Changes v1.0.18rc2 v1.0.18rc3

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Changelog between 1.0.18rc2 and 1.0.18rc3 releases

alsa-utils

Core

Remove unneeded files from EXTRA_DIST in root Makefile.am
Release v1.0.18rc3

ALSA Control (alsactl)

Add missing files to be included in alsactl/Makefile.am

aplay/arecord

aplay - show sample arrangement (mmap areas) in verbose more for debugging

alsa-plugins

Core

Fix build with recent ffmpeg
Release v1.0.18rc3

A52 Output plugin

Fix build with recent ffmpeg

PulseAudio -> ALSA plugin

Reindent to Linux kernel style
Add Emacs-style /*-*- linux-c -*-*/ header comment
Make pulse_new() a proper C function
Don't modify the SIGPIPE handler
Call pa_context_disconnect() explicitly
use SNDERR instead of fprintf to print error messages
Support S32 sample types
Add trailing NUL character to snprintf output
Get rid of pulse_poll_revents()
Add more error checking
Remove fix for bug 0003470
Rework hardware parameter selection
A bag of minor clean ups for ctl_pulse.c
Make pulse_ext_callback const
Drop our own implementation of the poll() callbacks
A bag of minor clean-ups for pulse.c
Split out O_NONBLOCK setting into seperate function
Save a byte of memory
Adjust buffering metrics to match what PA internally uses
Make sure we always have a sensible channel mapping
Use PA_STREAM_EARLY_REQUESTS if available
Use S32/FLOAT32 only where available in the PA libs
Add const to our snd_pcm_ioplug_callback_t instances
Don't implement our own poll handlers, we can use the default ones
Remove our own poll handler implementation entirely
A bag of clean-ups for pcm_ctl.c

libavcodec's resampler

Fix build with recent ffmpeg

Changelog between 1.0.18rc1 and 1.0.18rc3 releases

alsa-driver

Sound Core

Fix include files in configure checks
Fix ARCH detection for ppc and ppc64
Add arch/include check to configure
soc - Add a workaround for incompatible debugfs
Release v1.0.18rc3
sound: make OSS sound core optional
Fix CONFIG_AC97_BUS dependency

ALSA Core

Add patches for sparc/*.c
ALSA: Clean up SG-buffer helper functions and macros
ALSA: Allocate larger pages in sgbuf
sound: make OSS sound core optional

SoC PXA2xx Core

ALSA: ASoC: fix pxa2xx-i2s clk_get call

Control Midlevel

ALSA: use correct lock in snd_ctl_dev_disconnect()

PCM Midlevel

Regenerated pcm_native.patch
ALSA: Clean up SG-buffer helper functions and macros
ALSA: Allocate larger pages in sgbuf
ALSA: add dummy function to support shared mmap in nommu Blackfin arch

/soc/Makefile

sound: ASoC: Blackfin: Include Blackfin architecture support in build

/soc/codecs/Makefile

ALSA: ASoC: Add WM8903 CODEC driver
sound: ASoC codec: SSM2602 audio codec driver
ALSA: ASoC: Sort Kconfig and Makefile entries for codecs

/sparc/Makefile

Add patches for sparc/*.c

AC97 Codec

ALSA: ac97 - vt1618 7.1 Audio Support
ALSA: ac97_patch make functions static

AD1816A driver

ALSA: snd-ad1816a: add different TerraTec EWS64 S model

AD1848 driver

ALSA: remove stale files

ADI Blackfin ASoC

Add soc/blackfin build stub
sound: ASoC: Blackfin: SPORT peripheral interface driver
sound: ASoC: Blackfin: DMA Driver for AC97 sound chip
sound: ASoC: Blackfin: AC97 Blackfin CPU DAI driver
sound: ASoC: Blackfin: DMA Driver for I2S sound chip
sound: ASoC: Blackfin: I2S CPU DAI driver
sound: ASoC: Blackfin: board driver for AD1980/1 audio codec
sound: ASoC: Blackfin: board driver for SSM2602 sound chip
sound: ASoC: Blackfin: add Blackfin arch ASoC Kconfig and Makefile

AK4XXX AD/DA converters

ALSA: revo51: add headphone output

ALS4000 driver

ALSA: als4000 - Code clean up
ALSA: ALS4000 driver work, step 2

ALSA Version

ALSA: Release v1.0.18rc3

ARM AACI PL041 driver

ALSA: aaci - Fix NULL test at error path

ATIIXP driver

ALSA: atiixp - Add PCI ID for SB600 (1002:4382)

Apple Onboard Audio driver

ALSA: aoa - Remove duplicate assignment of SNDRV_PCM_FMTBIT_S{16,24}_BE bits

Asihpi driver

split public and private defs
add control-specific query functions
use new query functions
add extra checks to control cache
Update to hpi_internal.h. Minor cleanups
No longer used
Rewrite PAD control string caching
Simplify DSP crashed flag
Add ASI1723, log buffer properties. Tweak function params for SWIG.
Remove duplicate func prototypes (also in hpimsginit.h)
add missing comma in list
add buffer size member to adapter struct
HPI Stream buffers allocated according to application request
style fixes
need hpimsginit.h
Better error handling in tuner query. Need hpimsginit.h
checkpatch fixes
asihpi - Fix compile warnings regarding printf and size_t
Fix format in portable way.
Include adapter buffer alloc inside adapter mutex
Add support for ASI50xx SSX (multichannel) mode.
asihpi - Mem buffer alloc inside adapter mutex.
asihpi - Avoid null pointer dereference
asihpi - Return from timer func if XRUN
asihpi - Turn off verbose logging

BT87x driver

ALSA: Clean up SG-buffer helper functions and macros

CA0106 driver

ALSA: CA0106 on MSI K8N Diamond PLUS Motherboard

CMI8788 (Oxygen) driver

ALSA: oxygen: prevent muting of nonexistent AC97 controls
ALSA: oxygen: fix distorted output on AK4396-based cards

Documentation

ALSA: hda - Add AD1882A codec support
ALSA: hda - Add descriptions of missing model values
ALSA: hda - Add support of Toshiba RX1
ALSA: Documentation/sound/alsa/ALSA-Configuration.txt: fix typo

HDA Codec driver

ALSA: hda - Fix capture source widgets on ALC codecs
ALSA: hda - Fix call of alc888_coef_init()
ALSA: hda - Add definitions of HDMI-related verbs
ALSA: hda - set up HDMI channels
ALSA: hda - Add AD1882A codec support
ALSA: hda - Create beep control on ALC269 codec
ALSA: hda - Fix ALC269 capture source
ALSA: hda - fix spaces in patch_realtek.c
ALSA: hda - Add support of HP DC7600
ALSA: hda - Add support of Toshiba S06
ALSA: hda - Add support of Acer Aspire One
ALSA: hda - Add support of Quanta FL1
ALSA: hda - support more ALC888 devices
ALSA: hda - Add more ALC662/663 codec support
sound: Revert "ALSA: hda - Added model selection for iMac 24""
ALSA: hda - Add descriptions of missing model values
ALSA: hda - Add support of Toshiba RX1
ALSA: hda: Distortion fix for dell_m6_core_init
ALSA: hda - Fix ALC663 auto-probe
ALSA: hda - Add mic-boost controls to ALC662/663 auto configuration
ALSA: hda - Add model for Toshiba L305 laptop
ALSA: hda: SPDIF mux controls
ALSA: hda: removed unneeded hp_nid references
ALSA: hda: slave digital out support
ALSA: hda: digital slave support for IDT codecs
ALSA: hda: SPDIF mux name change
ALSA: hda - add missing slave_dig_outs for 92HD73bxx
ALSA: HDA patch_via.c: cleanup
ALSA: HDA patch_via.c: HP and CD pin connect config
ALSA: HDA patch_via.c: Fix inversion of surround and side channels
ALSA: HDA patch_via.c: Add VT1708S and VT1702 support
ALSA: HDA patch_via.c: Mute on headphone plug-in
ALSA: HDA patch_via.c: Independent DAC for headphone
ALSA: HDA patch_via.c: Second S/PDIF (HDMI) support

