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Commit Graph

18320 Commits

Author SHA1 Message Date
Takashi Iwai
927cdab3b6 ASoC: Core fixes for v3.15
A few things here:
 
  - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
    have audio paths which shouldn't be present causing spurious powerups
    and potential audible issues for users.
  - Ensure the suspend->off transition doesn't have spurious transitions
    to prepare added to the sequence.
  - Fix incorrect skipping of PCM suspension for active audio streams.
  - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
    this and Timur no longer has the boards that he was using.
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Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Core fixes for v3.15

A few things here:

 - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to
   have audio paths which shouldn't be present causing spurious powerups
   and potential audible issues for users.
 - Ensure the suspend->off transition doesn't have spurious transitions
   to prepare added to the sequence.
 - Fix incorrect skipping of PCM suspension for active audio streams.
 - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing
   this and Timur no longer has the boards that he was using.
2014-05-14 14:23:48 +02:00
Mark Brown
cf86197ec5 Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus 2014-05-14 12:52:41 +01:00
Mengdong Lin
7189eb9b8f ALSA: hda - mask buggy stream DMA0 for Broadwell display controller
Broadwell display controller has 3 stream DMA engines. DMA0 cannot update DMA
postion buffer properly while DMA1 and DMA2 can work well. So this patch masks
the buggy DMA0 by keeping it as opened.

This is a tentative workaround, so keep the change small as Takashi suggested.

Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-13 12:11:58 +02:00
Aaron Plattner
ec5fe98886 ALSA: hda - Add new GPU codec ID to snd-hda
Vendor ID 0x10de0071 is used by a yet-to-be-named GPU chip.

Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-13 09:14:13 +02:00
Lars-Peter Clausen
ce85a4d726 ASoC: dapm: Fix SUSPEND -> OFF bias sequence
Currently when the DAPM context bias level is SUSPEND and the target bias level
is OFF dapm_pre_sequence_async() will first transition to PREPARE and
dapm_post_sequence_async() will then transition back from PREPARE to STANDBY and
then to OFF.

This patch makes sure that dapm_pre_sequence_async() only transitions to PREPARE
when either going to ON or away from ON. This avoids the extra unnecessary
transitions.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:06:34 +01:00
Lars-Peter Clausen
ca5106ae3d ASoC: dapm: Skip CODEC<->CODEC links in connect_dai_link_widgets()
For CODEC to CODEC DAI links the paths are created in snd_soc_dapm_new_pcm().
Also for CODEC to CODEC links the widgets are connected cross-over via a DAI
link widget, meaning that the capture widget of one CODEC will be connected to
the playback widget of the other and vice versa. Whereas
snd_soc_dapm_connect_dai_link_widgets() directly connects the playback widget of
the CPU DAI to the playback widget of the CODEC DAI and the capture widget of
the CPU DAI to the capture widget of the CODEC DAI. So not skipping
CODEC<->CODEC links in snd_soc_dapm_connect_dai_link_widgets() will create
incorrect connections between the two CODECs which will cause DAPM to detect
active paths where there are none and unnecessarily power up widgets.

Fixes: b893ea5 ("ASoC: sapm: Automatically connect DAI link widgets in DAPM graph.")
Cc: <stable@vger.kernel.org> (for 3.14+)
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:33:36 +01:00
Nicolin Chen
868a6ca84e ASoC: pcm: Fix incorrect condition check for case SNDRV_PCM_TRIGGER_SUSPEND
The regular state before we execute SNDRV_PCM_TRIGGER_SUSPEND should be
SNDRV_PCM_TRIGGER_START, not SNDRV_PCM_TRIGGER_STOP. Thus fix it.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:16:06 +01:00
Hui Wang
a1f3b5fa11 ALSA: hda - add headset mic detect quirks for three Dell laptops
When we plug a 3-ring headset on the Dell machines (VID: 0x10ec0255,
SID: 0x1028065c; VID: 0x10ec0255, SID: 0x10280680; VID: 0x10ec0292,
SID: 0x10280684), the headset mic can't be detected, after apply this
patch, the headset mic can work well.

And on the machine with SID 0x10280684, and the Lineout and external
microphone should be routed to docking, this patch also fix this
problem.

BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-09 07:25:44 +02:00
Anssi Hannula
f06ab794af ALSA: hda - hdmi: Set converter channel count even without sink
Since commit 1df5a06a ("ALSA: hda - hdmi: Fix programmed active channel
count") channel count is no longer being set if monitor_present is 0.
This is because setting the count was moved after the CA value is
determined, which is only after the monitor_present check in
hdmi_setup_audio_infoframe().

Unfortunately, in some cases, such as with a non-spec-compliant codec or
with a problematic video driver, monitor_present is always 0. As a
specific example, this seems to happen with gen1 ATV (SiI1390 codec),
causing left-channel-only stereo playback (multi-channel playback has
apparently never worked with this codec despite it reporting 8 channels,
reason unknown).

Simply setting converter channel count without setting the pin infoframe
and channel mapping as well does not theoretically make much sense as
this will just mean they are out-of-sync and multichannel playback will
have a wrong channel mapping.

However, adding back just setting the converter channel count even in
no-monitor case is the safest change which at least fixes the stereo
playback regression on SiI1390 codec. Do that.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Stephan Raue <stephan@openelec.tv>
Tested-by: Stephan Raue <stephan@openelec.tv>
Cc: <stable@vger.kernel.org> # 3.12+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-05 16:28:10 +02:00
Clemens Ladisch
7040b6d1fe ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data
The TEAC UD-H01 firmware sends wrong feedback frequency values, thus
causing the PC to send the samples at a wrong rate, which results in
clicks and crackles in the output.

Add a workaround to detect and fix the corruption.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
[mick37@gmx.de: use sender->udh01_fb_quirk rather than
 ep->udh01_fb_quirk in snd_usb_handle_sync_urb()]
Reported-and-tested-by: Mick <mick37@gmx.de>
Reported-and-tested-by: Andrea Messa <andr.messa@tiscali.it>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02 18:21:55 +02:00
Takashi Iwai
1ee23fe07e ALSA: usb-audio: Fix deadlocks at resuming
The recent addition of the USB audio mixer suspend/resume may lead to
deadlocks when the driver tries to call usb_autopm_get_interface()
recursively, since the function tries to sync with the finish of the
other calls.  For avoiding it, introduce a flag indicating the resume
operation and avoids the recursive usb_autopm_get_interface() calls
during the resume.

Reported-and-tested-by: Bryan Quigley <gquigs@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02 18:17:06 +02:00
Takashi Iwai
1c53e7253e ALSA: usb-audio: Save mixer status only once at suspend
The suspend callback of usb-audio driver may be called multiple times
per suspend when multiple USB interfaces are bound to a single sound
card instance.  In such a case, it's superfluous to save the mixer
values multiple times.  This patch fixes it by checking the counter.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02 18:14:42 +02:00
Sander Eikelenboom
b7a7723513 ALSA: usb-audio: Prevent printk ratelimiting from spamming kernel log while DEBUG not defined
This (widely used) construction:

if(printk_ratelimit())
	dev_dbg()

Causes the ratelimiting to spam the kernel log with the "callbacks suppressed"
message below, even while the dev_dbg it is supposed to rate limit wouldn't
print anything because DEBUG is not defined for this device.

[  533.803964] retire_playback_urb: 852 callbacks suppressed
[  538.807930] retire_playback_urb: 852 callbacks suppressed
[  543.811897] retire_playback_urb: 852 callbacks suppressed
[  548.815745] retire_playback_urb: 852 callbacks suppressed
[  553.819826] retire_playback_urb: 852 callbacks suppressed

So use dev_dbg_ratelimited() instead of this construction.

Signed-off-by: Sander Eikelenboom <linux@eikelenboom.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-02 18:10:59 +02:00
Hui Wang
91943954e3 ALSA: hda - add headset mic detect quirk for a Dell laptop
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x1028067e), the headset mic can't be detected, after apply this
patch, the headset mic can work well.

BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-30 12:36:48 +02:00
Takashi Iwai
6ba736dd02 ALSA: hda - Suppress CORBRP clear on Nvidia controller chips
The recent commit (ca460f8652) changed the CORB RP reset procedure to
follow the specification with a couple of sanity checks.
Unfortunately, Nvidia controller chips seem not following this way,
and spew the warning messages like:
  snd_hda_intel 0000:00:10.1: CORB reset timeout#1, CORBRP = 0

This patch adds the workaround for such chips.  It just skips the new
reset procedure for the known broken chips.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-29 18:41:22 +02:00
Hui Wang
e32dfbed8c ALSA: hda - add headset mic detect quirk for a Dell laptop
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x10280674), the headset mic can't be detected, after apply this
patch, the headset mic can work well.

BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-28 12:15:02 +02:00
Mark Brown
98810a6dcf Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/jz4740', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/tlv320aic3x' into asoc-linus 2014-04-22 22:01:07 +01:00
Mark Brown
22e0c14280 Merge remote-tracking branches 'asoc/fix/alc5623', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73' and 'asoc/fix/fsl-spdif' into asoc-linus 2014-04-22 22:01:05 +01:00
Mark Brown
31835046be Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus 2014-04-22 22:01:04 +01:00
Lars-Peter Clausen
eebdec044e ASoC: jz4740: Remove Makefile entry for removed file
Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver")
jz4740-pcm.c file, but neglected to remove the Makefile entries.

Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver")
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Reported-by: Ralf Baechle <ralf@linux-mips.org>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 21:51:58 +01:00
Wenkai Du
95e9ee92e2 ASoC: Intel: Fix audio crash due to negative address offset
There were occasional ADSP crash during reboot testing:

[   11.883364] BUG: unable to handle kernel paging request at ffffc90121700000
[   11.883380] IP: [<ffffffffc024d8bc>] sst_module_insert_fixed_block+0x24f/0x26d [snd_soc_sst_dsp]
[   11.883397] PGD 7800b067 PUD 0
[   11.883405] Oops: 0002 [#1] SMP
[   11.886418] gsmi: Log Shutdown Reason 0x03

The virtual address, ffffc90121700000, was out of range. The virtual
address is calculated by adding LPE base address with an offset:

sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size);

The offset is calculated in sst_byt_parse_module, by subtraction of
two virtual addresses dsp->addr.fw_ext and dsp->addr.lpe:

block_data.offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe);

These virtual addresses are assigned by kernel from ioremap:

sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size);
sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size);

In current driver code, offset is defined as unsigned int32:

struct sst_module_data {
...
	u32 offset;		/* offset in FW file */
};

Most of the time kernel assigned virtual addresses with addr.fw_ext
greater than addr.lpe. But sometimes it was the other way round.

Fix the problem by declaring offset as signed int32_t.

Signed-off-by: Wenkai Du <wenkai.du@intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 19:22:53 +01:00
Jarkko Nikula
2697e4fb92 ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrol
Commit 9e1fda4ae158 ("ASoC: dapm: Implement mixer input auto-disable")
is trying to free the widget it allocated by snd_soc_dapm_new_control()
call in dapm_kcontrol_data_alloc() by adding kfree(data->widget) to
dapm_kcontrol_free().

This is causing a widget double free with auto-disabled DAPM kcontrols
in sound card unregistration because widgets are already freed before
dapm_kcontrol_free() is called.

Reason for that is all widgets are added into dapm->card->widgets list
in snd_soc_dapm_new_control() and freed in dapm_free_widgets() during
execution of snd_soc_dapm_free().

Now snd_soc_dapm_free() calls for different DAPM contexts happens before
snd_card_free() call from where the call chain to dapm_kcontrol_free()
begins:

soc_cleanup_card_resources()
  soc_remove_dai_links()
    soc_remove_link_dais()
      snd_soc_dapm_free(&cpu_dai->dapm)
    soc_remove_link_components()
      soc_remove_platform()
        snd_soc_dapm_free(&platform->dapm)
      soc_remove_codec()
        snd_soc_dapm_free(&codec->dapm)
  snd_soc_dapm_free(&card->dapm)
  snd_card_free()
    snd_card_do_free()
      snd_device_free_all()
        snd_device_free()
          snd_ctl_dev_free()
            snd_ctl_remove()
              snd_ctl_free_one()
                dapm_kcontrol_free()

This wasn't making harm with ordinary DAPM kcontrols since data->widget is NULL for
them.

