Commit Graph

18398 Commits

Author SHA1 Message Date
Takashi Sakamoto
83d8d72dff ALSA: firewire-lib: Add support for MIDI capture/playback
For capturing/playbacking MIDI messages, this commit adds one MIDI conformant
data channel. This data channel has multiplexed 8 MIDI data streams. So this
data channel can transfer messages from/to 8 MIDI ports.

And this commit allows to set PCM format even if AMDTP streams already start.
I suppose the case that PCM substreams are going to be joined into AMDTP
streams when AMDTP streams are already started for MIDI substreams. Each
driver must count how many PCM/MIDI substreams use AMDTP streams to stop
AMDTP streams.

There are differences between specifications about MIDI conformant data.

About the multiplexing, IEC 61883-6:2002, itself, has no information. It
describes labels and bytes for MIDI messages and refers to MMA/AMEI RP-027
for 'successfull implementation'. MMA/AMEI RP-027 describes 8 MPX-MIDI data
streams for one MIDI conformant data channel. IEC 61883-6:2005 adds
'sequence multiplexing' and apply this way and describe incompatibility
between 2002 and 2005.

So this commit applies IEC 61883-6:2005. When we find some devices compliant
to IEC 61883-6:2002, then this difference should be handles as device quirk
in additional work.

About the number of bytes in an MIDI conformant data, IEC 61883-6:2002 describe
0,1,2,3 bytes. MMA/AMEI RP-027 describes 'MIDI1.0-1x-SPEED', 'MIDI1.0-2x-SPEED',
'MIDI1.0-3x-SPEED' modes and the maximum bytes for each mode corresponds to 1,
2, 3 bytes. The 'MIDI1.0-2x/3x-SPEED' modes are accompanied with 'negotiation
procedure' and 'encapsulation details' but there is no specifications for them.

So this commit implements 'MIDI1.0-1x-SPEED' mode for playback, but allows
to pick up 1-3 bytes for capturing.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:12:44 +02:00
Takashi Sakamoto
2b3fc456fe ALSA: firewire-lib: Add support for AMDTP in-stream and PCM capture
For capturing PCM, this commit adds the functionality to handle in-stream.
This is also applied for dual-wire mode.

Currently, capturing 32bit samples are supported.

When the sequence of in-packet has discontinuity of dbc, in-stream isn't handled
and amdtp_streaming_error() returns true.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:12:35 +02:00
Takashi Sakamoto
4b7da117e5 ALSA: firewire-lib: Split some codes into functions to reuse for both streams
Some codes can be reused to handle in-stream. This commit adds new functions.
This commit also renames some functions to keep naming consistency.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:57 +02:00
Takashi Sakamoto
3ff7e8f0d4 ALSA: firewire-lib: Add 'direction' member to 'amdtp_stream' structure
This patch adds 'direction' member to amdtp_stream structure to indicate its
direction. This patch also adds 'direction' argument to amdtp_stream_init()
function to determine its direction.

The amdtp_stream_init() function is exported and used by firewire-speakers and
dice so this patch also affects them.

This patch just add them. Actual implementation will be done by followed
patches.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:42 +02:00
Takashi Sakamoto
b445db440c ALSA: firewire-lib: Add macros instead of fixed value for AMDTP
This patch adds some macros instead of fixed value for AMDTP according to
IEC 61883-1/6. These macros will also be used by followed patches.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:22 +02:00
Takashi Sakamoto
be4a28940a ALSA: firewire-lib: Rename functions, structure, member for AMDTP
This patch renames some functions, a structure and its member to reuse them
in both AMDTP in/out stream.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:11:10 +02:00
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 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' 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 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' 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 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
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
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
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
Joe Perches
2a1c23e339 ASoC: tlv320aic31xx: Convert /n to \n
Use a newline character appropriately.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-25 12:14:46 +01:00
Jarkko Nikula
de30a2ccb2 ASoC: Intel: Cancel hsw_notification_work before freeing the stream
I suppose there is a possibility that hsw_notification_work() may run after
sst_hsw_stream_free() which can lead to a kernel crash since struct
sst_hsw_stream is freed at that point and
stream = container_of(work, struct sst_hsw_stream, notify_work) is not valid
when hsw_notification_work() is run.

