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Commit Graph

18403 Commits

Author SHA1 Message Date
Mark Brown
6bf88ab2ec Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next 2014-05-22 00:23:42 +01:00
Mark Brown
1450da3cf6 Merge remote-tracking branch 'asoc/topic/core' into asoc-next 2014-05-22 00:23:41 +01:00
Mark Brown
0f4019e6f4 Merge remote-tracking branch 'asoc/topic/component' into asoc-next 2014-05-22 00:23:41 +01:00
Mark Brown
228704bbdd Merge remote-tracking branch 'asoc/fix/max98090' into asoc-linus 2014-05-22 00:23:37 +01:00
Mark Brown
95b9cff321 ASoC: Intel fixes for v3.15
This is a relatively large batch of fixes for the newly added
 Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
 this point in the cycle but it's all for a newly added driver so not so
 worrying as it might otherwise be.  Some of it's integration problems,
 some of it's the sort of problem usually turned up in stress tests.
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Merge tag 'asoc-v3.15-rc5-intel' into asoc-linus

ASoC: Intel fixes for v3.15

This is a relatively large batch of fixes for the newly added
Haswell/Baytrail drivers from Intel.  It's a bit larger than is good for
this point in the cycle but it's all for a newly added driver so not so
worrying as it might otherwise be.  Some of it's integration problems,
some of it's the sort of problem usually turned up in stress tests.

# gpg: Signature made Wed 14 May 2014 12:40:27 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:36 +01:00
Mark Brown
dd97254f5c ASoC: Driver fixes for v3.15
A small set of driver fixes, nothing remarkable in itself or of any
 relevance outside of the driver.
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Merge tag 'asoc-v3.15-rc5-drivers' into asoc-linus

ASoC: Driver fixes for v3.15

A small set of driver fixes, nothing remarkable in itself or of any
relevance outside of the driver.

# gpg: Signature made Wed 14 May 2014 12:49:57 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:31 +01:00
Mark Brown
266bd275b9 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' into asoc-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.

# gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>"
# gpg:                 aka "Mark Brown <broonie@debian.org>"
# gpg:                 aka "Mark Brown <broonie@kernel.org>"
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>"
# gpg:                 aka "Mark Brown <broonie@linaro.org>"
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>"
2014-05-22 00:23:30 +01:00
Liam Girdwood
f1c0bc9145 ASoC: max98090: Mark cache as dirty prior to restoring
Make sure the cache is fully flushed at resume time.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:57:15 +01:00
Liam Girdwood
46b0e97dcf ASoC: max98090: Reset codec on resume
Make sure we reset codec and clear any IRQs on resume. This matches
the init sequence in probe.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:57:15 +01:00
Liam Girdwood
25b4ab430f ASoC: max98090: Fix reset at resume time
Reset needs to wait 20ms before other codec IO is performed. This wait
was not being performed. Fix this by making sure the reset register is not
restored with the cache, but use the manual reset method in resume with
the wait.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-05-16 19:56:23 +01:00
Liam Girdwood
729af1ce6c ASoC: max98090: Fix digital sidetone gain TLV
TLV for digital sidetone volume is wrong, this fix matches it to the
datasheet.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-16 19:56:20 +01:00
Dan Carpenter
15b8e94f74 ASoC: compress: indent an if statement
The return statement was not indented correctly.  I lined up the
condition a bit as well.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-14 16:15:03 +01:00
Mark Brown
cf86197ec5 Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus 2014-05-14 12:52:41 +01:00
Mark Brown
f9a405961e Merge remote-tracking branches 'asoc/fix/audmux', 'asoc/fix/cs42l52', 'asoc/fix/fsl-esai', 'asoc/fix/fsl-spdif', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/wm8962' into asoc-linus 2014-05-14 12:49:10 +01:00
Charles Keepax
44330ab516 ASoC: wm8962: Update register CLASS_D_CONTROL_1 to be non-volatile
The register CLASS_D_CONTROL_1 is marked as volatile because it contains
a bit, DAC_MUTE, which is also mirrored in the ADC_DAC_CONTROL_1
register. This causes problems for the "Speaker Switch" control, which
will report an error if the CODEC is suspended because it relies on a
volatile register.

