Commit Graph

1688 Commits

Author SHA1 Message Date
Richard Purdie
abadfc928a [ALSA] ASoC codecs: WM8750 support
This patch adds ASoC support for the WM8750 codec.
Supported features:-
o Capture/Playback/Sidetone/Bypass.
o 16 & 24 bit audio.
o 8k - 96k sample rates.
o DAPM.

Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:22 +01:00
Richard Purdie
40e0aa6466 [ALSA] ASoC codecs: WM8731 support
This patch adds ASoC support for the WM8731 codec.
Supported features:-
 o Capture/Playback/Sidetone/Bypass.
 o 16 & 24 bit audio.
 o 8k - 96k sample rates.
 o DAPM.

Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:21 +01:00
Liam Girdwood
a3288176de [ALSA] ASoC: Build files
This patch adds support for building the ASoC core and the dynamic audio
power management support.

Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:19 +01:00
Richard Purdie
2b97eabc09 [ALSA] ASoC: dynamic audio power management (DAPM)
This patch adds Dynamic Audio Power Management (DAPM) to ASoC.
Dynamic Audio Power Management (DAPM) is designed to allow portable and
handheld Linux devices to use the minimum amount of power within the
audio subsystem at all times. It is independent of other kernel PM and
as such, can easily co-exist with the other PM systems.
DAPM is also completely transparent to all user space applications as
all power switching is done within the ASoC core. No code changes or
recompiling are required for user space applications. DAPM makes power
switching decisions based upon any audio stream (capture/playback)
activity and audio mixer settings within the device.
DAPM spans the whole machine. It covers power control within the entire
audio subsystem, this includes internal codec power blocks and machine
level power systems.
There are 4 power domains within DAPM:-
1. Codec domain - VREF, VMID (core codec and audio power)
Usually controlled at codec probe/remove and suspend/resume, although
can be set at stream time if power is not needed for sidetone, etc.
2. Platform/Machine domain - physically connected inputs and outputs
Is platform/machine and user action specific, is configured by the
machine driver and responds to asynchronous events e.g when HP are
inserted
3. Path domain - audio subsystem signal paths
Automatically set when mixer and mux settings are changed by the user.
e.g. alsamixer, amixer.
4. Stream domain - DAC's and ADC's.
Enabled and disabled when stream playback/capture is started and stopped
respectively. e.g. aplay, arecord.
All DAPM power switching decisions are made automatically by consulting
an audio routing map of the whole machine. This map is specific to each
machine and consists of the interconnections between every audio
component (including internal codec components).

Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:18 +01:00
Frank Mandarino
db2a416556 [ALSA] ASoC: core code
This patch is the core of ASoC functionality.
The ASoC core is designed to provide the following features :-
o  Codec independence. Allows reuse of codec drivers on other platforms
and machines.
o Platform driver code reuse. Reuse of platform specific audio DMA and
DAI drivers on different machines.
o Easy I2S/PCM digital audio interface configuration between codec and
SoC. Each SoC interface and codec registers their audio interface
capabilities with the core at initialisation. The capabilities are
subsequently matched and configured at run time for best power and
performance when the application hw params are known.
 o Machine specific controls/operations: Allow machines to add controls
and operations to the audio subsystem. e.g. volume control for speaker
amp.
To achieve all this, ASoC splits an embedded audio system into 3
components :-
1. Codec driver: The codec driver is platform independent and contains
audio controls, audio interface capabilities, codec dapm and codec IO
functions.
2. Platform driver: The platform driver contains the audio dma engine
and audio interface drivers (e.g. I2S, AC97, PCM) for that platform.
3. Machine driver: The machine driver handles any machine specific
controls and audio events. i.e. turning on an amp at start of playback.

Signed-off-by: Frank Mandarino <fmandarino@endrelia.com>
Signed-off-by: Richard Purdie <rpurdie@rpsys.net>
Signed-off-by: Liam Girdwood <liam.Girdwood@wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:16 +01:00
Takashi Iwai
3388c37e04 [ALSA] intel8x0 - Use pci_iomap
Use pci_iomap and ioread*/iowrite*() functions for accessing
hardwares.  pci_iomap is suitable for hardwares like ICH and
compatible that have both PIO and MMIO.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:14 +01:00
Jaroslav Kysela
c7132aeb72 [ALSA] pcm core: add prealloc_max file to substream directory to show maximum DMA size
Users ask us many times about the maximum DMA size for PCM devices. This
file gives them a hint in KB.

Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:13 +01:00
Jaroslav Kysela
b66b3cfe6c [ALSA] hda_intel: increase maximum DMA buffer size to 1024MB
See ALSA bug#2481 .

Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:12 +01:00
Takashi Iwai
bbb53551e3 [ALSA] emu10k1 - Fix compile warning
Fixed a compile warning regarding print format for size_t.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:11 +01:00
Johannes Berg
9244b2c307 [ALSA] alsa core: convert to list_for_each_entry*
This patch converts most uses of list_for_each to list_for_each_entry all
across alsa. In some place apparently an item can be on a list with
different pointers so of course that isn't compatible with list_for_each, I
therefore didn't touch those places.

Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:10 +01:00
Johannes Berg
d595ee7e01 [ALSA] aoa: fix up i2sbus_attach_codec
This patch changes i2sbus_attach_codec to implement a proper error handling
strategy using labels to jump to the right part. Since it has an elaborate
set-up sequence it also needs that tear-down, which I had hard-coded
inbetween all the checks. This increases readability and should reduce .text
size as well.

Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:09 +01:00
Johannes Berg
73e85fe845 [ALSA] aoa: set device pointer in pcms
This patch makes a few whitespace cleanups and makes i2sbus assign the new
struct device pointer in struct snd_pcm so that the proper device symlink
shows up in sysfs.

Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:08 +01:00
Johannes Berg
c78085fcd2 [ALSA] alsa core: add struct device pointer to struct snd_pcm
This patch adds a struct device pointer to struct snd_pcm in order to be
able to give it a different device than the card. It defaults to the card's
device, however, so it should behave identically for drivers not touching
the field.

Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:07 +01:00
Johannes Berg
12b131c4cf [ALSA] allow registering an alsa device with struct device pointer
This patch adds snd_register_device_for_dev taking a struct device
pointer to link the new device to and makes snd_register_device a simple
static inline wrapper around it.

Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:06 +01:00
Jochen Voss
feaa6a74d8 [ALSA] Enable the analog loopback of the Revolution 5.1
Enable the analog loopback of the Revolution 5.1 card.
This patch adds support for the PT2258 volume controller and modifies
the Revolution 5.1 driver to make use of this facility.  This allows
to control the analog loopback of the card.

Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:05 +01:00
Jochen Voss
a58e7cb16d [ALSA] Enable capture from line-in and CD on Revolution 5.1
Enable capture from line-in and CD on the Revolution 5.1 card.
This patch adds support for switching between the 5 input channels of
the AK5365 ADC and modifies the Revolution 5.1 driver to make use of
this facility.  Previously the capture channel was fixed to channel 0
(microphone on the Revolution 5.1 card).

Signed-off-by: Jochen Voss <voss@seehuhn.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:03 +01:00
Clemens Ladisch
e4f8e656d8 [ALSA] usb-audio: allow pausing
Add pause capabilities for both USB playback and capture streams.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:02 +01:00
Clemens Ladisch
1700f3080d [ALSA] usb-audio: merge playback/capture hardware information structs
The hardware information structures for playback and capture streams,
respectively, are the same, so we can use just one structure for both
streams.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 09:00:00 +01:00
James Courtier-Dutton
9f4bd5dde8 [ALSA] snd-emu10k1: Added support for emu1010, including E-Mu 1212m and E-Mu 1820m
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-09 08:59:59 +01:00
Takashi Iwai
7d2aae1e88 [PATCH] ALSA: Fix sysfs breakage
The recent change for a new sysfs tree with card* object breaks the
/sys/class/sound tree if CONFIG_SYSFS_DEPRECATED is enabled.
The device in each entry doesn't point the correct device object:

  /sys/class/sound
  ...
  |-- pcmC0D0c
  |   |-- dev
  |   |-- device -> ../../../class/sound/card0
  |   |-- pcm_class
  |   |-- power
  |   |   `-- wakeup
  |   |-- subsystem -> ../../../class/sound
  |   `-- uevent

Also, this change breaks some drivers (like sound/arm/*) referring
card->dev directly to obtain the device object for memory handling.

This patch reverts the semantics of card->dev to the former version,
which points to a real device object.  The card* object is stored in a
new card->card_dev field, instead.  The device parent is chosen either
card->dev or card->card_dev according to CONFIG_SYSFS_DEPRECATED to
keep the tree compatibility.
Also, card* isn't created if CONFIG_SYSFS_DEPRECATED is enabled.  The
reason of card* object is a root of all beloing devices, and it makes
little sense if each sound device points to the real device object
directly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Monty Montgomery <xiphmont@gmail.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2007-01-29 14:32:53 -08:00
Karsten Wiese
bc6191b100 [ALSA] Repair snd-usb-usx2y over OHCI
The previous patch 'Repair snd-usb-usx2y for usb 2.6.18' assumed
urb->start_frame roll over beyond MAX_INT for both UHCI & OHCI.
This isn't true until now (kernel 2.6.20).
Fix this by only looking at the common between OHCI & UHCI Frame number
range.
This is for mainline and stable kernels >= 2.6.18.