Reported-by: Derek Basehore <dbasehore@chromium.org>
Reported-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>
2014-04-24 11:32:23 +01:00
Lars-Peter Clausen
b8909783a2 ASoC: imx-audmux: Fix section mismatch
audmux_debugfs_init() is marked as __init, but is called from imx_audmux_probe()
which is not marked as __init. This creates a section mismatch and a potential
runtime crash (if imx_audmux_probe() is called after the .init section was
dropped). This patch removes the __init annotation from audmux_debugfs_init(),
which fixes the following warning:
	WARNING: sound/soc/built-in.o(.text+0x86960): Section mismatch in reference
	from the function imx_audmux_probe() to the function
	.init.text:audmux_debugfs_init()

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-24 11:22:53 +01:00
Kuninori Morimoto
836b31fe1a ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock
rsnd_soc_dai_trigger() will be called
after rsnd_dai_pointer_update() function
which is using rsnd_lock().
Thus, it should be called from outside of rsnd_lock().
Kernel will be hangup without this patch.
Special thanks to Kataoka-san

Reported-by: Ryo Kataoka <ryo.kataoka.wt@renesas.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-23 12:17:12 +01:00
Wenkai Du
d132cb0a16 ASoC: Intel: Fix audio crash due to race condition in stream deletion
There is a race between sst_byt_stream_free() and sst_byt_get_stream()
if sst_byt_get_stream() called from sst_byt_irq_thread() context is
accessing the byt->stream_list while a stream is deleted from the list.

A stream is added to byt->stream_list in sst_byt_stream_new() and deleted in
sst_byt_stream_free(). sst_byt_get_stream() is always protected by
sst->spinlock, but the stream addition and deletion are not protected.

The patch adds spinlock to both stream addition and deletion.

[Jarkko: Same fix added to sst-haswell-ipc.c too]

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>
2014-04-23 12:11:20 +01:00
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
Kailang Yang
be8ef16a25 ALSA: hda/realtek - Change model name alias for ChromeOS
Chrome OS was use model name of alc283-dac-wcaps for loading model as default.
Change the model name to same as model name of Chrome OS for future support.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-08 10:38:56 +02:00
Uwe Kleine-König
ce816fa88c Kconfig: rename HAS_IOPORT to HAS_IOPORT_MAP
If the renamed symbol is defined lib/iomap.c implements ioport_map and
ioport_unmap and currently (nearly) all platforms define the port
accessor functions outb/inb and friend unconditionally.  So
HAS_IOPORT_MAP is the better name for this.

Consequently NO_IOPORT is renamed to NO_IOPORT_MAP.

The motivation for this change is to reintroduce a symbol HAS_IOPORT
that signals if outb/int et al are available.  I will address that at
least one merge window later though to keep surprises to a minimum and
catch new introductions of (HAS|NO)_IOPORT.

The changes in this commit were done using:

	$ git grep -l -E '(NO|HAS)_IOPORT' | xargs perl -p -i -e 's/\b((?:CONFIG_)?(?:NO|HAS)_IOPORT)\b/$1_MAP/'

Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Andrew Morton <akpm@linux-foundation.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2014-04-07 16:36:11 -07:00
Axel Lin
05b0006734 ASoC: da732x: Print correct major id
DA732X_ID_MAJOR_MASK is 0xF0, so the major id is
(reg & DA732X_ID_MAJOR_MASK) >> 4.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-07 12:57:56 +01:00
Kailang Yang
de9481cb40 ALSA: hda/realtek - Improve HP depop when system change power state on Chromebook
It is better to change Mic2-Vref to manual mode.
Manual control Mic2-Vref will solve pop noise issue.
It will improve pop noise for power on, power off, S3 and resume.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-07 12:32:52 +02:00
Axel Lin
a14bf88749 ASoC: cs42l52: Fix mask for REVID
BIT[0:2] of register 01h is REVID, so the mask for REVID should be 0x7.
Also updates the code to use CS42L52_CHIP_REV_MASK.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-07 10:48:50 +01:00
Linus Torvalds
cbda94e039 ARM: SoC: driver changes
These changes are mostly for ARM specific device drivers that either
 don't have an upstream maintainer, or that had the maintainer ask
 us to pick up the changes to avoid conflicts. A large chunk of this
 are clock drivers (bcm281xx, exynos, versatile, shmobile), aside from
 that, reset controllers for STi as well as a large rework of the
 Marvell Orion/EBU watchdog driver are notable.
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Merge tag 'drivers-3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc

Pull ARM SoC driver changes from Arnd Bergmann:
 "These changes are mostly for ARM specific device drivers that either
  don't have an upstream maintainer, or that had the maintainer ask us
  to pick up the changes to avoid conflicts.

