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2013-01-11ALSA: usb-audio: Fix NULL dereference by access to non-existing substreamTakashi Iwai1-0/+10
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>
2013-01-10Merge tag 'asoc-fix-3.8-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linusTakashi Iwai15-118/+189
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.
2013-01-10Merge remote-tracking branch 'asoc/fix/wm5100' into tmpMark Brown1-6/+0
2013-01-10Merge remote-tracking branch 'asoc/fix/wm2200' into tmpMark Brown1-7/+1
2013-01-10Merge remote-tracking branch 'asoc/fix/wm2000' into tmpMark Brown1-2/+2
2013-01-10Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmpMark Brown1-1/+22
2013-01-10Merge remote-tracking branch 'asoc/fix/sta529' into tmpMark Brown1-4/+5
2013-01-10Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmpMark Brown1-2/+2
2013-01-10Merge remote-tracking branch 'asoc/fix/pxa' into tmpMark Brown1-3/+23
2013-01-10Merge remote-tracking branch 'asoc/fix/lm49453' into tmpMark Brown1-66/+40
2013-01-10Merge remote-tracking branch 'asoc/fix/cs42l52' into tmpMark Brown1-3/+1
2013-01-10Merge remote-tracking branch 'asoc/fix/cs4271' into tmpMark Brown1-3/+3
2013-01-10Merge remote-tracking branch 'asoc/fix/core' into tmpMark Brown2-3/+33
2013-01-10Merge remote-tracking branch 'asoc/fix/arizona' into tmpMark Brown3-18/+57
2013-01-10ALSA: hda - Add support of new codec ALC284Kailang Yang1-1/+23
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>
2013-01-10ALSA: usb-audio: Make ebox44_table staticSachin Kamat1-1/+1
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>
2013-01-09ALSA: hdspm - Fix wordclock status on AES32Andre Schramm1-5/+12
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>
2013-01-09Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"David Henningsson1-16/+0
This reverts commit 697c373e34613609cb5450f98b91fefb6e910588. 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>
2013-01-09ALSA: hda - Disable runtime D3 for Intel CPT & coTakashi Iwai1-5/+8
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>
2013-01-08ALSA: pxa27x: fix ac97 warm resetMike Dunn1-1/+17
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
2013-01-08ALSA: pxa27x: fix ac97 cold resetMike Dunn1-2/+6
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
2013-01-07ASoC: wm_adsp: Ensure that block writes are from DMA aligned addressesMark Brown1-1/+22
Otherwise we won't run correctly on systems that require this for larger data transfers. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-07ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)David Henningsson1-0/+30
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>
2013-01-07ALSA: au88x0: fix incorrect left shiftNickolai Zeldovich1-1/+1
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>
2013-01-04ASoC: wm2000: Fix sense of speech clarity enableMark Brown1-2/+2
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-01-04ASoC: wm5100: Remove DSP B and left justified formatsMark Brown1-6/+0
These are not supported Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-01-04ASoC: arizona: Remove DSP B and left justified AIF modesMark Brown1-6/+0
These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-01-04ASoC: wm2200: Remove DSP B and left justified AIF modesMark Brown1-6/+0
These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-01-04sound: oss/pas2: Fix possible access out of arrayAsim Kadav1-0/+5
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>
2013-01-04ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirkDamien Zammit4-15/+8
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>
2013-01-03ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirksAlexander Schremmer1-0/+22
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>
2013-01-03ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrolDavid Henningsson1-1/+1
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>
2013-01-02ASoC: wm5102: Improve speaker enable performanceMark Brown1-2/+46
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-12-27ASoC: core: fix the memory leak in case of remove_aux_dev()Chuansheng Liu1-1/+1
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>
2012-12-27ASoC: core: fix the memory leak in case of device_add() failureChuansheng Liu1-0/+2
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>
2012-12-24ASoC: cs42l52: Catch no-match case in cs42l52_get_clkAxel Lin1-3/+1
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>
2012-12-24ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-BMR.Swami.Reddy@ti.com1-46/+15
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>
2012-12-24ASoC: lm49453: Fix adc, mic and sidetone volume rangesMR.Swami.Reddy@ti.com1-19/+24
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>
2012-12-24ASoC: arizona: Correct FLL source definitionsMark Brown1-9/+9
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
2012-12-24ASoC: arizona: Do proper shift for setting AIF rateAxel Lin1-1/+2
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
2012-12-24ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()Axel Lin1-1/+1
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>
2012-12-24ASoC: sgtl5000: Fix maximum value for microphone gainFabio Estevam1-2/+2
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>
2012-12-24ASoC: soc-core: Remove unused 'ret' variableFabio Estevam1-1/+0
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>
2012-12-21ASoC: wm2200: Fix setting dai format in wm2200_set_fmtAxel Lin1-1/+1
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
2012-12-20ASoC: core: Fix SOC_DOUBLE_RANGE() macrosMark Brown1-2/+30
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>
2012-12-20ASoC: sta529: Fix update register bits in sta529_set_dai_fmtAxel Lin1-4/+5
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
2012-12-20ASoC: pcm: allow backend hardware to be freed in pause statePatrick Lai1-0/+1
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
2012-12-20Merge tag 'sound-3.8' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/soundLinus Torvalds14-76/+365
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
2012-12-19ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:Damien Zammit4-0/+183
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>
2012-12-18ALSA: HDA: Fix sound resume hangDaniel J Blueman1-2/+2
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>
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