To resolve this issue mark CLASS_D_CONTROL_1 as non-volatile and
manually keep the register cache in sync by updating both bits when
changing the mute status.

Reported-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Tested-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
2014-05-13 19:02:30 +01:00
Jarkko Nikula
cffd6665f5 ASoC: Intel: Fix Baytrail SST DSP firmware loading
Commit 10df350977 ("ASoC: Intel: Fix Audio DSP usage when IOMMU is
enabled.") caused following regression in Baytrail SST:

baytrail-pcm-audio baytrail-pcm-audio: error: DMA alloc failed
baytrail-pcm-audio baytrail-pcm-audio: error: failed to load firmware

Fix this by calling dma_coerce_mask_and_coherent() in sst_byt_init() with
the same dma_dev device what is now used in sst_fw_new() when allocating the
DMA buffer.

Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-13 18:21:02 +01:00
Nicolin Chen
4f8210f66e ASoC: fsl_esai: Set PCRC and PRRC registers at the end of hw_params()
According to Reference Manual -- ESAI Initialization chapter, as the
standard procedure of ESAI personal reset, the PCRC and PRRC registers
should be remained in its reset value and then configured after T/RCCR
and T/RCR configurations's done but before TE/RE's enabling.

So this patch moves PCRC and PRRC settings to the end of hw_params().

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:13:13 +01:00
Nicolin Chen
57ebbcafab ASoC: fsl_esai: Only bypass sck_div for EXTAL source
ESAI can only output EXTAL clock source directly. But for FSYS clock source,
ESAI can not output it without getting through PSR PM dividers.

So this patch adds an extra check in the code.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:13:13 +01:00
Nicolin Chen
89e47f62cf ASoC: fsl_esai: Fix incorrect condition within ratio range check for FP
The range here from 1 to 16 is confined to FP divider only while the
sck_div indicates if the calculation contains PSR and PM dividers. So
for the case using PSR and PM since the sck_div is true, the range of
ratio would simply become bigger than 16.

So this patch fixes the condition here and adds one line comments to
make the purpose here clear.

Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 23:13:13 +01:00
Lars-Peter Clausen
797f283b61 ASoC: Remove runtime field from DAI
This was initially removed in commit 6423c1875 ("ASoC: Remove runtime field from
DAI"), but was, presumably by accident, brought back in commit f0fba2ad1 ("ASoC:
multi-component - ASoC Multi-Component Support"). But has never been
initialized to anything but NULL ever since. This commit removes it again.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:08:36 +01:00
Lars-Peter Clausen
b74f7be90f ASoC: atmel-pcm-pdc: Remove broken suspend/resume code
Suspend/resume support for the atmel-pcm-pdc driver was broken in commit
f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support"). It
essentially reverted the modifications done in commit 10cab262 ("ASoC: Change
how suspend and resume obtain the PCM runtime"). The suspend and resume handlers
at the beginning check if dai->runtime is not NULL, but dai->runtime is always
NULL, hence the code never runs. Considering that nobody noticed any problems in
the last 4 years since the code was broken and that the driver does not set
SNDRV_PCM_INFO_RESUME, which means applications are expected to stop and restart
the audio stream during suspend/resume, it is probably safe to assume that his
code is not needed and can be removed.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 22:08:36 +01:00
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
94986198f5 ASoC: dapm: Handle SND_SOC_DAPM_REG() generically
Commit commit de9ba98b6d ("ASoC: dapm: Make widget power register settings more
flexible") added generic support for on_val/off_val in the DAPM core. With this
in place there is no need anymore for having a special event callback for
SND_SOC_DAPM_REG() widgets.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:48:08 +01:00
Lars-Peter Clausen
0f9bd7b194 ASoC: dapm: Simplify snd_soc_dapm_link_dai_widgets()
If we find a widget who's stream name matches the name of a DAI widget then
thats the one it should be connected to. Based on the widget id we can say in
which direction the path should be. No need to go back to the DAI and check the
stream names.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:37:17 +01:00
Lars-Peter Clausen
fe83897fc5 ASoC: dapm: Use snd_soc_dapm_add_path() in snd_soc_dapm_new_pcm()
We already know the widgets we want to connect, so use snd_soc_dapm_add_path()
instead of snd_soc_dapm_add_route() in snd_soc_dapm_new_pcm().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:34:49 +01:00
Lars-Peter Clausen
9887c20b9f ASoC: dapm: Use snd_soc_dapm_add_path() in connect_dai_link_widgets()
We already know which two widgets should be connected, so use
snd_soc_dapm_add_path() instead of snd_soc_dapm_add_route() in
snd_soc_dapm_connect_dai_link_widgets().