  A large chunk of this are clock drivers (bcm281xx, exynos, versatile,
  shmobile), aside from that, reset controllers for STi as well as a
  large rework of the Marvell Orion/EBU watchdog driver are notable"

* tag 'drivers-3.15' of git://git.kernel.org/pub/scm/linux/kernel/git/arm/arm-soc: (99 commits)
  Revert "dts: socfpga: Add DTS entry for adding the stmmac glue layer for stmmac."
  Revert "net: stmmac: Add SOCFPGA glue driver"
  ARM: shmobile: r8a7791: Fix SCIFA3-5 clocks
  ARM: STi: Add reset controller support to mach-sti Kconfig
  drivers: reset: stih416: add softreset controller
  drivers: reset: stih415: add softreset controller
  drivers: reset: Reset controller driver for STiH416
  drivers: reset: Reset controller driver for STiH415
  drivers: reset: STi SoC system configuration reset controller support
  dts: socfpga: Add sysmgr node so the gmac can use to reference
  dts: socfpga: Add support for SD/MMC on the SOCFPGA platform
  reset: Add optional resets and stubs
  ARM: shmobile: r7s72100: fix bus clock calculation
  Power: Reset: Generalize qnap-poweroff to work on Synology devices.
  dts: socfpga: Update clock entry to support multiple parents
  ARM: socfpga: Update socfpga_defconfig
  dts: socfpga: Add DTS entry for adding the stmmac glue layer for stmmac.
  net: stmmac: Add SOCFPGA glue driver
  watchdog: orion_wdt: Use %pa to print 'phys_addr_t'
  drivers: cci: Export CCI PMU revision
  ...
2014-04-05 15:37:40 -07:00
Joe Perches
3af5d0524a sound/oss: Remove uncompilable DBG macro use
Most of it duplicates function tracing and one
of them has an uncompilable printf %P use.
Others have format/argument mismatches.

Remove unused DBG1 macro definition

Neaten uart401.c use of ok test around this
DBG macro removal.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-04 18:20:10 +02:00
Ondrej Zary
6ea0cae73d ALSA: ice1712: Save/restore routing and rate registers
Save/restore routing and rate registers during suspend/resume.
This fixes S/PDIF input being disabled after resume.
Tested with Audiophile 24/96.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-04 16:22:44 +02:00
Ondrej Zary
6e61246f5a ALSA: ice1712: restore AK4xxx volumes on resume
Also restore AK4xxx mixer volumes on resume for M-Audio ICE1712-based cards.
This fixes incorrect (sound working) zero mixer volumes after resume.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-04 16:22:34 +02:00
Christoph Jaeger
f78b1e0a8b ASoC: alc56(23|32): fix undefined return value of probing code
Commit 5d6be5aa ("ASoC: codec: Simplify ASoC probe code.") left variable
'ret', whose value is returned, uninitialized. Since it is not used
otherwise, remove it.

Signed-off-by: Christoph Jaeger <christophjaeger@linux.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-04 13:47:48 +01:00
Axel Lin
06b4b81305 ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value
Fix checking return value of regmap_read().
Also fix reporting the chip_id value. CS42XX8_CHIPID_CHIP_ID_MASK is 0xF0,
so the chip_id value is (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4).