HDA Intel driver

ALSA: hda_intel: enable snoop for nvidia HDA controller
ALSA: hda - disable delayed period-ack with bdl_pos_adj=0
ALSA: hda - check page continuity
ALSA: hda - Fix VIA recording problem
ALSA: hda - allow probing of 4 codecs

HDA generic driver

ALSA: hda - print a few HDMI information in proc

ICE1712 driver

ALSA: ice1724: enable MIDI on Terratec Aureon 7.1 Universe
ALSA: ice1724: Enable MIDI on TerraTec PHASE 22 and PHASE 28
ALSA: ice1712/aureon.c: fix coding style
ALSA: ice1712/wtm: fix coding style
ALSA: ice1712/ice1724: Coding style fixes part 1 (more coming up)
ALSA: ice1712.c: fix coding style

ICE1724 driver

ALSA: revo51: add headphone output
ALSA: ice1724 - limit channels for multi-channel playback
ALSA: ice1712/ice1724: Coding style fixes part 1 (more coming up)
ALSA: ice1724.c: fix coding style

ISA

ALSA: remove obsolete *_FIRMARE_IN_KERNEL in isa/Kconfig

Intel8x0 driver

ALSA: intel8x0: implement ac97_clock whitelist
ALSA: intel8x0 - use snd_pci_quirk for clock list

MIPS AU1x00 driver

removed unused #include <linux/version.h>'s

Memalloc module

ALSA: Clean up SG-buffer helper functions and macros
ALSA: Fix allocation size calculation in snd_dma_alloc_pages_fallback()
ALSA: Allocate larger pages in sgbuf

OPL3SA2 driver

ALSA: opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
ALSA: opl3sa2: replace nd_opl3sa2_info_xxx() with snd_wss_info_xxx()

OSS device core

sound: make OSS sound core optional
ALSA: fix section mismatch in the sound_core

PARISC Harmony driver

ALSA: harmony - fix a typo

PPC AWACS driver

ALSA: snd-powermac: mixers for PowerMac G4 AGP
ALSA: snd-powermac: HP detection for 1st iMac G3 SL

SA11xx UDA1341 driver

ALSA: fix section for sa11xx-uda1341 platform driver

SB drivers

ALSA: ALS4000 driver work, step 2

SPARC AMD7930 driver

Add patches for sparc/*.c

SoC Audio for the Atmel AT32 System-on-Chip

sound: ASoC: Convert wm8510 to a new-style i2c driver

SoC Audio for the Atmel AT91 System-on-Chip

removed unused #include <linux/version.h>'s
ALSA: ASoC: Add missing new-line at end of strings
ALSA: ASoC: Convert wm8731 to a new-style i2c driver

SoC Audio for the Samsung S3C24XX chips

sound: ASoC: Fix an error path in neo1973_wm8753
sound: ASoC: Convert wm8753 to a new-style i2c driver
sound: ASoC: Convert neo1973/lm4857 to a new-style i2c driver

SoC Codec AD1980

ALSA: ASoC codec: fix compiling error in ad1980 driver after ASoC API changed

SoC Codec AK4535

ALSA: ASoC: Fix double free and memory leak in many codec drivers
ALSA: ASoC: Convert ak4535 to a new-style i2c driver
ALSA: ASoC: Don't suggest compile time selection of codec access

SoC Codec Philips UDA1380

ALSA: ASoC: Fix double free and memory leak in many codec drivers
ALSA: ASoC: Convert uda1380 to a new-style i2c driver
ALSA: ASoC: Don't suggest compile time selection of codec access

SoC Codec SSM2602

sound: ASoC codec: SSM2602 audio codec driver

SoC Codec TLV320AIC3X

ALSA: ASoC: Fix double free and memory leak in many codec drivers

SoC Codec WM8510

sound: ASoC: Convert wm8510 to a new-style i2c driver

SoC Codec WM8731

ALSA: ASoC: Convert wm8731 to a new-style i2c driver
ALSA: ASoC: Add SPI support for WM8731

SoC Codec WM8750

ALSA: ASoC: Convert wm8750 to a new-style i2c driver
ALSA: ASoC: Don't suggest compile time selection of codec access

SoC Codec WM8753

sound: ASoC: Convert wm8753 to a new-style i2c driver

SoC Codec WM8903

Add wm8903 build stub
ALSA: ASoC: Add WM8903 CODEC driver
ALSA: ASoC: WM8903 I2C cleanups

SoC Codec WM8990

ALSA: wm8990: Fix routing of left DAC to speaker mixer
ALSA: wm8990: Implement speaker volume PGA
ALSA: ASoC: Convert wm8990 to a new-style i2c driver

SoC Codec WM9713

ALSA: ASoC: Fix indentation in WM9713

SoC DaVinci

ALSA: ASoC: Add missing new-line at end of strings

SoC Dynamic Audio Power Management

soc - Add a workaround for incompatible debugfs
ALSA: ASoC: Convert DAPM pop time configuration to debugfs

SoC Layer

ALSA: ASoC: Add WM8903 CODEC driver
ALSA: ASoC: Make all codecs depend on rather than selecting I2C
ALSA: ASoC: Add SPI support for WM8731
sound: ASoC codec: SSM2602 audio codec driver
sound: ASoC: Blackfin: Include Blackfin architecture support in build
ALSA: ASoC: Sort Kconfig and Makefile entries for codecs
ALSA: ASoC: Use codec digital mute when stopping playback

SoC PXA2xx Spitz

ALSA: ASoC: Convert wm8750 to a new-style i2c driver

SoC Texas Instruments OMAP

ALSA: ASoC: Fix error paths in N810 machine driver init and release clocks at exit

USB

ALSA: us122l - Add missing dependency on CONFIG_SND_HWDEP

Utils

alsa-info.sh: fix --pastebin option
alsa-info.sh: Add output from 'amixer info'

VIA82xx driver

ALSA: via82xx - ignore buffer preallocation errors
ALSA: via82xx - check continuous pages in buffer setup

WSS library

ALSA: wss_lib: snd_wss_calibrate_mute improvement
ALSA: wss_lib: do not mess mixer settings during probe

au88x0 driver

ALSA: Clean up SG-buffer helper functions and macros

Changelog between 1.0.17 and 1.0.18rc3 releases

alsa-lib

Core

IEC958 definition for consumer status channel update
Release v1.0.17a
Release v1.0.18rc3

Mixer API

alsa-lib: Don't call the simple mixer interface "legacy"

PCM API

Add boolean (mute) functionality to softvol plugin
clamp snd_pcm_rewind()/snd_pcm_forward() into the right direction
fix type of internally used sframes variable, to avoid unnecessary casts
fix return value of snd_pcm_rewind()/snd_pcm_forward() to return how much actually has been rewound, instead of what actually could have rewound
Fix interleave check in pcm_direct.c
Fix non-tread mode with PCM direct plugins
alsa-lib: Make sure SND_PCM_NO_xxx flags don't get lost when nonblocking mode is enabled
alsa-lib: Reword doxygen doc for snd_pcm_delay()
Add snd_pcm_avail() and snd_pcm_avail_delay() functions. Make snd_pcm_hwsync() deprecated.
pcm_softvol plugin: remove access type change for refine

Detailed changelog between 1.0.18rc2 and 1.0.18rc3 releases

alsa-utils

Core

- Remove unneeded files from EXTRA_DIST in root Makefile.am
automake packages such filse automatically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Release v1.0.18rc3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ALSA Control (alsactl)

- Add missing files to be included in alsactl/Makefile.am
Signed-off-by: Takashi Iwai <tiwai@suse.de>

aplay/arecord

- aplay - show sample arrangement (mmap areas) in verbose more for debugging
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

alsa-plugins

Core

- Fix build with recent ffmpeg
Add a check of pkgconfig libavcodec in configure script.
Since the header file was moved to <libavcodec/*>, its location has
to be checked in configure, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Release v1.0.18rc3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