Fixes: 9e1fda4ae158 (ASoC: dapm: Implement mixer input auto-disable)
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-04-21 11:49:53 +01:00
Christian Engelmayer
bf657d2479 ASoC: Intel: Fix incorrect sizeof() in sst_hsw_stream_get_volume()
Fix an incorrect sizeof() usage in sst_hsw_stream_get_volume(). sst_dsp_read()
is called to read into a variable of type u32, but is passed sizeof(u32 *) for
argument 'size_t bytes'. Detected by Coverity: CID 1195260.

Signed-off-by: Christian Engelmayer <cengelma@gmx.at>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-18 17:11:16 +01:00
Dan Carpenter
7897ab78f6 ASoC: Intel: some incorrect sizeof() usages
The intent was to say "sizeof(*pos)" and not "sizeof(pos)".

The sizeof(*pos) is 8 bytes so the bug won't show up on 64 bit systems.
The sizeof(*dx) is 172 bytes so that will be a bugfix.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-18 17:11:03 +01:00
Axel Lin
2b21694f15 ASoC: cs42l73: Convert to use devm_gpio_request_one
Current code missed a gpio_free() call in cs42l73_i2c_remove().
Convert to use devm_gpio_request_one() to fix it.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-18 15:15:11 +01:00
Axel Lin
4e17d2d33a ASoC: cs42l52: Convert to use devm_gpio_request_one
Current code missed a gpio_free() call in cs42l52_i2c_remove().
Convert to use devm_gpio_request_one() to fix it.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-18 15:14:39 +01:00
Nicolin Chen
6ae6698276 ASoC: fsl_spdif: Fix wrong OFFSET of STC_SYSCLK_DIV
It should use STC_SYSCLK_DIV_OFFSET. Thus fix it.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-18 15:05:30 +01:00
Kailang Yang
8dc9abb93d ALSA: hda/realtek - Add headset Mic support for Dell machine
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-16 10:30:46 +02:00
Hui Wang
67807ce55f ALSA: hda - add headset mic detect quirk for a Dell laptop
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255,
SID: 0x1028067f), the headset mic can't be detected, after apply this
patch, the headset mic can work well.

BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-16 07:47:50 +02:00
Andrew Lunn
8aaa414fad ASoC: alc5623: Fix regmap endianness
Commit 0cd257bf9b, "ASoC: alc5623:
Convert to direct regmap API usage" broke probing of the codec,
because of wrong endinness of the ID and codec version read from the
device. Fix this by removing the existing flipping of the endiannes,
and extracting the codec type byte from the word from the regmap.

Signed-off-by: Andrew Lunn <andrew@lunn.ch>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-15 14:21:21 +01:00
Sebastian Reichel
3b5b243157 ASoC: tlv320aic3x: fix shared reset pin for DT
Currently the second tlv320aic3x instance fails to
be probed from DT if the reset pin is shared with
the first one.

This patch fixes it by moving the list add of the
reset pin into the i2c_probe method.

Signed-off-by: Sebastian Reichel <sre@kernel.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-15 13:23:52 +01:00
Li, Zhen-Hua
5e0c187991 ALSA: es18xx driver should use udelay error
udelay with more than 20000 may cause __bad_udelay.
Use mdelay for instead.

[fixed a typo spotted by Clemens -- tiwai]

Signed-off-by: Li, Zhen-Hua <zhen-hual@hp.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-14 14:27:27 +02:00
Kailang Yang
7c66593286 ALSA: hda/realtek - Add support of ALC288 codec
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-14 10:42:39 +02:00
Ben Dooks
79861bbb75 ASoC: rsnd: fix clock prepare/unprepare
As with the previous commit, before a clock can be used it must be prepared
for use. Change from clk_enable() and clk_disable() to the versions of the
calls which also prepare and un-prepare the clocks.

Will fix warnings from the clock code when this is used.

