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* ALSA: hda/hdmi - Work around "alsactl restore" errorsTakashi Iwai2013-01-151-1/+1
| | | | | | | | | | | | | | | | | When "alsactl restore" is performed on HDMI codecs, it tries to restore the channel map value since the channel map controls are writable. But hdmi_chmap_ctl_put() returns -EBADFD when no PCM stream is assigned yet, and this results in an error message from alsactl. Although the error is harmless, it's certainly ugly and can be regarded as a regression. As a workaround, this patch changes the return code in such a case to be zero for making others happy. (A slight excuse is: when the chmap is changed through the proper alsa-lib API, the PCM status is checked there anyway, so we don't have to be too strict in the kernel side.) Cc: <stable@vger.kernel.org> [v3.7+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: selector map for M-Audio FT C400Eldad Zack2013-01-141-0/+13
| | | | | | | | Add names of the clock sources for the M-Audio Fast Track C400. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: M-Audio FT C400 skip packet quirkEldad Zack2013-01-141-0/+11
| | | | | | | | Attain constant real-world latency by skipping 16 data packets. The number of packets to be skipped was found by trial and error. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: correct M-Audio C400 clock source quirkEldad Zack2013-01-141-2/+2
| | | | | | | | | | | | | | | Taking another look at the C400 descriptors, I see now that there is a clock selector (0x80) for this device. Right now, the clock source points to the internal clock (0x81), which is also valid. When the external clock source (0x82) is selected in the mixer, and the rates mismatch (if it's free-running it is fixed to 48KHz), xruns will occur. Set the clock ID to the clock selector unit (0x81), which then allows the validation code to function correctly. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb - fix race in creation of M-Audio Fast track pro driverDavid Henningsson2013-01-141-3/+5
| | | | | | | | | | | | | | | | | | | A patch in the 3.2 kernel caused regression with hotplugging the M-Audio Fast track pro, or sound after suspend. I don't have the device so I haven't done a full analysis, but it seems userspace (both udev and pulseaudio) got confused when a card was created, immediately destroyed, and then created again. However, at least one person in the bug report (martin djfun) reports that this patch resolves the issue for him. It also leaves a message in the log: "snd-usb-audio: probe of 1-1.1:1.1 failed with error -5" which is a bit misleading. It is better than non-working audio, but maybe there's a more elegant solution? BugLink: https://bugs.launchpad.net/bugs/1095315 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: Fix NULL dereference by access to non-existing substreamTakashi Iwai2013-01-111-0/+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>
* Merge tag 'asoc-fix-3.8-rc2' of ↵Takashi Iwai2013-01-1015-118/+189
|\ | | | | | | | | | | | | | | | | 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.
| * Merge remote-tracking branch 'asoc/fix/wm5100' into tmpMark Brown2013-01-101-6/+0
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| | * ASoC: wm5100: Remove DSP B and left justified formatsMark Brown2013-01-041-6/+0
| | | | | | | | | | | | | | | | | | | | | These are not supported Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | Merge remote-tracking branch 'asoc/fix/wm2200' into tmpMark Brown2013-01-101-7/+1
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| | * | ASoC: wm2200: Remove DSP B and left justified AIF modesMark Brown2013-01-041-6/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * | ASoC: wm2200: Fix setting dai format in wm2200_set_fmtAxel Lin2012-12-211-1/+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
| * | | Merge remote-tracking branch 'asoc/fix/wm2000' into tmpMark Brown2013-01-101-2/+2
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| | * | | ASoC: wm2000: Fix sense of speech clarity enableMark Brown2013-01-041-2/+2
| | | |/ | | |/| | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | Merge remote-tracking branch 'asoc/fix/wm-adsp' into tmpMark Brown2013-01-101-1/+22
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| | * | | ASoC: wm_adsp: Ensure that block writes are from DMA aligned addressesMark Brown2013-01-071-1/+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>
| * | | Merge remote-tracking branch 'asoc/fix/sta529' into tmpMark Brown2013-01-101-4/+5
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| | * | | ASoC: sta529: Fix update register bits in sta529_set_dai_fmtAxel Lin2012-12-201-4/+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
| * | | Merge remote-tracking branch 'asoc/fix/sgtl5000' into tmpMark Brown2013-01-101-2/+2
