Commit Graph

14286 Commits

Author SHA1 Message Date
Takashi Iwai 31be5425d7 ALSA: usb-audio: Fix NULL dereference by access to non-existing substream
The commit [0d9741c0: ALSA: usb-audio: sync ep init fix for
audioformat mismatch] introduced the correction of parameters to be
set for sync EP.  But since the new code assumes that the sync EP is
always paired with the data EP of another direction, it triggers Oops
when a device only with a single direction is used.

This patch adds a proper check of sync EP type and the presence of the
paired substream for avoiding the crash.

Reported-and-tested-by: Jens Axboe <axboe@kernel.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-11 11:12:17 +01:00
Takashi Iwai c18ab0bac4 ASoC: Fixes for v3.8
Nothing terribly exciting here except for the DOUBLE_RANGE fix which
 just hadn't worked before, nobody noticed due to lack of use.
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Merge tag 'asoc-fix-3.8-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v3.8

Nothing terribly exciting here except for the DOUBLE_RANGE fix which
just hadn't worked before, nobody noticed due to lack of use.
2013-01-10 17:41:54 +01:00
Mark Brown 49a170bcf2 Merge remote-tracking branch 'asoc/fix/wm5100' into tmp 2013-01-10 12:22:30 +00:00
Mark Brown 921c038d87 Merge remote-tracking branch 'asoc/fix/wm2200' into tmp 2013-01-10 12:22:29 +00:00
Mark Brown 28f2675db8 Merge remote-tracking branch 'asoc/fix/wm2000' into tmp 2013-01-10 12:22:26 +00:00
Mark Brown 92a9d1524e Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmp 2013-01-10 12:22:25 +00:00
Mark Brown a883eae513 Merge remote-tracking branch 'asoc/fix/sta529' into tmp 2013-01-10 12:22:22 +00:00
Mark Brown fd2eab87a2 Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmp 2013-01-10 12:22:17 +00:00
Mark Brown 87fee06c5b Merge remote-tracking branch 'asoc/fix/pxa' into tmp 2013-01-10 12:22:16 +00:00
Mark Brown c31b71de6f Merge remote-tracking branch 'asoc/fix/lm49453' into tmp 2013-01-10 12:22:15 +00:00
Mark Brown fa17cb4a02 Merge remote-tracking branch 'asoc/fix/cs42l52' into tmp 2013-01-10 12:22:14 +00:00
Mark Brown 587691ea39 Merge remote-tracking branch 'asoc/fix/cs4271' into tmp 2013-01-10 12:22:11 +00:00
Mark Brown a18a31a161 Merge remote-tracking branch 'asoc/fix/core' into tmp 2013-01-10 12:21:50 +00:00
Mark Brown ae1abb0c3b Merge remote-tracking branch 'asoc/fix/arizona' into tmp 2013-01-10 12:21:42 +00:00
Kailang Yang 065380f088 ALSA: hda - Add support of new codec ALC284
Added the support for a new codec ALC284, which is compatible with
ALC269.  Also add more codec variants to handle the SSID check
properly.

Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:27:46 +01:00
Sachin Kamat e8e7da23c9 ALSA: usb-audio: Make ebox44_table static
Fixes the following sparse warning:
sound/usb/mixer_quirks.c:1209:23: warning:
symbol 'ebox44_table' was not declared. Should it be static?

Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-10 10:22:25 +01:00
Andre Schramm 56bde0f328 ALSA: hdspm - Fix wordclock status on AES32
Use correct bitmask for AES32 cards to determine wordclock lock state,
add missing bitmask for sync check and make output of the corresponding
control and /proc coherent.

Signed-off-by: Andre Schramm <andre.schramm@iosono-sound.com>
Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 16:59:24 +01:00
David Henningsson 7ed4165e2d Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"
This reverts commit 697c373e34.

The original patch was meant to remove clicking, but in fact caused even
more clicking instead.

Thanks to c4pp4 for doing most of the work with this bug.

BugLink: https://bugs.launchpad.net/bugs/886975
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:03:38 +01:00
Takashi Iwai d7dab4dbbb ALSA: hda - Disable runtime D3 for Intel CPT & co
We've got a few bug reports that the runtime D3 results in the dead
HD-audio controller.  It seems that the problem is in a deeper level
than the sound driver itself, so as a temporal solution, disable the
feature for these controllers again.