Signed-off-by: Ben Dooks <ben.dooks@codethink.co.uk>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-11 12:13:52 +01:00
Linus Torvalds
e7990d45bb sound fixes for 3.15-rc1
Here is a bunch of small fixes that have been collected since the
 previous pull request.  In addition to various misc fixes, the
 following are included:
 
 - HD-audio quirks for Dell, HP, Chromebook, and ALC28x codecs
 - HD-audio AMD HDMI regression fix
 - Continued PM support/fixes for ice1712 driver
 - Multiplatform fixes for ASoC samsung drivers
 - Addition of device id tables to a few ASoC drivers
 - Bit clock polarity config and error flag fixes in ASoC fsl_sai
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Merge tag 'sound-fix-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Here is a bunch of small fixes that have been collected since the
  previous pull request.  In addition to various misc fixes, the
  following are included:

   - HD-audio quirks for Dell, HP, Chromebook, and ALC28x codecs
   - HD-audio AMD HDMI regression fix
   - Continued PM support/fixes for ice1712 driver
   - Multiplatform fixes for ASoC samsung drivers
   - Addition of device id tables to a few ASoC drivers
   - Bit clock polarity config and error flag fixes in ASoC fsl_sai"

* tag 'sound-fix-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (32 commits)
  ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
  ALSA: hda - Make full_reset boolean
  ALSA: hda - add headset mic detect quirk for a Dell laptop
  sound: dmasound: use module_platform_driver_probe()
  ALSA: au1x00: use module_platform_driver()
  ALSA: hda - Use runtime helper to check active state.
  ALSA: ice1712: Fix boundary checks in PCM pointer ops
  ASoC: davinci-mcasp: Fix bit clock polarity settings
  ASoC: samsung: Fix build on multiplatform
  ASoC: fsl_sai: Fix Bit Clock Polarity configurations
  ALSA: hda - Do not assign streams in reverse order
  ALSA: hda/realtek - Add eapd shutup to ALC283
  ALSA: hda/realtek - Change model name alias for ChromeOS
  ASoC: da732x: Print correct major id
  ALSA: hda/realtek - Improve HP depop when system change power state on Chromebook
  ASoC: cs42l52: Fix mask for REVID
  sound/oss: Remove uncompilable DBG macro use
  ALSA: ice1712: Save/restore routing and rate registers
  ALSA: ice1712: restore AK4xxx volumes on resume
  ASoC: alc56(23|32): fix undefined return value of probing code
  ...
2014-04-10 09:19:44 -07:00
Tim Gardner
a5065eb6da ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
BugLink: http://bugs.launchpad.net/bugs/1305133

Malfunctioning or slow devices can cause a flood of dmesg SPAM.

I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour
of prior art in sound/usb/pcm.c.

WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit
+	if (printk_ratelimit() &&

Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Eldad Zack <eldad@fogrefinery.com>
Cc: Daniel Mack <zonque@gmail.com>
Cc: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Tim Gardner <tim.gardner@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 21:07:38 +02:00
Takashi Iwai
50487c3a4a ASoC: Fixes for v3.15
A smattering of device specific fixes, nothing stands out here except
 for the multiplatform fixes for Samsung and the device IDs being added
 by Stephen Warren - there's no real code changes from those and they
 give better robustness to the enumeration with DT.
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Merge tag 'asoc-v3.15-5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.15

A smattering of device specific fixes, nothing stands out here except
for the multiplatform fixes for Samsung and the device IDs being added
by Stephen Warren - there's no real code changes from those and they
give better robustness to the enumeration with DT.
2014-04-09 17:50:02 +02:00
Thierry Reding
17c3ad0302 ALSA: hda - Make full_reset boolean
The full_reset argument to azx_init_chip() carries boolean rather than
numerical information, so update the type to reflect that.

Signed-off-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 14:48:07 +02:00
Hui Wang
137bcc33c6 ALSA: hda - add headset mic detect quirk for a Dell laptop
When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0283,
SID: 0x10280667), the headset mic can't be detected, after apply this
patch, the headset mic can work well.

BugLink: https://bugs.launchpad.net/bugs/1297581
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 10:05:51 +02:00
Christoph Jaeger
292ab81df0 sound: dmasound: use module_platform_driver_probe()
Eliminate boilerplate code by using module_platform_driver_probe().

Signed-off-by: Christoph Jaeger <christophjaeger@linux.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 10:02:21 +02:00
Christoph Jaeger
34f980630f ALSA: au1x00: use module_platform_driver()
Eliminate boilerplate code by using module_platform_driver().