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:34:48 +01:00
Lars-Peter Clausen
a4e9154c42 ASoC: dapm: Revert "ASoC: dapm: Fix double prefix addition"
This reverts commit bd23c5b661.

The patch claims that the patch is necessary to avoid double prefix addition
when calling snd_soc_dapm_add_route() from snd_soc_dapm_connect_dai_link_widgets().
But snd_soc_dapm_add_route() is called with the card's DAPM context, which does
not have a prefix, which means there is no prefix that could be added a second
time.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-12 21:34:43 +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
Lars-Peter Clausen
db88a8e3ca ASoC: Remove unused num_dai field from CODEC
Commit d191bd8de8 ("ASoC: snd_soc_codec includes snd_soc_component") removed the
last user of the num_dai field. Also remove the field itself.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 10:21:26 +01:00
Lars-Peter Clausen
af0881ffbd ASoC: Remove unused 'list' field form card
The global card list was removed in commit b19e6e7b7 ("ASoC: core: Use driver
core probe deferral"). The 'list' field of the snd_soc_card struct has been
unused since then. This patch removes the field.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 10:21:25 +01:00
Lars-Peter Clausen
24faf76568 ASoC: Remove card's DAI list
Commit f0fba2ad1 ("ASoC: multi-component - ASoC Multi-Component Support") added
a per card list that keeps track of all the DAIs that have been registered with
the card, but the list has never been used. This patch removes it again.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 10:21:25 +01:00
Mark Brown
387f837b3d Merge branch 'topic/component' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-core 2014-05-07 10:21:22 +01:00
Liam Girdwood
2b39aab18a ASoC: Intel: Fix block offset calculations.
Block offset calculations are done in the contiguous allocator so
are not required here.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-07 09:38:29 +01:00
Liam Girdwood
e9024f0ba3 ASoC: Intel: Fix check for pdata usage before dereference.
This patch fixes the following dereference check ordering.

 sound/soc/intel/sst-haswell-pcm.c:749 hsw_pcm_probe() warn: variable dereferenced before check 'pdata' (see line 746)

 git remote add asoc git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git
 git remote update asoc
 git checkout 0b708c87f6
 vim +/pdata +749 sound/soc/intel/sst-haswell-pcm.c

 a4b12990 Mark Brown    2014-03-12  740  };
 a4b12990 Mark Brown    2014-03-12  741
 a4b12990 Mark Brown    2014-03-12  742  static int hsw_pcm_probe(struct snd_soc_platform *platform)
 a4b12990 Mark Brown    2014-03-12  743  {
 a4b12990 Mark Brown    2014-03-12  744  	struct sst_pdata *pdata = dev_get_platdata(platform->dev);
 a4b12990 Mark Brown    2014-03-12  745  	struct hsw_priv_data *priv_data;
 0b708c87 Liam Girdwood 2014-05-02 @746  	struct device *dma_dev = pdata->dma_dev;
 0b708c87 Liam Girdwood 2014-05-02  747  	int i, ret = 0;
 a4b12990 Mark Brown    2014-03-12  748
 a4b12990 Mark Brown    2014-03-12 @749  	if (!pdata)
 a4b12990 Mark Brown    2014-03-12  750  		return -ENODEV;
 a4b12990 Mark Brown    2014-03-12  751
 a4b12990 Mark Brown    2014-03-12  752  	priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL);

Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:42:00 -07:00
Lars-Peter Clausen
c9e065c27f ASoC: dapm: Make sure to always update the DAPM graph in _put_volsw()
When using auto-muted controls it may happen that the register value will not
change when changing a control from enabled to disabled (since the control might
be physically disabled due to the auto-muting). We have to make sure to still
update the DAPM graph and disconnect the mixer input.

Fixes: commit 5729507 ("ASoC: dapm: Implement mixer input auto-disable")
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-05 12:31:14 -07:00
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
Vinod Koul
d98812082c ASoC: add SND_SOC_BYTES_EXT
we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data
for DSP modules

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 13:44:24 -07:00
Liam Girdwood
51b4e24f38 ASoC: Intel: Fix stream position pointer.
Read the stream offset and presentation position from DSP memory rather
than using the old estimated position. This fixes timing issues with
pulseaudio.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 09:54:05 -07:00
Liam Girdwood
916152c488 ASoC: Intel: Fix allow hw_params to be called more than once.
hw_params() can be called multiple times. Make sure we release the DSP
stream that was allocated on previous hw_params() calls before allocating
a new DSP stream.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 09:53:02 -07:00
Liam Girdwood
10df350977 ASoC: Intel: Fix Audio DSP usage when IOMMU is enabled.
The Intel IOMMU requires that the ACPI device is used to allocate all
DMA memory buffers. This means we need to pass the DMA device pointer into child
component devices that allocate DMA memory.

We also only set the DMA mask for the ACPI device now instead of for each
component device.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 09:53:02 -07:00
Liam Girdwood
0b708c87f6 ASoC: Intel: Fix Haswell/Broadwell DSP page table creation.
Fix page table creation on Haswell and Broadwell to remove unsafe
virt_to_phys mappings and use more portable SG buffer. Use audio buffer
APIs to allocate DMA buffers.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 09:53:01 -07:00
Liam Girdwood
84fbdd5861 ASoC: Intel: Fix allocated block list usage when adding blocks.
Make sure we add the allocated blocks to the modules list of blocks.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 09:53:01 -07:00
Liam Girdwood
48695f3d4e ASoC: Intel: Fix block allocation so we only allocate blocks once.
Make sure we dont alloc blocks twice with requests spanning more
than one block.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 09:51:58 -07:00
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
Jarkko Nikula
4da533932d ASoC: core: Fix component_list corruption when unloading modules
This fixes module unload regressions introduced by commits 98e639fb8a
("ASoC: Track which components have been registered with
snd_soc_register_component()") and b37f1d123c ("ASoC: Let snd_soc_platform
subclass snd_soc_component").

First commit causes component_list to be corrupted when removing codec and
second when removing platform. Reason for both is that components associated
with platform or codec are never removed from the list because for them
registered_as_component field in struct snd_soc_component is always false.

Now list becomes corrupted when snd_soc_unregister_platform() or
snd_soc_unregister_codec() frees the platform or codec structure and where
the associated struct snd_soc_component is embedded.

Fix these by moving component unregistration and cleanup to a new local
function __snd_soc_unregister_component() that takes component as its
argument.

Since component is known for platforms and codecs the
__snd_soc_unregister_component() can be called directly and
snd_soc_unregister_component() takes care to find and unregister only
components that were registered using snd_soc_register_component().

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>
2014-04-29 10:09:11 -07:00