Signed-off-by: Karsten Wiese <fzu@wemgehoertderstaat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-23 09:23:36 +01:00
Takashi Iwai
2a2a5dddec [ALSA] usbaudio - Fix kobject_add() error at reconnection
Fixed the error from kobject_add() at reconnection the usb audio device.
This happens when an app keeps opening a device while the device is
replugged, due to the confliction of the internal bookkept index and
the really empty slot.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:14:15 +01:00
Mariusz Kozlowski
7fbe3ca571 [ALSA] usb: usbmixer error path fix
Without the patch below namelist[0] will not be freed in case
of kmalloc error.

Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:19 +01:00
Timofei V. Bondarenko
8c67071422 [ALSA] _snd_cmipci_uswitch_put doesn't set zero flags
Playing with spdif output on cmipci i've noticed the SPDO5V option does
not change appropriate bits the register.
The _snd_cmipci_uswitch_put checks the change in flags in wrong way.
If 'active' state of an option corresponds to a _zero_ bits in a hw
register then function fails. The SPDO5V is the sample.
In the most cases 'active' state of option is set through an non-zerio
bits in a register. This case works fine.
The fix attached.
Unfortunately i was unable to change spdif output voltage anyway.
Although the register changes right at least.
From: Timofei V. Bondarenko <tim@ipi.ac.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:18 +01:00
Takashi Iwai
6afeb11de5 [ALSA] hda-codec - Fix NULL dereference in generic hda code
Fix NULL dereference in hda_generic.c.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:17 +01:00
Jason Gaston
f9cc8a8b18 [ALSA] hda_intel: ALSA HD Audio patch for Intel ICH9
This patch adds the Intel ICH9 HD Audio controller DID's for ALSA.

Signed-off-by: Jason Gaston <jason.d.gaston@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:16 +01:00
Clemens Ladisch
987411b7d8 [ALSA] usb-audio: work around wrong frequency in CM6501 descriptors
The C-Media CM6501 chip's descriptors say that altsetting 5 supports
48 kHz, but it actually plays at 96 kHz.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:15 +01:00
Giuliano Pochini
5929546a96 [ALSA] Fix potential NULL pointer dereference in echoaudio midi
Fix races between the timer handler and the close function.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:13 +01:00
Peer Chen
5b005a0126 [ALSA] Audio: Add nvidia HD Audio controllers of MCP67 support to hda_intel.c
Add the support for HD audio controllers of MCP51,MCP55,MCP61,MCP65 & MCP67.

Signed-off-by: Peer Chen <pchen@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-01-09 09:06:12 +01:00
David S. Miller
f9af1d9dea [SOUND] Sparc CS4231: Use 64 for period_bytes_min
This matches what the ISA cs4231 driver uses.

Tested by Georg Chini.

Signed-off-by: David S. Miller <davem@davemloft.net>
2007-01-03 18:51:54 -08:00
Georg Chini
d35a1b9e10 [SOUND] Sparc CS4231: Fix IRQ return value and initialization.
SBUS: Change IRQ-handler return value from 0 to IRQ_HANDLED and
fix some initialisation problems.

Change period_bytes_min from 4096 to 256 to allow driver to work with
low latency (VOIP) applications. Hope this does not break EBUS.

Signed-off-by: Georg Chini <georg.chini@triaton-webhosting.com>
Signed-off-by: David S. Miller <davem@davemloft.net>
2007-01-03 18:37:26 -08:00
Leonard Norrgård
e22a9a8b70 [PATCH] sound: hda: detect ALC883 on MSI K9A Platinum motherboards (MS-7280)
Recognize the Realtek ALC883 chip on MSI K9A Platinum motherboards
(model no. MS-7280), enabling full sound capabilities.

Signed-off-by: Leonard Norrgård <leonard.norrgard@refactor.fi>
Signed-off-by: Linus Torvalds <torvalds@osdl.org>
2007-01-01 11:16:32 -08:00
James Courtier-Dutton
f8cb2c450e [ALSA] ac97: Identify CMI9761 chips.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:56:11 +01:00
James C Georgas
e8bb036a46 [ALSA] ac97_codec - trivial fix for bit update functions
This patch fixes a couple of bit update functions in
alsa-kernel/pci/ac97/ac97_codec.c, which could possibly corrupt bits not
in the given mask.
Specifically, it'll clobber unset bits in the target that are not in the
mask, when the corresponding bit in the given new value is set.