Signed-off-by: Christoph Jaeger <christophjaeger@linux.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 10:02:14 +02:00
Dylan Reid
7b0a48f340 ALSA: hda - Use runtime helper to check active state.
From azx_interrupt, use the helper to check if the device is active
instead of checking the state.  This will do the right thing if
runtime pm is disabled in addition to if the device is suspended.

Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-09 09:33:03 +02:00
Mark Brown
ab5d6fbdb7 Merge remote-tracking branches 'asoc/fix/rt5640', 'asoc/fix/samsung', 'asoc/fix/tlv320aic23' and 'asoc/fix/warn' into asoc-linus 2014-04-08 21:22:14 +01:00
Mark Brown
bfef92bb97 Merge remote-tracking branches 'asoc/fix/alc5632', 'asoc/fix/cs42l52', 'asoc/fix/cs42xxx8', 'asoc/fix/da732x', 'asoc/fix/davinci', 'asoc/fix/fsl-sai', 'asoc/fix/fsl-ssi' and 'asoc/fix/max98090' into asoc-linus 2014-04-08 21:22:11 +01:00
Takashi Iwai
4f8e940095 ALSA: ice1712: Fix boundary checks in PCM pointer ops
PCM pointer callbacks in ice1712 driver check the buffer size boundary
wrongly between bytes and frames.  This leads to PCM core warnings
like:
   snd_pcm_update_hw_ptr0: 105 callbacks suppressed
   ALSA pcm_lib.c:352 BUG: pcmC3D0c:0, pos = 5461, buffer size = 5461, period size = 2730

This patch fixes these checks to be placed after the proper unit
conversions.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-08 16:58:34 +02:00
Peter Ujfalusi
74ddd8c40d ASoC: davinci-mcasp: Fix bit clock polarity settings
IB_NF, NB_IF and IB_IF configured the bc polarity incorrectly. The receive
polarity was set to the same edge as the TX in these cases.

Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-08 12:56:45 +01:00
Sachin Kamat
71e5222cbe ASoC: samsung: Fix build on multiplatform
PCM and S/PDIF drivers referenced mach headers for a trivial
data structure. This caused build errors on multiplatform builds
as machine headers are not accessible from driver files. Move the data
structure definition to the driver header and remove the dependency.
While at it rename the structure to avoid multiple definition errors
as the same structure is also used by the platform code.

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-08 12:51:57 +01:00
Nicolin Chen
ef33bc3217 ASoC: fsl_sai: Fix Bit Clock Polarity configurations
The BCP bit in TCR4/RCR4 register rules as followings:
  0 Bit clock is active high with drive outputs on rising edge
    and sample inputs on falling edge.
  1 Bit clock is active low with drive outputs on falling edge
    and sample inputs on rising edge.

For all formats currently supported in the fsl_sai driver, they're exactly
sending data on the falling edge and sampling on the rising edge.

However, the driver clears this BCP bit for all of them which results click
noise when working with SGTL5000 and big noise with WM8962.

Thus this patch corrects the BCP settings for all the formats here to fix
the nosie issue.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Acked-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-08 12:49:55 +01:00
Anssi Hannula
dcb32ecd9a ALSA: hda - Do not assign streams in reverse order
Currently stream numbers are assigned in reverse order.

Unfortunately commit 7546abfb8e ("ALSA: hda - Increment
default stream numbers for AMD HDMI controllers") assumed this was not
the case (specifically, it had the "old cards had single device only"
=> "extra unused stream numbers do not matter" assumption), causing
non-working audio regressions for AMD Radeon HDMI users.

Change the stream numbers to be assigned in forward order.

The benefit is that regular audio playback will still work even if the
assumed stream count is too high, downside is that a too high stream
count may remain hidden.

Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=77002
Reported-by: Christian Güdel <cg@dmesg.ch>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: Christian Güdel <cg@dmesg.ch> # 3.14
Cc: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-08 11:44:21 +02:00
Kailang Yang
0435b3ffba ALSA: hda/realtek - Add eapd shutup to ALC283
Add eapd shutup function to alc283_shutup.
It could avoid pop noise from speaker.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-08 11:27:29 +02:00