A52 Output plugin

- Fix build with recent ffmpeg
Add a check of pkgconfig libavcodec in configure script.
Since the header file was moved to <libavcodec/*>, its location has
to be checked in configure, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

PulseAudio -> ALSA plugin

- Reindent to Linux kernel style
The PulseAudio plugin was horrible mix-up of different indenting styles even in
the same files. Indenting by 4, by 2, by 8 characters, sometimes with tabs,
sometimes with spaces. To fix this chaos I ran all files through "indent -kr
-i8". Hopefully future patches will follow kernel style indenting too.
- Add Emacs-style /*-*- linux-c -*-*/ header comment
As suggested in the kernel's CodingStyle document force Emacs into kernel-style
indenting to prevent future indenting chaos, at least for Emacs users.
- Make pulse_new() a proper C function
C functions taking no argument need a (void) as argument list. It's C++ where
() is allowed too.
- Don't modify the SIGPIPE handler
Since quite a while (2 years or so) PulsAudio doesn't require SIGPIPE anymore
to be set to SIG_IGN. Since resetting this was incredibly ugly in the first
place it is now time to get rid of this.
- Call pa_context_disconnect() explicitly
Make sure the pa_context is disconnected when we unref it. This is also done
implicitly when the last reference to the pa_context is dropped, but that might
be later then expected in case someone else took a reference on our object. To
make sure that the pa_context will than not make any call into our code,
disconnect it explicitly.
- use SNDERR instead of fprintf to print error messages
We shouldn't spam on stderr unconditionally, so let's use ALSA subsystem for
printing errors.
- Support S32 sample types
PulseAudio has been supporting these for a while, so let's make use of them
here.
- Add trailing NUL character to snprintf output
snprintf doesn't necessarily append a trailing NUL to the strings it writes. So
let's do it explicitly.
- Get rid of pulse_poll_revents()
pulse_poll_revents() has no real purpose, so let's remove it.
- Add more error checking
Add a bit more error checking where necessary
- Remove fix for bug 0003470
This fix is completely broken (introduces new segfaults) and the assert is
tries to fix is superfluous anyway, so let's just remove this altogether.
- Rework hardware parameter selection
Follow more closely the real limits of PulseAudio's API.
- A bag of minor clean ups for ctl_pulse.c
Use more error checking where applicable, optimize a few things, use PulseAudio
API functions for comparing pa_cvolume structures.
- Make pulse_ext_callback const
We can make our instance of snd_ctl_ext_callback_t const without any problems,
so let's do it.
- Drop our own implementation of the poll() callbacks
The ctl extension interfaces support poll() via a single file descriptor
anyway, so let's use it instead of rolling our own poll support callbacks.
- A bag of minor clean-ups for pulse.c
Use more error checking where appropriate, optimize a few things.
- Split out O_NONBLOCK setting into seperate function
Since we set O_NONBLOCK more than once, make it a nice and correct function.
- Save a byte of memory
Since we only need a single char, we should allocate only a single char,
instead of a string of two bytes.
Woah! We're saving memory, baby!
- Adjust buffering metrics to match what PA internally uses
Follow more closely the buffering metrics PA uses internally. This avoids
clamping of the metricsin PA internally.
- Make sure we always have a sensible channel mapping
If PA doesn't know a default channel map for the specified number of channels,
synthesize one by extending a known one with a smaller channel number.
- Use PA_STREAM_EARLY_REQUESTS if available
Use PA_STREAM_EARLY_REQUESTS if PA knows it. PA >= 0.9.12 learned this flag
which will make the buffering more similar to traditional fragment/period-based
scheduling. Instead of delaying buffer fillup until the latest moment possible
this will cause fillups much earlier.
This is useful for compatibility for broken ALSA clients that don't care about
'readability' of the ALSA fd but instead schedule audio via usleep() or a
similar call. (One exampler being mplayer)
- Use S32/FLOAT32 only where available in the PA libs
S32 samples are a recent addition to the PA API, so check if they are available
before actually using them.
- Add const to our snd_pcm_ioplug_callback_t instances
The fields are constant so let's declare that they actually are.
- Don't implement our own poll handlers, we can use the default ones
As long as we have only a single fd to watch we can use the default poll
handler implementations. So let's do this.
- Remove our own poll handler implementation entirely
Neither pcm_pulse.c nor ctl_pulse.c is using our poll handler implementation
anymore, so let's get rid of the common parts previously used by them, too
- A bag of clean-ups for pcm_ctl.c
Use more error checking where appropriate, optimize a few things, split common
code into seperate functions.

libavcodec's resampler

- Fix build with recent ffmpeg
Add a check of pkgconfig libavcodec in configure script.
Since the header file was moved to <libavcodec/*>, its location has
to be checked in configure, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Detailed changelog between 1.0.18rc1 and 1.0.18rc3 releases

alsa-driver

Sound Core

- Fix include files in configure checks
Don't include from $CONFIG_SND_KERNELDIR directly in configure checks.
The header files can be in different places.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix ARCH detection for ppc and ppc64
arch/ppc is obsoleted on 2.6.27.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add arch/include check to configure
Added an include path to arch/include if exists (for 2.6.27 or later)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- soc - Add a workaround for incompatible debugfs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Release v1.0.18rc3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: make OSS sound core optional
sound/sound_core.c implements soundcore.ko and contains two parts -
sound_class which is shared by both ALSA and OSS and device
redirection support for OSS. It's always compiled when any sound
support is enabled although it's necessary only when OSS (the actual
one or emulation) is enabled. This is slightly wasteful and as device
redirection always registers character device region for major 14, it
prevents alternative implementation.
This patch introduces a new config SOUND_OSS_CORE which is selected
iff OSS support is actually necessary and build the OSS core part
conditionally.
If OSS is disabled, soundcore merely contains sound_class but leaving
it that way seems to be the simplest approach as otherwise sound_class
should be in ALSA core file if OSS is disabled but should be in
soundcore if OSS is enabled. Also, there's also the user confusion
factor.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- Fix CONFIG_AC97_BUS dependency
CONFIG_AC97_BUS is used from both sound and ucb1400 drivers.
The recent change in Kconfig introduced the exclusive dependency on
CONFIG_SOUND, and disabled the ucb1400 build without sound.
This patch makes CONFIG_AC97_BUS independent.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tested-by: Randy Dunlap <randy.dunlap@oracle.com>

ALSA Core

- Add patches for sparc/*.c
2.6.28 and later use the generic of_*().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: Clean up SG-buffer helper functions and macros
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: Allocate larger pages in sgbuf
Most hardwares have limited buffer-descriptor table length. This
also restricts the max buffer size of the sound driver.
For example, snd-hda-intel has 1MB buffer size limit, and this is
because it can have at most 256 BDL entries. For supporting larger
buffers, we need to allocate larger pages even for sg-buffers.
This patch changes the sgbuf allocation code to try to allocate
larger pages first. At each head of the allocated pages, the
number of allocated pages is stored in the lowest bits of the
corresponding entry of the table addr field. This change isn't
visible as long as the driver uses snd_sgbuf_get_addr() helper.
Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size().
This returns the size of the chunk on continuous pages starting at
the given position offset. If the chunk reaches to a non-continuous
page, it returns the size to the boundary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: make OSS sound core optional
sound/sound_core.c implements soundcore.ko and contains two parts -
sound_class which is shared by both ALSA and OSS and device
redirection support for OSS. It's always compiled when any sound
support is enabled although it's necessary only when OSS (the actual
one or emulation) is enabled. This is slightly wasteful and as device
redirection always registers character device region for major 14, it
prevents alternative implementation.
This patch introduces a new config SOUND_OSS_CORE which is selected
iff OSS support is actually necessary and build the OSS core part
conditionally.
If OSS is disabled, soundcore merely contains sound_class but leaving
it that way seems to be the simplest approach as otherwise sound_class
should be in ALSA core file if OSS is disabled but should be in
soundcore if OSS is enabled. Also, there's also the user confusion
factor.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC PXA2xx Core

