device, 0, 1, &pcm);

	if (err < 0)

/****************************************************************************

				CONTROL INTERFACE

 ****************************************************************************/

static int snd_cx88_capture_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *info)

static int snd_cx88_capture_volume_info(struct snd_kcontrol *kcontrol,

					struct snd_ctl_elem_info *info)

{

	info->type = SNDRV_CTL_ELEM_TYPE_INTEGER;

	info->count = 1;

}

/* OK - TODO: test it */

static int snd_cx88_capture_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *value)

static int snd_cx88_capture_volume_get(struct snd_kcontrol *kcontrol,

				       struct snd_ctl_elem_value *value)

{

	snd_cx88_card_t *chip = snd_kcontrol_chip(kcontrol);

	struct cx88_core *core=chip->core;

}

/* OK - TODO: test it */

static int snd_cx88_capture_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *value)

static int snd_cx88_capture_volume_put(struct snd_kcontrol *kcontrol,

				       struct snd_ctl_elem_value *value)

{

	snd_cx88_card_t *chip = snd_kcontrol_chip(kcontrol);

	struct cx88_core *core=chip->core;

	return v != old_control;

}

static snd_kcontrol_new_t snd_cx88_capture_volume = {

static struct snd_kcontrol_new snd_cx88_capture_volume = {

	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,

	.name = "Capture Volume",

	.info = snd_cx88_capture_volume_info,

/*

 * Component Destructor

 */

static void snd_cx88_dev_free(snd_card_t * card)

static void snd_cx88_dev_free(struct snd_card * card)

{

	snd_cx88_card_t *chip = card->private_data;

 */

static int devno;

static int __devinit snd_cx88_create(snd_card_t *card, struct pci_dev *pci,

static int __devinit snd_cx88_create(struct snd_card *card,

				     struct pci_dev *pci,

				     snd_cx88_card_t **rchip)

{

	snd_cx88_card_t   *chip;

static int __devinit cx88_audio_initdev(struct pci_dev *pci,

				    const struct pci_device_id *pci_id)

{

	snd_card_t       *card;

	struct snd_card  *card;

	snd_cx88_card_t  *chip;

	int              err;