Commit Graph

18342 Commits

Author SHA1 Message Date
Dan Carpenter 665ebe926e ALSA: sb_mixer: missing return statement
The if condition here was supposed to return on error but the return
statement is missing.  The effect is that the ->mixername is set to
"???" instead of "DT019X".

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-14 16:46:48 +02:00
Takashi Iwai ff2354bc6e Merge tag 'asoc-v3.15-rc5-intel' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-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.
2014-05-14 14:27:12 +02:00
Takashi Iwai 7ca33c7a1d Merge tag 'asoc-v3.15-rc5-drivers' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-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.
2014-05-14 14:24:09 +02:00
Takashi Iwai 927cdab3b6 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
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
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
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 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
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
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
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