- ALSA: ASoC: fix pxa2xx-i2s clk_get call
pxa2xx-i2s: probe actual device and use it for clk_get call
thus fixing error during startup hook
Signed-off-by: Dmitry Baryshkov <dbaryshkov@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Control Midlevel

- ALSA: use correct lock in snd_ctl_dev_disconnect()
The lock used in snd_ctl_dev_disconnect() should be card->ctl_files_rwlock
for protection of card->ctl_files entries, instead of card->controls_rwsem.
Reported-by: Vegard Nossum <vegard.nossum@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

PCM Midlevel

- Regenerated pcm_native.patch
For addition of get_unmapped_area fops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: Clean up SG-buffer helper functions and macros
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: Allocate larger pages in sgbuf
Most hardwares have limited buffer-descriptor table length. This
also restricts the max buffer size of the sound driver.
For example, snd-hda-intel has 1MB buffer size limit, and this is
because it can have at most 256 BDL entries. For supporting larger
buffers, we need to allocate larger pages even for sg-buffers.
This patch changes the sgbuf allocation code to try to allocate
larger pages first. At each head of the allocated pages, the
number of allocated pages is stored in the lowest bits of the
corresponding entry of the table addr field. This change isn't
visible as long as the driver uses snd_sgbuf_get_addr() helper.
Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size().
This returns the size of the chunk on continuous pages starting at
the given position offset. If the chunk reaches to a non-continuous
page, it returns the size to the boundary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: add dummy function to support shared mmap in nommu Blackfin arch
Cc: Mike Frysinger <vapier.adi@gmail.com>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

/soc/Makefile

- sound: ASoC: Blackfin: Include Blackfin architecture support in build
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

/soc/codecs/Makefile

- ALSA: ASoC: Add WM8903 CODEC driver
The WM8903 is a high performance ultra-low power stereo CODEC optimised
for portable audio applications. Features include:
* 5mW power consumption for DAC to headphone playback
* Stereo DAC SNR 96dB typical, THD -86dB typical
* Stereo ADC SNR 93dB typical, THD -80dB typical
* Up to 3 single ended inputs per stereo channel
* Up to 2 pseudo differential inputs per stereo channel
* Up to 1 fully differential mic input per stereo channel
* Digital Dynamic Range Controller (compressor/limiter)
* Digital sidetone mixing
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC codec: SSM2602 audio codec driver
[Some checkpatch fixups done by Mark Brown.]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Sort Kconfig and Makefile entries for codecs
Hopefully this will make merges a little bit easier.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

/sparc/Makefile

- Add patches for sparc/*.c
2.6.28 and later use the generic of_*().
Signed-off-by: Takashi Iwai <tiwai@suse.de>

AC97 Codec

- ALSA: ac97 - vt1618 7.1 Audio Support
1. Implement 7.1 Output for vt1618 codec.
2. Fix typos in comments from my previous 1617a patch.
Signed-off-by: John L. Utz III <john.utz@dmx.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ac97_patch make functions static
Only used in ac97_codec by including ac97_patch.c directly, effectively static
Found by sparse:
sound/pci/ac97/ac97_patch.c:3551:5: warning: symbol 'patch_vt1617a' was not declared. Should it be static?
sound/pci/ac97/ac97_patch.c:3767:5: warning: symbol 'patch_vt1618' was not declared. Should it be static?
Signed-off-by: Harvey Harrison <harvey.harrison@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

AD1816A driver

- ALSA: snd-ad1816a: add different TerraTec EWS64 S model
Signed-off-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

AD1848 driver

- ALSA: remove stale files
Empty files remained likely due to wrong patching.
Remove them now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ADI Blackfin ASoC

- Add soc/blackfin build stub
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- sound: ASoC: Blackfin: SPORT peripheral interface driver
SPORT is a serial port which can support serveral serial communication
protocols. It can be used as I2C/PCM/AC97. For further information,
please look up the HRM.
[Additional coding standards fixes by Mark Brown.]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: DMA Driver for AC97 sound chip
[Additional coding standards fixes by Mark Brown.]
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: AC97 Blackfin CPU DAI driver
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: DMA Driver for I2S sound chip
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: I2S CPU DAI driver
[Additional coding standards fixes by Mark Brown.]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: board driver for AD1980/1 audio codec
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: board driver for SSM2602 sound chip
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: add Blackfin arch ASoC Kconfig and Makefile
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

AK4XXX AD/DA converters

- ALSA: revo51: add headphone output
It is ALSA bug #1754.
D6 bit in 02 register is PW4.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ALS4000 driver

- ALSA: als4000 - Code clean up
- use specs-derived register name enums instead of open-coded numeric
values, for better understanding of things
- fix naming confusion ("gcr" was _NOT_ the GCR register stuff, but
instead the io _base_ which has multiplexed _access_ to GCR config
space, at _sub_ registers 0x08 and 0x0c)
- add FIXME comments about a race condition and MPU401 features
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ALS4000 driver work, step 2
- more register naming work
- finally figured out that weird CR register stuff
(and did I mention that I hate _really_ undecipherable open-coded values?)
- fix handling of IRQ sharing in interrupt handler
(hopefully properly, otherwise I'd be grateful to hear your
pedantic comments ;)
- add handy SPECS_PAGE references wherever useful
- comments, cleanup
- add me as module author
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ALSA Version

- ALSA: Release v1.0.18rc3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ARM AACI PL041 driver

- ALSA: aaci - Fix NULL test at error path
The original fix by Julien Brunel <brunel@diku.dk>.
aaci_init_card() returns a pointer with ERR_PTR(), but in aaci_init()
NULL is supposed at this error path.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ATIIXP driver

- ALSA: atiixp - Add PCI ID for SB600 (1002:4382)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

Apple Onboard Audio driver

- ALSA: aoa - Remove duplicate assignment of SNDRV_PCM_FMTBIT_S{16,24}_BE bits
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

Asihpi driver

- split public and private defs
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add control-specific query functions
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use new query functions
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add extra checks to control cache
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Update to hpi_internal.h. Minor cleanups
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- No longer used
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Rewrite PAD control string caching
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Simplify DSP crashed flag
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add ASI1723, log buffer properties. Tweak function params for SWIG.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Remove duplicate func prototypes (also in hpimsginit.h)
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add missing comma in list
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- add buffer size member to adapter struct
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- HPI Stream buffers allocated according to application request
HPI Stream buffers allocated according to application request. E.g. if HPI ioctl is only used for control, no buffers are needed. Buffers can be preallocated but default size is 0
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- style fixes
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- need hpimsginit.h
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Better error handling in tuner query. Need hpimsginit.h
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- checkpatch fixes
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- asihpi - Fix compile warnings regarding printf and size_t
 %z prefix is necessary for arguments with size_t.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix format in portable way.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Include adapter buffer alloc inside adapter mutex
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Add support for ASI50xx SSX (multichannel) mode.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- asihpi - Mem buffer alloc inside adapter mutex.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- asihpi - Avoid null pointer dereference
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- asihpi - Return from timer func if XRUN
Return from timer func if XRUN.
Avoids read/write of reset stream.
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- asihpi - Turn off verbose logging
Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

BT87x driver

- ALSA: Clean up SG-buffer helper functions and macros
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

CA0106 driver

- ALSA: CA0106 on MSI K8N Diamond PLUS Motherboard
Correct a previous patch for the ca0106 onboard the MSI K8N Diamond PLUS
motherboard. Confirmed to have Line/Mic/Aux working for input, and sound
output working as expected.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

CMI8788 (Oxygen) driver

- ALSA: oxygen: prevent muting of nonexistent AC97 controls
The Xonar DX does not have CD Capture controls, so we have to check that
a control actually exists before muting it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
- ALSA: oxygen: fix distorted output on AK4396-based cards
When changing the sample rate, the CMI8788's master clock output becomes
unstable for a short time. The AK4396 needs the master clock to do SPI
writes, so writing to an AK4396 control register directly after a sample
rate change will garble the value. In our case, this leads to the DACs
being misconfigured to I2S sample format, which results in a wrong
output level and horrible distortions on samples louder than -6 dB.
To fix this, we need to wait until the new master clock signal has
become stable before doing SPI writes.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
CC: <stable@kernel.org>