Reported-and-tested-by: Vincent Blut <vincent.debian@free.fr>
Reported-and-tested-by: Maurizio Avogadro <mavoga@gmail.com>
Cc: <stable@vger.kernel.org> [v3.7]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-09 11:00:08 +01:00
Mike Dunn 3b4bc7bccc ALSA: pxa27x: fix ac97 warm reset
This patch fixes some code that implements a work-around to a hardware bug in
the ac97 controller on the pxa27x.  A bug in the controller's warm reset
functionality requires that the mfp used by the controller as the AC97_nRESET
line be temporarily reconfigured as a generic output gpio (AF0) and manually
held high for the duration of the warm reset cycle.  This is what was done in
the original code, but it was broken long ago by commit fb1bf8cd
    ([ARM] pxa: introduce processor specific pxa27x_assert_ac97reset())
which changed the mfp to a GPIO input instead of a high output.

The fix requires the ac97 controller to obtain the gpio via gpio_request_one(),
with arguments that configure the gpio as an output initially driven high.

Tested on a palm treo 680 machine.  Reportedly, this broken code only prevents a
warm reset on hardware that lacks a pull-up on the line, which appears to be the
case for me.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Signed-off-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-08 11:27:35 +00:00
Mike Dunn 41b645c862 ALSA: pxa27x: fix ac97 cold reset
Cold reset on the pxa27x currently fails and

     pxa2xx_ac97_try_cold_reset: cold reset timeout (GSR=0x44)

appears in the kernel log.  Through trial-and-error (the pxa270 developer's
manual is mostly incoherent on the topic of ac97 reset), I got cold reset to
complete by setting the WARM_RST bit in the GCR register (and later noticed that
pxa3xx does this for cold reset as well).  Also, a timeout loop is needed to
wait for the reset to complete.

Tested on a palm treo 680 machine.

Signed-off-by: Mike Dunn <mikedunn@newsguy.com>
Acked-by: Igor Grinberg <grinberg@compulab.co.il>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-08 11:27:27 +00:00
Mark Brown a76fefab5c ASoC: wm_adsp: Ensure that block writes are from DMA aligned addresses
Otherwise we won't run correctly on systems that require this for larger
data transfers.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-07 19:13:35 +00:00
David Henningsson 6d3cd5d444 ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)
The mute LED is in this case connected to the Mic1 VREF.

The machine also exposes the following string in BIOS:
"HP_Mute_LED_0_A", so if more machines are coming, it probably
makes sense to try to do something more generic, like for the
IDT codec.

Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1096789
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-07 17:29:55 +01:00
Nickolai Zeldovich 61ed1dca16 ALSA: au88x0: fix incorrect left shift
vortex_wt_setdsout performs bit-negation on the bit position (wt&0x1f)
rather than on the resulting bitmask.  This code is never actually
invoked (vortex_wt_setdsout is always called with en=1), so this does
not currently cause any problem, and this patch is simply cleanup.

Signed-off-by: Nickolai Zeldovich <nickolai@csail.mit.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-07 09:33:56 +01:00
Mark Brown 267f8fa2e1 ASoC: wm2000: Fix sense of speech clarity enable
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 21:19:42 +00:00
Mark Brown 5f960294e2 ASoC: wm5100: Remove DSP B and left justified formats
These are not supported

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 21:06:08 +00:00
Mark Brown d71753e22b ASoC: arizona: Remove DSP B and left justified AIF modes
These are not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 11:33:22 +00:00
Mark Brown 0cc411b934 ASoC: wm2200: Remove DSP B and left justified AIF modes
These are not supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2013-01-04 11:31:57 +00:00
Asim Kadav dc30a43690 sound: oss/pas2: Fix possible access out of array
Added a fix for hardware dependence bug where a sound card failure
should not result in reading beyond array memory index.

Signed-off-by: Asim Kadav <kadav@cs.wisc.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 10:38:27 +01:00
Damien Zammit b7b435e81b ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirk
This patch is based on 3.8-rc1. It fixes two things:
1) A kernel panic caused by incorrect allocation of a u8 variable
   "bootresponse".
2) A noisy dmesg (urb status -32) caused by broken pipe to an
   invalid midi endpoint.

It is also a little cleaner because there is no need for a new
QUIRK_MIDI type as suggested by kernel developers, since the device
follows exactly the MIDIMAN protocol.

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-04 09:53:17 +01:00
Alexander Schremmer 8f7f3ab15e ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirks
Support the Creative BT-D1 Bluetooth USB audio device. Before this
patch, Linux had trouble finding the correct USB descriptors and bailed
out with these messages:

 no or invalid class specific endpoint descriptor

Now it still prints these messages on hotplug:

 snd-usb-audio: probe of ...:1.0 failed with error -5
 snd-usb-audio: probe of ...:1.2 failed with error -5
 snd-usb-audio: probe of ...:1.3 failed with error -5

But the device works correctly, including the HID support.