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| | * | | ASoC: sgtl5000: Fix maximum value for microphone gainFabio Estevam2012-12-241-2/+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>
| * | | Merge remote-tracking branch 'asoc/fix/pxa' into tmpMark Brown2013-01-101-3/+23
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| | * | | ALSA: pxa27x: fix ac97 warm resetMike Dunn2013-01-081-1/+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
| | * | | ALSA: pxa27x: fix ac97 cold resetMike Dunn2013-01-081-2/+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
| * | | Merge remote-tracking branch 'asoc/fix/lm49453' into tmpMark Brown2013-01-101-66/+40
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| | * | | ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-BMR.Swami.Reddy@ti.com2012-12-241-46/+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>
| | * | | ASoC: lm49453: Fix adc, mic and sidetone volume rangesMR.Swami.Reddy@ti.com2012-12-241-19/+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>
| | * | | ASoC: lm49453: Fix mask for setting mode bit in lm49453_set_dai_fmt()Axel Lin2012-12-241-1/+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>
| * | | Merge remote-tracking branch 'asoc/fix/cs42l52' into tmpMark Brown2013-01-101-3/+1
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| | * | | ASoC: cs42l52: Catch no-match case in cs42l52_get_clkAxel Lin2012-12-241-3/+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>
| * | | Merge remote-tracking branch 'asoc/fix/cs4271' into tmpMark Brown2013-01-101-3/+3
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| | * | | ASoC: cs4271: fix property checkDaniel Mack2012-12-021-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The driver had the property check for 'cirrus,amutec_eq_bmutec' the wrong way around. That happens if you misspell the property in the bindings during tests. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | ASoC: cs4271: fix sparse warningDaniel Mack2012-12-021-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Make the flag in the pdata of type bool to fix a sparse warning. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Fengguang Wu <fengguang.wu@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | Merge remote-tracking branch 'asoc/fix/core' into tmpMark Brown2013-01-102-3/+33
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| | * | | | ASoC: core: fix the memory leak in case of remove_aux_dev()Chuansheng Liu2012-12-271-1/+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>
| | * | | | ASoC: core: fix the memory leak in case of device_add() failureChuansheng Liu2012-12-271-0/+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>
| | * | | | ASoC: soc-core: Remove unused 'ret' variableFabio Estevam2012-12-241-1/+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>
| | * | | | ASoC: core: Fix SOC_DOUBLE_RANGE() macrosMark Brown2012-12-201-2/+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>
| | * | | | ASoC: pcm: allow backend hardware to be freed in pause statePatrick Lai2012-12-201-0/+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
| * | | | Merge remote-tracking branch 'asoc/fix/arizona' into tmpMark Brown2013-01-103-18/+57
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| | * | | | ASoC: arizona: Remove DSP B and left justified AIF modesMark Brown2013-01-041-6/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * | | | ASoC: wm5102: Improve speaker enable performanceMark Brown2013-01-021-2/+46
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | ASoC: arizona: Correct FLL source definitionsMark Brown2012-12-241-9/+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
| | * | | | ASoC: arizona: Do proper shift for setting AIF rateAxel Lin2012-12-241-1/+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
* | | | | ALSA: hda - Add support of new codec ALC284Kailang Yang2013-01-101-1/+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>
* | | | | ALSA: usb-audio: Make ebox44_table staticSachin Kamat2013-01-101-1/+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>
* | | | | ALSA: hdspm - Fix wordclock status on AES32Andre Schramm2013-01-091-5/+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>
* | | | | Revert "ALSA: hda - Shut up pins at power-saving mode with Conexnat codecs"David Henningsson2013-01-091-16/+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>
* | | | | ALSA: hda - Disable runtime D3 for Intel CPT & coTakashi Iwai2013-01-091-5/+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>
* | | | ALSA: hda - add mute LED for HP Pavilion 17 (Realtek codec)David Henningsson2013-01-071-0/+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>
* | | | ALSA: au88x0: fix incorrect left shiftNickolai Zeldovich2013-01-071-1/+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>