Documentation

- ALSA: hda - Add AD1882A codec support
Added the basic support of AD1882A codec chip.
It's almost compatible with AD1882, but with a digital mic and some
differences in connections.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add descriptions of missing model values
Added the descriptions of missing model values.
Also, fixed the model strings of a few new models to a standard style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add support of Toshiba RX1
Added the support of Toshiba RX1 laptop with ALC262 codec chip.
Related ALSA bug#3386:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3386
Signed-off-by: Hiroshi Miura <miurahr@acm.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: Documentation/sound/alsa/ALSA-Configuration.txt: fix typo
As noted by Gu Rui in http://bugzilla.kernel.org/show_bug.cgi?id=11444,
there is a typo in Documentation/sound/alsa/ALSA-Configuration.txt.
After checking the source (sound/pci/hda/hda_intel.c), the report
looks correct to me.
Signed-off-by: Michael Kerrisk <mtk.manpages@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

HDA Codec driver

- ALSA: hda - Fix capture source widgets on ALC codecs
On some Realtek codecs like ALC882 or ALC883, the capture source is
no mux but sum widget. We have to initialize all channels properly
for this type, otherwise noises may come in from the unused route.
The patch assures to mute unused routes, and unmute the currently
selected route.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Tested-by: Daniel J Blueman <daniel.blueman@gmail.com>
- ALSA: hda - Fix call of alc888_coef_init()
Using init_hook to call alc888_coef_init() is problematic for configurations
that already set another init_hook. Better to put it in alc_init() as is
(although it looks a bit hackish).
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: hda - Add definitions of HDMI-related verbs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - set up HDMI channels
Set up channel information for HDMI widgets. This will allow LPCM
with multiple channels supported on some HDMI devices.
TODO: It still doesn't check ELD and doesn't change PCM parameters
dynamically.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add AD1882A codec support
Added the basic support of AD1882A codec chip.
It's almost compatible with AD1882, but with a digital mic and some
differences in connections.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Create beep control on ALC269 codec
ALC269 codec has a beep, but it was not used, so far.
Create a beep control appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Fix ALC269 capture source
ALC269 capture source wasn't properly set up.
It's an independent MUX (0x23), not a source of ADC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - fix spaces in patch_realtek.c
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add support of HP DC7600
Fixed the support of HP DC7600 with ALC260, which was formerly handled
as model=hp-3013.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add support of Toshiba S06
Added the support of Toshiba S06 with ALC262 codec chip.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add support of Acer Aspire One
Added the support of Acer Aspire One with ALC269 codec chip.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add support of Quanta FL1
Added the support of Quanta FL1 with ALC269 code chip.
Also a bit space clean-ups.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - support more ALC888 devices
- Add Lenovo Sky support (17aa:101d)
- Add ASUS M90V support (1043:8317)
- Add ASUS Eee 1601 support (1043:835f)
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add more ALC662/663 codec support
- Add support for ECS (1019:9087)
- Delete G71V record from alc662_cfg_tbl[]
The PCI SubsystemID was the same with M70V
- Add more ASUS notebook support
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: Revert "ALSA: hda - Added model selection for iMac 24""
This reverts commit 3e0e469fa216ec70c93b1593821b759d19ee2e6b.
The patch introduced a wrong detection of other intel Macs with
ALC88* codec because they share the same PCI SSID (but have different
codec subsystem-IDs). See http://lkml.org/lkml/2008/8/24/143
Reported-and-tested-by: Guillaume Chazarain <guichaz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add descriptions of missing model values
Added the descriptions of missing model values.
Also, fixed the model strings of a few new models to a standard style.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Add support of Toshiba RX1
Added the support of Toshiba RX1 laptop with ALC262 codec chip.
Related ALSA bug#3386:
https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3386
Signed-off-by: Hiroshi Miura <miurahr@acm.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda: Distortion fix for dell_m6_core_init
Added the EQ distortion fix to the dell_m6_core_init.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: hda - Fix ALC663 auto-probe
Fix the wrong DAC assignment for NID 0x17 mono-pin on ALC663.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: hda - Add mic-boost controls to ALC662/663 auto configuration
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: hda - Add model for Toshiba L305 laptop
Added Subsystem IDs (0x1179, 0xff64) for the Toshiba Satellite L305
laptop, so it automatically uses the ALC268_TOSHIBA quirk.
Signed-off-by: Travis Place <wishie@wishie.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda: SPDIF mux controls
Dynamically create mux controls for SPDIF outs on certain IDT/Sigmatel codecs.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda: removed unneeded hp_nid references
Removed unneeded hp_nid references for 92hd73xx codec family.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda: slave digital out support
Added support for playing a stream on multiple digital outs. This is done
by defining codec->slave_dig_outs as array of hda_nid_t with a null-terminated entry to set the
slave SPDIF outs, in which the slave outs have cloned settings of the master out (e.g. dig_out_nid).
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda: digital slave support for IDT codecs
Added slave_dig_outs entries for several IDT codecs that have multiple
SPDIF outs, and enabled these SPDIF outs in several pin configs.
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda: SPDIF mux name change
Changed the mux naming scheme from "IEC9258 Mux" to "IEC958 Playback Source" to match
the coding style.
Signed-by-off: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - add missing slave_dig_outs for 92HD73bxx
Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: cleanup
* add extra parenthesis to make code more readable
* use kzalloc() for alloc+zero rather than kcalloc()
* ensure that AUTO_SEQ_* starts at 0
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: HP and CD pin connect config
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: Fix inversion of surround and side channels
In the current driver, there is a consistent mistake between the SURROUND and
the SIDE channels. This patch fixes it.
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: Add VT1708S and VT1702 support
The VT1702 and VT1708S codecs are new HDA codecs by VIA.
This patch adds support for them to the patch_via.c file for HDA
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: Mute on headphone plug-in
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: Independent DAC for headphone
This mode allows an output stream to have two substreams, one for the
speakers and one for the headphone. Each of the substreams has independent
PCM data and uses a different DAC.
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: HDA patch_via.c: Second S/PDIF (HDMI) support
The VT1702 and VT1708S have a second S/PDIF output which is used to
connect to a HDMI transmitter. This patch adds support for it.
Signed-off-by: Harald Welte <HaraldWelte@viatech.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

HDA Intel driver

- ALSA: hda_intel: enable snoop for nvidia HDA controller
Enable the snoop for nvidia hda controller to avoid data coherence issue.
Signed-off-by: Peer Chen <peerchen@gmail.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: hda - disable delayed period-ack with bdl_pos_adj=0
When bdl_pos_adj=0 is given, disable the position-check and the delayed
period update mechanism. Usually bdl_pos_adj=0 is set only for the
debugging purpose on really broken hardwares. It's better to disable
the extra complexity in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - check page continuity
Check the continuity of allocated pages to reduce the BDL size as much
as possible so that it can use more than 1MB buffers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - Fix VIA recording problem
Add a workaround for bad DMA-position reporting on VIA chipset.
Signed-off-by: Joseph Chan <josephchan@via.com.tw>
[modified and cleaned up by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: hda - allow probing of 4 codecs
Allow probing of 4 codecs on known good situations.
On some known bad situations, it should be avoided.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

HDA generic driver

- ALSA: hda - print a few HDMI information in proc
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ICE1712 driver