The patch is diff'ed against 3.8-rc1 but should apply to older kernels
as well.

Signed-off-by: Alexander Schremmer <alex@alexanderweb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:26:48 +01:00
David Henningsson c86c2d440c ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrol
The vmaster hook sends 1 for enabled/unmuted and 0 for disabled/muted,
but "Mute-LED Mode" being "On" refers to the LED being on, not the
volume being on.
Therefore "On" and "Off" should be switched.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-01-03 14:22:34 +01:00
Mark Brown 1b8d52e63c ASoC: wm5102: Improve speaker enable performance
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-02 13:08:42 +00:00
Chuansheng Liu d3bf156125 ASoC: core: fix the memory leak in case of remove_aux_dev()
When probing aux_dev, initializing is as below:
device_initialize()
device_add()

So when remove aux_dev, we need do as below:
device_del()
device_put()
Otherwise, the rtd_release() will not be called.

So here using device_unregister() to replace device_del(),
like the action in soc_remove_link_dais().
Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:14:43 +00:00
Chuansheng Liu 865df9cb12 ASoC: core: fix the memory leak in case of device_add() failure
After called device_initialize(), even device_add() returns
error, we still need use the put_device() to release the reference
to call rtd_release(), which will do the free() action.

Signed-off-by: liu chuansheng <chuansheng.liu@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27 16:14:43 +00:00
Axel Lin 3271a4fc7d ASoC: cs42l52: Catch no-match case in cs42l52_get_clk
In the case of no-match, return -EINVAL instead of 0.

Since we assign i to ret in the for loop, ret always less than
ARRAY_SIZE(clk_map_table). Thus remove the boundary checking for ret.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 16:02:19 +00:00
MR.Swami.Reddy@ti.com 9dc754dfa7 ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-B
Update lm49453_reg_defs values as per LM49453 HW revision-B

Signed-off-by: M R Swami Reddy <mr.swami.reddy@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:45:10 +00:00
MR.Swami.Reddy@ti.com 88ac43924b ASoC: lm49453: Fix adc, mic and sidetone volume ranges
Add adc, mic, sidetone volume ranges and appropriately added the controls.
Fix the DAC HP/EP/LS/LO/HA maximum gain values.

Signed-off-by: MR Swami Reddy <mr.swami.reddy@ti.com>
Tested-by: Vinod Koul <vinod.koul@intel.com>

--
 sound/soc/codecs/lm49453.c |   43 ++++++++++++++++++++++++-------------------
 1 files changed, 24 insertions(+), 19 deletions(-)
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:43:56 +00:00
Mark Brown a8c02db029 ASoC: arizona: Correct FLL source definitions
The FLL source constants were numbered as a simple enumeration but were
being used in the code as direct values to be written to the registers.
Renumber the constants to reflect the usage.

Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-24 15:41:44 +00:00
Axel Lin 7110a287ff ASoC: arizona: Do proper shift for setting AIF rate
ARIZONA_AIF1_RATE_MASK is 0x7800 /* AIF1_RATE - [14:11] */
Thus we need left shift ARIZONA_AIF1_RATE_SHIFT when setting aif1 rate.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-24 15:41:44 +00:00
Axel Lin ec20f2f8d3 ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()
The mode variable is either 0 or 1.
To update mode setting, the mask should be BIT(0) rather than BIT(1).

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Tested-by: Omair M. Abdullah <omair.m.abdullah@intel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:32:28 +00:00
Fabio Estevam b50684da6c ASoC: sgtl5000: Fix maximum value for microphone gain
sgtl5000 microphone gain only has 2 bits of resolution, so maximum value is 3.

From Eric Nelson:
"We also found that for the microphones we have here (commodity PC boom mics) a
default value of 2 for the gain gives the best results."

So change the default microphone gain as well.

Signed-off-by: Eric Nelson <eric.nelson@boundarydevices.com>
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:31:25 +00:00
Fabio Estevam 5db1bc1892 ASoC: soc-core: Remove unused 'ret' variable
commit 9bde4f0b1c (ASoC: core: Fix SOC_DOUBLE_RANGE() macros) introduced
the following build warning:

sound/soc/soc-core.c:2999:6: warning: unused variable 'ret' [-Wunused-variable]

Remove the unused 'ret' variable.

Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-24 15:20:16 +00:00
Axel Lin 2a5f431592 ASoC: wm2200: Fix setting dai format in wm2200_set_fmt
According to the defines in wm2200.h:
/*
 * R1284 (0x504) - Audio IF 1_5
 */

We should not left shift 1 bit for fmt_val when setting dai format.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-21 09:32:20 +00:00
Mark Brown 9bde4f0b1c ASoC: core: Fix SOC_DOUBLE_RANGE() macros
Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2012-12-20 17:46:55 +00:00
Axel Lin ad1937cdd5 ASoC: sta529: Fix update register bits in sta529_set_dai_fmt
Both the mask and mode settings are wrong in current code.

According to the datasheet:

S2PCFG0 (0x0A)
BIT[3:1] DATA_FORMAT
        serial interface protocol format:
        000: left Justified
        001: I2S (default)
        010: right justified
        100: PCM no delay
        101: PCM delay
        111: DSP

Thus fixes the defines for LEFT_J_DATA_FORMAT, I2S_DATA_FORMAT, and
RIGHT_J_DATA_FORMAT.
Also adds define for DATA_FORMAT_MSK.

Signed-off-by: Axel Lin <axel.lin@ingics.com>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-20 16:01:26 +00:00
Patrick Lai 08b27848da ASoC: pcm: allow backend hardware to be freed in pause state
When front-end PCM session is in paused state, back-end
PCM session will be put in paused state as well if given
front-end PCM session is the only client of given back-end.
Then, application closes front-end PCM session, DPCM
framework will not allow back-end enters HW_FREE state
so back-end will never get shutdown completely.

Signed-off-by: Patrick Lai <plai@codeaurora.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
2012-12-20 15:59:46 +00:00
Linus Torvalds 03c850ec32 Sound fixes for 3.8-rc1
This update contains overall only driver-specific fixes.
 Slightly large LOC are seen in usb-audio driver for a couple of new
 device quirks and cs42l71 ASoC driver for enhanced features.
 The others are a few small (regression) fixes HD-audio, and yet other
 small / trival ASoC fixes.
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Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This update contains overall only driver-specific fixes.  Slightly
  large LOC are seen in usb-audio driver for a couple of new device
  quirks and cs42l71 ASoC driver for enhanced features.  The others are
  a few small (regression) fixes HD-audio, and yet other small / trival
  ASoC fixes."

* tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
  ALSA: HDA: Fix sound resume hang
  ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pins
  ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixup
  ALSA: hda - Set codec->single_adc_amp flag for Realtek codecs
  ASoC: atmel-ssc: change disable to disable in dts node
  ASoC: Prevent pop_wait overwrite
  ALSA: usb-audio: ignore-quirk for HP Wireless Audio
  ALSA: hda - Always turn on pins for HDMI/DP
  ALSA: hda - Fix pin configuration of HP Pavilion dv7
  ASoC: core: Fix splitting of log messages
  ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUT
  ASoC: cs42l73: Add DAPM events for power down.
  ASoC: cs42l73: Add DMIC's as DAPM inputs.
  ASoC: sigmadsp: Fix endianness conversion issue
  ASoC: tpa6130a2: Use devm_* APIs
2012-12-20 07:52:13 -08:00
Damien Zammit cb99864d40 ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:
This patch is the result of a lot of trial and error, since there are no specs
available for the device.

Full duplex support is provided, i.e. playback and recording in stereo.
The format is hardcoded at 48000Hz @ 24 bit, which is the maximum that the
device supports.  Also, MIDI in and MIDI out both work.

Users will notice that the S/PDIF light also flashes when playback or recording
is active.  I believe this means that S/PDIF input/output is simultaneously
activated with the analogue i/o during use.
But this particular functionality remains untested.

Note that this particular version of the patch is so far untested on the
physical hardware because I have not compiled a full kernel with the changes.
However, extensive testing has been done by many users of the hardware
who believe other versions of my patch have worked since circa 2009.

[Modified to make a function static by tiwai]

Signed-off-by: Damien Zammit <damien@zamaudio.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-19 11:27:22 +01:00
Daniel J Blueman 44728e97c3 ALSA: HDA: Fix sound resume hang
Resuming a switcheroo'd HDA controller hangs since the completion
is one-shot (thus works the first time). Fix by using completions
that explictly need rearming, so remain fired before.

Signed-off-by: Daniel J Blueman <daniel@quora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-12-18 17:07:11 +01:00