Signed-off-by: James C Georgas <jgeorgas@rogers.com>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:56:10 +01:00
James Courtier-Dutton
d5f6a38d98 [ALSA] snd-ca0106: Fix typos.
Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:56:09 +01:00
James Courtier-Dutton
e4f55d8010 [ALSA] snd-ca0106: Add new card variant.
Fixed ALSA bug#2326

Signed-off-by: James Courtier-Dutton <James@superbug.co.uk>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:56:08 +01:00
Akinobu Mita
4d36128592 [ALSA] sound: fix PCM substream list
If snd_pcm_new_stream() fails to initalize a substream (if
snd_pcm_substream_proc_init() returns error), snd_pcm_new_stream()
immediately return without unlinking that kfree()d substram.
It causes oops when snd_pcm_free() iterates the list of substream to
free them by invalid reference.

Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:56:00 +01:00
Akinobu Mita
c13893d7be [ALSA] sound: initialize rawmidi substream list
If snd_rawmidi_new() failed to allocate substreams for input
(snd_rawmidi_alloc_substreams() failed to populate a
&rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT]), it will try to
free rawmidi instance by snd_rawmidi_free().
But it will cause oops because snd_rawmidi_free() tries to free
both of substreams list but list for output
(&rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT]) is not initialized yet.

Signed-off-by: Akinobu Mita <akinobu.mita@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:59 +01:00
Nickolay V. Shmyrev
a48bb28c16 [ALSA] snd_hda_intel 3stack mode for ASUS P5P-L2
I have ASUS P5PL2 motherboard and it's embedded sound card requires
the following patch which sets '3stack' model to operate properly:
00:1b.0 0403: 8086:27d8 (rev 01)
        Subsystem: 1043:817f
        Flags: bus master, fast devsel, latency 0, IRQ 177
        Memory at dfdf8000 (64-bit, non-prefetchable) [size=16K]
        Capabilities: [50] Power Management version 2
        Capabilities: [60] Message Signalled Interrupts: 64bit+
Queue=0/0 Enable-
        Capabilities: [70] Express Unknown type IRQ 0

Signed-off-by: Nickolay V. Shmyrev <nshmyrev@yandex.ru>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:58 +01:00
Takashi Iwai
437a5a4606 [ALSA] Remove IRQF_DISABLED for shared PCI irqs
Fix IRQ flags for PCI devices.
The shared IRQs for PCI devices shouldn't be allocated with
IRQF_DISABLED.  Also, when MSI is enabled, IRQF_SHARED shouldn't
be used.
The patch removes unnecessary cast in request_irq and free_irq,
too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:57 +01:00
Takashi Iwai
01f681da49 [ALSA] Fix invalid assignment of PCI revision
Fix the type of PCI revision to char from int and avoid invalid
assignment with pointer cast.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:55 +01:00
Takashi Iwai
e3a5d59a17 [ALSA] Fix races in PCM OSS emulation
Fixed the race among multiple threads accessing the OSS PCM
instance concurrently by simply introducing a mutex for protecting
a setup of the PCM.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:54 +01:00
Christian Hesse
ba8bdf8584 [ALSA] hda-codec - fix typo in PCI IDs
my notebook is a Samsung X11 of course... The attached patch fixes the typo.

Signed-off-by: Christian Hesse <mail@earthworm.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:52 +01:00
Takashi Iwai
c19bcdc64a [ALSA] ac97 - Fix potential negative array index
Fix the case cidx2 >= 0 and cidx2 < 0 which may result in negative
array index.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:50 +01:00
Takashi Iwai
b90d7760ea [ALSA] hda-codec - Verbose proc output for PCM parameters
Make the output for PCM parameters more verbose, showing
each rate, bits and format.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:49 +01:00
Takashi Iwai
ee3527b036 [ALSA] hda-codec - Fix detection of supported sample rates
Don't include 9.6kHz in the list of supported sample rates.
Since this rate isn't indicated in AC_PAR_PCM parameter,
the driver might guess wrongly as if it's available.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:48 +01:00
Takashi Iwai
4a95cd8660 [ALSA] hda-codec - Fix model for ASUS V1j laptop
Add a proper model entry (laptop-eapd) for ASUS V1j laptop with
AD1986A codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:47 +01:00
Adrian Bunk
0e5d720ced [ALSA] sound/core/control.c: remove dead code
This patch removes some obviously dead code spotted by the Coverity
checker.

Signed-off-by: Adrian Bunk <bunk@stusta.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:46 +01:00
Takashi Iwai
efeccac5b7 [ALSA] hda-codec - Add model for HP q965
Added a model entry (HP_BPC) for HP q965 with ALC262 codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:45 +01:00
Clemens Ladisch
9a826ddba6 [ALSA] pcm core: fix silence_start calculations
The case where silence_size < boundary was broken because different
parts of the snd_pcm_playback_silence() function disagreed about whether
silence_start should point to the start or to the end of the buffer part
to be silenced.
This patch changes the code to always use to the start, which also
simplifies several calculations.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-20 08:55:43 +01:00