- ALSA: ice1724: enable MIDI on Terratec Aureon 7.1 Universe
This enables MIDI on Terratec Aureon 7.1 Universe cards specifically.
Apparently the other envy24ht-based Aureon cards do not have MIDI ports,
hence I added a Universe specific eeprom array.
The newer cards (Aureon 7.1 PCI / 5.1 PCI/Fun) have MIDI but use a CMI
DSP so this driver does not concern them at all.
Signed-off-by: Maximilian Rehkopf <otakon@gmx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1724: Enable MIDI on TerraTec PHASE 22 and PHASE 28
Even though MIDI was fixed on ice1724 chips a while ago, it wasn't yet
enabled for some cards as it didn't get enough testing. This was tested
with MIDI keyboard on PHASE 22 and with looping back output to input and
it works stable, so it's safe to enable it.
Besides this, there are some more minor fixes, not exactly user visible:
* added info about PHASE 28 (collected, as I don't have a card)
* added info about TS22PCI and new revisions of PHASE 22
* disable 192k on PHASE 22 as AK4524 I2S doesn't support it
* enable SPDIF reciever on PHASE 22
Signed-off-by: Vedran Miletic <rivanvx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1712/aureon.c: fix coding style
before:
total: 304 errors, 137 warnings, 2259 lines checked
after:
total: 0 errors, 121 warnings, 2284 lines checked
Compile tested, size is different because of include was changed,
but without that change md5sum is different because of cmp cx,dx/cmp
dx,cx swap and __LINE__ was changed in printk function.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1712/wtm: fix coding style
before:
total: 2 errors, 0 warnings, 20 lines checked
total: 49 errors, 2 warnings, 518 lines checked
after:
total: 0 errors, 0 warnings, 20 lines checked
total: 0 errors, 0 warnings, 518 lines checked
Compile tested, size and code are equal.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1712/ice1724: Coding style fixes part 1 (more coming up)
Inspired by Alexander Beregalov's patches for wtm and aureon.c,
I decided to run checkpatch on some more files. After some work
checkpatch.pl-0.23 --no-tree --file --strict <file> reports
0 errors, 0 warnings, 0 checks, n lines checked for:
phase.c
phase.h
juli.c (1 check about unused code, maybe we should comment it)
juli.h (no changes necessary)
In other files I have just fixed // comments and long lines along the
way (but not all of them), more coming up.
Signed-off-by: Vedran Miletic <rivanvx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1712.c: fix coding style
before:
total: 113 errors, 169 warnings, 2786 lines checked
total: 26 errors, 24 warnings, 504 lines checked
after:
total: 14 errors, 163 warnings, 2799 lines checked
total: 0 errors, 24 warnings, 504 lines checked
Compile tested only.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ICE1724 driver

- ALSA: revo51: add headphone output
It is ALSA bug #1754.
D6 bit in 02 register is PW4.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1724 - limit channels for multi-channel playback
Limit the available channels for multi-channel playback device to
the real number of channels. Until now, always up to 8 channels
are created, which are simply useless without the real outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1712/ice1724: Coding style fixes part 1 (more coming up)
Inspired by Alexander Beregalov's patches for wtm and aureon.c,
I decided to run checkpatch on some more files. After some work
checkpatch.pl-0.23 --no-tree --file --strict <file> reports
0 errors, 0 warnings, 0 checks, n lines checked for:
phase.c
phase.h
juli.c (1 check about unused code, maybe we should comment it)
juli.h (no changes necessary)
In other files I have just fixed // comments and long lines along the
way (but not all of them), more coming up.
Signed-off-by: Vedran Miletic <rivanvx@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ice1724.c: fix coding style
before:
total: 96 errors, 66 warnings, 2612 lines checked
after:
total: 11 errors, 64 warnings, 2624 lines checked
Compile tested only.
Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

ISA

- ALSA: remove obsolete *_FIRMARE_IN_KERNEL in isa/Kconfig
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

Intel8x0 driver

- ALSA: intel8x0: implement ac97_clock whitelist
The AC97 clock detection is not accurate in some cases. This patch adds
an initial whitelist for audio devices gathered from RedHat's bugzilla:
https://bugzilla.redhat.com/show_bug.cgi?id=441087
As a side effect, white-listing might speedup kernel booting (AC97 clock
measuring code is not activated).
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: intel8x0 - use snd_pci_quirk for clock list
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

MIPS AU1x00 driver

- removed unused #include <linux/version.h>'s
This patch lets the files using linux/version.h match the files that
#include it.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>

Memalloc module

- ALSA: Clean up SG-buffer helper functions and macros
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: Fix allocation size calculation in snd_dma_alloc_pages_fallback()
snd_dma_alloc_pages_fallback() always tries to reduce the size in a half,
but it's not good when the given size isn't a power-of-two.
Check it first then try to align.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: Allocate larger pages in sgbuf
Most hardwares have limited buffer-descriptor table length. This
also restricts the max buffer size of the sound driver.
For example, snd-hda-intel has 1MB buffer size limit, and this is
because it can have at most 256 BDL entries. For supporting larger
buffers, we need to allocate larger pages even for sg-buffers.
This patch changes the sgbuf allocation code to try to allocate
larger pages first. At each head of the allocated pages, the
number of allocated pages is stored in the lowest bits of the
corresponding entry of the table addr field. This change isn't
visible as long as the driver uses snd_sgbuf_get_addr() helper.
Also, the patch adds a new function, snd_pcm_sgbuf_get_chunk_size().
This returns the size of the chunk on continuous pages starting at
the given position offset. If the chunk reaches to a non-continuous
page, it returns the size to the boundary.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

OPL3SA2 driver

- ALSA: opl3sa2: drop snd_card pointer from the snd_opl3sa2 structure
Remove card pointer from the snd_opl3sa2
structure and break circular reference
snd_card->snd_opl3sa2->snd_card.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Acked-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: opl3sa2: replace nd_opl3sa2_info_xxx() with snd_wss_info_xxx()
Replace snd_opl3sa2_info_xxx() with snd_wss_info_xxx().
Drop check of card->private_data which cannot be NULL
if card is not NULL (spotted by Rene Herman).
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Reviewed-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

OSS device core

- sound: make OSS sound core optional
sound/sound_core.c implements soundcore.ko and contains two parts -
sound_class which is shared by both ALSA and OSS and device
redirection support for OSS. It's always compiled when any sound
support is enabled although it's necessary only when OSS (the actual
one or emulation) is enabled. This is slightly wasteful and as device
redirection always registers character device region for major 14, it
prevents alternative implementation.
This patch introduces a new config SOUND_OSS_CORE which is selected
iff OSS support is actually necessary and build the OSS core part
conditionally.
If OSS is disabled, soundcore merely contains sound_class but leaving
it that way seems to be the simplest approach as otherwise sound_class
should be in ALSA core file if OSS is disabled but should be in
soundcore if OSS is enabled. Also, there's also the user confusion
factor.
Signed-off-by: Tejun Heo <tj@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: fix section mismatch in the sound_core
The __exit cleanup_oss_soundcore() is called from
the __init init_soundcore(). This causes section mismatch
and breaks kernel's linking on sparc64.
Remove the __exit attribute from the cleanup_oss_soundcore().
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

PARISC Harmony driver

- ALSA: harmony - fix a typo
Fix a typo in the patch to remove snd_assert().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

PPC AWACS driver

- ALSA: snd-powermac: mixers for PowerMac G4 AGP
Add mixer controls for PowerMac G4 AGP (Screamer).
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: snd-powermac: HP detection for 1st iMac G3 SL
Correct headphone detection for 1st generation iMac G3 Slot-loading (Screamer).
Signed-off-by: Risto Suominen <Risto.Suominen@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SA11xx UDA1341 driver

- ALSA: fix section for sa11xx-uda1341 platform driver
Don't use __init but __devinit to define probe function. A pointer to
sa11xx_uda1341_probe is passed to the core via platform_driver_register
and so the function must not disappear after the module is loaded. Using
__init and having HOTPLUG=y and SND_SA11XX_UDA1341=m the following
probably oopses:
echo -n sa11xx_uda1341.1 > /sys/bus/platform/driver/sa11xx_uda1341/unbind
echo -n sa11xx_uda1341.1 > /sys/bus/platform/driver/sa11xx_uda1341/bind
Signed-off-by: Uwe Kleine-König <ukleinek@informatik.uni-freiburg.de>
Cc: Russell King <rmk+kernel@arm.linux.org.uk>
Cc: Brian Avery <b.avery@hp.com>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SB drivers