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Paul Handrigan <paul.handrigan@cirrus.com>
Acked-by: Brian Austin <brian.austin@cirrus.com>
Acked-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-04 11:11:18 +01:00
Ondrej Zary
d272ccd0d0 ALSA: ice1712: Add S/PDIF suspend support for ICE1712-based M-Audio cards
Add S/PDIF suspend support for M-Audio cards based on ICE1712 chip.
Tested (playback only) on Audiophile 24/96. Capture will probably not work.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-03 15:00:15 +02:00
Ondrej Zary
9229bc1500 ALSA: cs8427: separate HW initialization
Separate HW initialization from device creation.
This is needed for suspend/resume support.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-03 14:59:48 +02:00
Takashi Iwai
415d555e6b ALSA: hda - Fix silent speaker output due to mute LED fixup
The recent fixups for HP laptops to support the mute LED made the
speaker output silent on some machines.  It turned out that they use
the NID 0x18 for the speaker while it's also used for controlling the
LED via VREF bits although the current driver code blindly assumes
that such a node is a mic pin (where 0x18 is usually so).

This patch fixes the problem by only changing the VREF bits and
keeping the other pin ctl bits.

Reported-and-tested-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-03 11:51:21 +02:00
Kailang Yang
8314f22589 ALSA: hda/realtek - Fixed single output machine get empty hp sense
If it only has single output of HP out of machine.
The driver parser will copy hp_pins to line_out_pins.
hp_pins will empty for alc283_init and alc283_shutup functions.
This will cause not have value for hp_pin_sense.
Add check line_out_type code will solve it .

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-04-03 11:48:33 +02:00
Linus Torvalds
467cbd207a Merge branch 'x86-nuke-platforms-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip
Pull x86 old platform removal from Peter Anvin:
 "This patchset removes support for several completely obsolete
  platforms, where the maintainers either have completely vanished or
  acked the removal.  For some of them it is questionable if there even
  exists functional specimens of the hardware"

Geert Uytterhoeven apparently thought this was a April Fool's pull request ;)

* 'x86-nuke-platforms-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tip/tip:
  x86, platforms: Remove NUMAQ
  x86, platforms: Remove SGI Visual Workstation
  x86, apic: Remove support for IBM Summit/EXA chipset
  x86, apic: Remove support for ia32-based Unisys ES7000
2014-04-02 13:15:58 -07:00
Stephen Warren
3534b842a8 ASoC: tlv320aic23: add an of_match table
Add a device tree match table. This serves to make the driver's support
of device tree more explicit. Perhaps the fallback for DT matching to
using the i2c_device_id table will go away one day, since it fails in
face of devices from different vendors with the same name.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-01 13:34:22 +01:00
Stephen Warren
2951f93f43 ASoC: max98090: add an of_match table
Add a device tree match table. This serves to make the driver's support
of device tree more explicit. Perhaps the fallback for DT matching to
using the i2c_device_id table will go away one day, since it fails in
face of devices from different vendors with the same name.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-01 12:46:12 +01:00
Stephen Warren
c31b0cb1f1 ASoC: alc5632: add an of_match table
Add a device tree match table. This serves to make the driver's support
of device tree more explicit. Perhaps the fallback for DT matching to
using the i2c_device_id table will go away one day, since it fails in
face of devices from different vendors with the same name.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-01 12:39:44 +01:00
Stephen Warren
e3efe3bedb ASoC: alc5632: fix uninit var in alc5632_probe()
alc5632_probe() returns ret, yet it is not initialized or set anywhere.
This ends up causing the function to appear to fail, and audio not to
work on the Toshiba AC100, with my compiler at least.

This function used to set ret in all cases, but recent cleanup removed
that.

Fixes: 5d6be5aa6b ("ASoC: codec: Simplify ASoC probe code.")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Reviewed-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-01 12:39:26 +01:00
Stephen Warren
03a620d8fa ASoC: rt5640: add an of_match table
Add a device tree match table. This serves to make the driver's support
of device tree more explicit. Perhaps the fallback for DT matching to
using the i2c_device_id table will go away one day, since it fails in
face of devices from different vendors with the same name.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-31 18:27:14 +01:00
Takashi Iwai
69dd89fd2b ASoC: Final updates for v3.15 merge window
A few more updates from last week - use of the tdm_slot mapping from
 Xiubo plus a few smaller fixes and cleanups.
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Merge tag 'asoc-v3.15-4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Final updates for v3.15 merge window

A few more updates from last week - use of the tdm_slot mapping from
Xiubo plus a few smaller fixes and cleanups.
2014-03-31 12:33:09 +02:00