- ALSA: ALS4000 driver work, step 2
- more register naming work
- finally figured out that weird CR register stuff
(and did I mention that I hate _really_ undecipherable open-coded values?)
- fix handling of IRQ sharing in interrupt handler
(hopefully properly, otherwise I'd be grateful to hear your
pedantic comments ;)
- add handy SPECS_PAGE references wherever useful
- comments, cleanup
- add me as module author
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SPARC AMD7930 driver

- Add patches for sparc/*.c
2.6.28 and later use the generic of_*().
Signed-off-by: Takashi Iwai <tiwai@suse.de>

SoC Audio for the Atmel AT32 System-on-Chip

- sound: ASoC: Convert wm8510 to a new-style i2c driver
Convert the wm8510 codec driver to the new (standard) device
driver binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Geoffrey Wossum <gwossum@acm.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Audio for the Atmel AT91 System-on-Chip

- removed unused #include <linux/version.h>'s
This patch lets the files using linux/version.h match the files that
#include it.
Signed-off-by: Adrian Bunk <bunk@kernel.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
- ALSA: ASoC: Add missing new-line at end of strings
Two warning messages lack their trailing new-line, add it.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Convert wm8731 to a new-style i2c driver
Convert the wm8731 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Tested-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Audio for the Samsung S3C24XX chips

- sound: ASoC: Fix an error path in neo1973_wm8753
The error handling in neo1973_init is incorrect:
* If platform_device_add fails, we go on with the rest of the
initialization instead of bailing out. Things will break when the
module is removed (platform_device_unregister called on a device
that wasn't registered.)
* If i2c_add_driver fails, we return an error so the module will not
load, but we don't unregister neo1973_snd_device, so we are leaking
resources.
Add the missing error handling.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Tim Niemeyer <reddog@mastersword.de>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Convert wm8753 to a new-style i2c driver
Convert the wm8753 codec driver to the new (standard) i2c device
driver binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Ville Syrjala <syrjala@sci.fi>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Convert neo1973/lm4857 to a new-style i2c driver
Convert the lm4857 driver in neo1973_wm8753 to the new (standard)
i2c device driver binding model. I assumed that the LM4857 was always
on the same I2C bus as the WM8753 codec.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Tim Niemeyer <reddog@mastersword.de>
Cc: Graeme Gregory <graeme@openmoko.org>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec AD1980

- ALSA: ASoC codec: fix compiling error in ad1980 driver after ASoC API changed
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec AK4535

- ALSA: ASoC: Fix double free and memory leak in many codec drivers
Many SoC audio codec drivers have improper freeing of memory in error
paths.
* codec is allocated in the platform device probe function, but is not
freed there in case of error. Instead it is freed in the i2c device
probe function's error path. However the success or failure of both
functions is not linked, so this could result in a double free (if
the platform device is successfully probed, the i2c device probing
fails and then the platform driver is unregistered.)
* codec->private_data is allocated in many platform device probe
functions but not freed in their error paths.
This patch hopefully solves all these problems.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: ASoC: Convert ak4535 to a new-style i2c driver
Convert the ak4535 codec driver to the new (standard) device driver
binding model. After this change, AK4535 devices are no longer
discovered automatically and must instead be instantiated explicitly.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Don't suggest compile time selection of codec access
Currently the boiler plate code used by most ASoC codecs to provide a
placeholder for SPI access suggests making the selection of SPI a
compile time option which is suboptimal when trying to build kernels
supporting multiple systems. Change this template to suggest allowing
runtime selection instead.
Leave the drivers not yet converted to new style I2C access for now to
avoid collisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec Philips UDA1380

- ALSA: ASoC: Fix double free and memory leak in many codec drivers
Many SoC audio codec drivers have improper freeing of memory in error
paths.
* codec is allocated in the platform device probe function, but is not
freed there in case of error. Instead it is freed in the i2c device
probe function's error path. However the success or failure of both
functions is not linked, so this could result in a double free (if
the platform device is successfully probed, the i2c device probing
fails and then the platform driver is unregistered.)
* codec->private_data is allocated in many platform device probe
functions but not freed in their error paths.
This patch hopefully solves all these problems.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: ASoC: Convert uda1380 to a new-style i2c driver
Convert the uda1380 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Don't suggest compile time selection of codec access
Currently the boiler plate code used by most ASoC codecs to provide a
placeholder for SPI access suggests making the selection of SPI a
compile time option which is suboptimal when trying to build kernels
supporting multiple systems. Change this template to suggest allowing
runtime selection instead.
Leave the drivers not yet converted to new style I2C access for now to
avoid collisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec SSM2602

- sound: ASoC codec: SSM2602 audio codec driver
[Some checkpatch fixups done by Mark Brown.]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec TLV320AIC3X

- ALSA: ASoC: Fix double free and memory leak in many codec drivers
Many SoC audio codec drivers have improper freeing of memory in error
paths.
* codec is allocated in the platform device probe function, but is not
freed there in case of error. Instead it is freed in the i2c device
probe function's error path. However the success or failure of both
functions is not linked, so this could result in a double free (if
the platform device is successfully probed, the i2c device probing
fails and then the platform driver is unregistered.)
* codec->private_data is allocated in many platform device probe
functions but not freed in their error paths.
This patch hopefully solves all these problems.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

SoC Codec WM8510

- sound: ASoC: Convert wm8510 to a new-style i2c driver
Convert the wm8510 codec driver to the new (standard) device
driver binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Geoffrey Wossum <gwossum@acm.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec WM8731

- ALSA: ASoC: Convert wm8731 to a new-style i2c driver
Convert the wm8731 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Tested-by: Manuel Lauss <mano@roarinelk.homelinux.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Add SPI support for WM8731
[Modified to allow runtime selection between I2C and SPI and to select
SPI_MASTER for all codecs build so this is included. -- broonie]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec WM8750

- ALSA: ASoC: Convert wm8750 to a new-style i2c driver
Convert the wm8750 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Don't suggest compile time selection of codec access
Currently the boiler plate code used by most ASoC codecs to provide a
placeholder for SPI access suggests making the selection of SPI a
compile time option which is suboptimal when trying to build kernels
supporting multiple systems. Change this template to suggest allowing
runtime selection instead.
Leave the drivers not yet converted to new style I2C access for now to
avoid collisions.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec WM8753

- sound: ASoC: Convert wm8753 to a new-style i2c driver
Convert the wm8753 codec driver to the new (standard) i2c device
driver binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Cc: Ville Syrjala <syrjala@sci.fi>
Cc: Frank Mandarino <fmandarino@endrelia.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec WM8903

- Add wm8903 build stub
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: ASoC: Add WM8903 CODEC driver
The WM8903 is a high performance ultra-low power stereo CODEC optimised
for portable audio applications. Features include:
* 5mW power consumption for DAC to headphone playback
* Stereo DAC SNR 96dB typical, THD -86dB typical
* Stereo ADC SNR 93dB typical, THD -80dB typical
* Up to 3 single ended inputs per stereo channel
* Up to 2 pseudo differential inputs per stereo channel
* Up to 1 fully differential mic input per stereo channel
* Digital Dynamic Range Controller (compressor/limiter)
* Digital sidetone mixing
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: WM8903 I2C cleanups
Fix a few issues identified by Jean Delvare:
- Add missing newline to error display.
- Return an error when we fail to locate the I2C adaptor.
- Remove the static wm8903_i2c_device Jean suggested, it's redundant.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec WM8990

- ALSA: wm8990: Fix routing of left DAC to speaker mixer
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: wm8990: Implement speaker volume PGA
The latest revisions of the WM8990 provide a programmable gain amplifier
for the speaker - configure the register cache and implement controls
for this. Older revisions of the device ignore writes to these controls.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: ASoC: Convert wm8990 to a new-style i2c driver
Convert the wm8990 codec driver to the new (standard) device driver
binding model. After this change, WM8990 devices are no longer
discovered automatically and must instead be instantiated explicitly.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Codec WM9713

- ALSA: ASoC: Fix indentation in WM9713
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC DaVinci

- ALSA: ASoC: Add missing new-line at end of strings
Two warning messages lack their trailing new-line, add it.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Dynamic Audio Power Management

- soc - Add a workaround for incompatible debugfs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- ALSA: ASoC: Convert DAPM pop time configuration to debugfs
Pop time configuration is a debugging feature which shouldn't be used in
normal operation so move it to debugfs where it is clearer. This also
simplifies the code since debugfs provides standard facilities for
exposing simple integer values to user space.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Layer

- ALSA: ASoC: Add WM8903 CODEC driver
The WM8903 is a high performance ultra-low power stereo CODEC optimised
for portable audio applications. Features include:
* 5mW power consumption for DAC to headphone playback
* Stereo DAC SNR 96dB typical, THD -86dB typical
* Stereo ADC SNR 93dB typical, THD -80dB typical
* Up to 3 single ended inputs per stereo channel
* Up to 2 pseudo differential inputs per stereo channel
* Up to 1 fully differential mic input per stereo channel
* Digital Dynamic Range Controller (compressor/limiter)
* Digital sidetone mixing
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Make all codecs depend on rather than selecting I2C
The I2C layer uses I/O operations that aren't available on all
architectures and since select bypasses Kconfig dependency checking
selecting I2C breaks builds like allmodconfig on some architectures.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Add SPI support for WM8731
[Modified to allow runtime selection between I2C and SPI and to select
SPI_MASTER for all codecs build so this is included. -- broonie]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC codec: SSM2602 audio codec driver
[Some checkpatch fixups done by Mark Brown.]
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- sound: ASoC: Blackfin: Include Blackfin architecture support in build
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Sort Kconfig and Makefile entries for codecs
Hopefully this will make merges a little bit easier.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: ASoC: Use codec digital mute when stopping playback
Muting the DAC masks artefacts introduced as the digital stream shuts
down, for example when the input stops being clocked.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC PXA2xx Spitz

- ALSA: ASoC: Convert wm8750 to a new-style i2c driver
Convert the wm8750 codec driver to the new (standard) device driver
binding model.
Signed-off-by: Jean Delvare <khali@linux-fr.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

SoC Texas Instruments OMAP

- ALSA: ASoC: Fix error paths in N810 machine driver init and release clocks at exit
Thanks to Felipe Balbi <felipe.balbi@nokia.com> by noticing that if clk_get
to sys_clkout2_src fails, then n810_snd_device is never released.
Add also sys_clkout2_src release into error path, error code return and
release the clocks at exit.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

USB

- ALSA: us122l - Add missing dependency on CONFIG_SND_HWDEP
CONFIG_SND_USB_US122L uses snd_hwdep_new(), so SND_HWDEP needs
to be enabled (selected).
ERROR: "snd_hwdep_new" [sound/usb/usx2y/snd-usb-us122l.ko] undefined!
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

Utils

- alsa-info.sh: fix --pastebin option
Fixed the output uploaded to pastebin.ca (when using --pastebin)
Signed-Off-By: Travis Place <wishie@wishie.net>
- alsa-info.sh: Add output from 'amixer info'
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

VIA82xx driver

- ALSA: via82xx - ignore buffer preallocation errors
The errors at buffer preallocations aren't fatal and safe to ignore.
The buffer will be allocated dynamically when opened.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: via82xx - check continuous pages in buffer setup
Check continuous pages in the buffer set up so that the number of
BDL is reduced. Also increased the max buffer size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

WSS library

- ALSA: wss_lib: snd_wss_calibrate_mute improvement
Mute sound by setting mute bit without
setting volume to 0. It makes both source code
and binary shorter.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- ALSA: wss_lib: do not mess mixer settings during probe
Use the wss_dout function which does not mess
shadowed register values during chip probing.
Otherwise, user ends up with stupid mixer settings
after driver loading.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Acked-by: Rene Herman <rene.herman@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

au88x0 driver

- ALSA: Clean up SG-buffer helper functions and macros
Clean up SG-buffer helper functions and macros. Helpers take substream
as arguments now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

Detailed changelog between 1.0.17 and 1.0.18rc3 releases

alsa-lib

Core

- IEC958 definition for consumer status channel update
Updated IEC958 consumer status channel definitions according
to the third edition of IEC60958-3 spec.
Signed-off-by: Pawel Moll <pawel.moll@st.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- Release v1.0.17a
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- Release v1.0.18rc3
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

Mixer API

- alsa-lib: Don't call the simple mixer interface "legacy"
The simpler mixer interface is not legacy, so the header file shouldn't claim
otherwise.
Fixes bug#3569
From: Lennart Poettering <mznyfn@0pointer.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>

PCM API

- Add boolean (mute) functionality to softvol plugin
When the resolution is set to 2, a boolean control is created as a
mute switch instead of a volume control.
Also, fixed the possible zero-division error.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- clamp snd_pcm_rewind()/snd_pcm_forward() into the right direction
The clamping of the input parameter in snd_pcm_rewind()/_forward() is
in the wrong direction.
Signed-off-by: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- fix type of internally used sframes variable, to avoid unnecessary casts
This minor patch fixes the type of the sframes variable in
snd_pcm_plugin_forward(). With this fix we need to cast less and the
code is less confusing.
Signed-off-by: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- fix return value of snd_pcm_rewind()/snd_pcm_forward() to return how much actually has been rewound, instead of what actually could have rewound
Make snd_pcm_plugin_rewind()/_forward() actually return how much has
been rewound/forwarded instead of how much could have been
rewounded/forwarded. This makes the code actually do what the
documentation of snd_pcm_rewind() suggests.
Signed-off-by: Lennart Poettering <lennart@poettering.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix interleave check in pcm_direct.c
Fix the check of interleaved format. It checked a totally bogus value
as PCM format.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- Fix non-tread mode with PCM direct plugins
tread=0 wasn't set properly for very old ALSA drivers with no timer
tread support.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- alsa-lib: Make sure SND_PCM_NO_xxx flags don't get lost when nonblocking mode is enabled
The plug PCM copies the 'mode' field from the slave PCM. If blocking mode is
enabled for the plug PCM the mode is subsequently overwritten with the original
requested 'mode'. If non-blocking mode is requested this does not happen.
Because the hw PCM synthesizes the 'mode' from the actual file descriptor flags
no SND_PCM_NO_xxx will ever be set for it. This has the effect that the 'mode'
of the plug PCM will also not include those flags anymore -- unless they are
overwritten as mentioned above. This basically means SND_PCM_NO_xxx is ignored
for plug:hw:4711 style device strings opened in non-blocking mode.
You can easily test this with "aplay --channels 7 --disable-channels -f S16_LE
-r 44100 -D plug:hw:0" on a device that cannot do 7 channels. Normally this
call should fail, however if you add "-N" to the command line this call will
succeed.
This patch simply copies the SND_PCM_NO_xxx flags back into the 'mode' field in
case we don't overwrite it with the original anyway.
Probably closes bug 3571 for good.
From: Lennart Poettering <mznyfn@0pointer.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- alsa-lib: Reword doxygen doc for snd_pcm_delay()
snd_pcm_delay() is for synchronization purposes, it returns the overall latency
of the stream, not just the latency induced by the hardware playback buffer.
The documentation is a bit misleading about this fact, and some people already
misunderstood it. So let's reword this to make the explanation clearer and
explicit.
This is a result of the discussions of the thread this mail belongs to:
http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008456.html
From: Lennart Poettering <mznyfn@0pointer.de>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- Add snd_pcm_avail() and snd_pcm_avail_delay() functions. Make snd_pcm_hwsync() deprecated.
As proposed in http://mailman.alsa-project.org/pipermail/alsa-devel/2008-June/008558.html
the snd_pcm_avail() and snd_pcm_avail_delay() functions are now available
to get accurate stream position in a straight way. The snd_pcm_avail_delay()
function was added to ensure full sync between avail and delay values. It's
actually implemented using delay() + avail_update() calls but it might be
changed in future.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
- pcm_softvol plugin: remove access type change for refine
The softvol plugin does in-place sample recalculation. The access type
cannot be changed like in linear plugins. Remove access type change
in refine functions.
This bug was detected with PulseAudio on Sony PS3 platform.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
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