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* Merge remote-tracking branch 'asoc/fix/sta529' into tmpMark Brown2013-01-10
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| * ASoC: sta529: Fix update register bits in sta529_set_dai_fmtAxel Lin2012-12-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | ASoC: sgtl5000: Fix maximum value for microphone gainFabio Estevam2012-12-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | | ALSA: pxa27x: fix ac97 warm resetMike Dunn2013-01-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Dunn2013-01-08
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | | ASoC: lm49453: Update lm49453_reg_defs values as per LM49453 HW revision-BMR.Swami.Reddy@ti.com2012-12-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | 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.com2012-12-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Lin2012-12-24
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | | ASoC: cs42l52: Catch no-match case in cs42l52_get_clkAxel Lin2012-12-24
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | | ASoC: cs4271: fix property checkDaniel Mack2012-12-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Mack2012-12-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | | | ASoC: core: fix the memory leak in case of remove_aux_dev()Chuansheng Liu2012-12-27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Liu2012-12-27
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Estevam2012-12-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2012-12-20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Lai2012-12-20
| | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Brown2013-01-10
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| * | | | ASoC: arizona: Remove DSP B and left justified AIF modesMark Brown2013-01-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These are not supported. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | ASoC: wm5102: Improve speaker enable performanceMark Brown2013-01-02
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | ASoC: arizona: Correct FLL source definitionsMark Brown2012-12-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Lin2012-12-24
| | |_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 mute LED for HP Pavilion 17 (Realtek codec)David Henningsson2013-01-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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 Zeldovich2013-01-07
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | sound: oss/pas2: Fix possible access out of arrayAsim Kadav2013-01-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: usb-audio: Fix kernel panic of Digidesign Mbox2 quirkDamien Zammit2013-01-04
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: usb-audio: Add support for Creative BT-D1 via usb sound quirksAlexander Schremmer2013-01-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | | ALSA: hda - Switch "On" and "Off" for "Mute-LED Mode" kcontrolDavid Henningsson2013-01-03
|/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
* | | Merge tag 'sound-3.8' of ↵Linus Torvalds2012-12-20
|\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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
| * | | ALSA: usb-audio: Support for Digidesign Mbox 2 USB sound card:Damien Zammit2012-12-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: HDA: Fix sound resume hangDaniel J Blueman2012-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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>
| * | | ALSA: hda - bug fix for invalid connection list of Haswell HDMI codec pinsMengdong Lin2012-12-18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Haswell HDMI codec pins may report invalid connection list entries, which will cause failure to play audio via HDMI or Display Port. So this patch adds fixup for Haswell to workaround this hardware issue: enable DP1.2 mode and override the pins' connection list entries with proper value. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Xingchao Wang <xingchao.wang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - Fix the wrong pincaps set in ALC861VD dallas/hp fixupTakashi Iwai2012-12-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The workaround to force VREF50 for dallas/hp model with ALC861VD was introduced in commit 8fdcb6fe4204bdb4c6991652717ab5063751414e, but it contained wrong pincap override bits. This patch fixes to exclude VREF80 pincap bit correctly. Cc: <stable@vger.kernel.org> [v3.2+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - Set codec->single_adc_amp flag for Realtek codecsTakashi Iwai2012-12-17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It turned out that Realtek codecs (ALC260, etc) with input amps in audio-input widgets don't handle the multiple individual input amps. Thus we need to set codec->single_adc_amp flag for them in general. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge tag 'asoc-3.8p1' of ↵Takashi Iwai2012-12-17
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: More updates for v3.8 Nothing terribly exciting here, just small localised changes. As well as fixes there are a couple of Cirrus changes and one devm_ change which were in prior to the merge window but got missed from the original pull to Takashi.
| | * | Merge remote-tracking branch 'asoc/topic/tpa6130a2' into asoc-nextMark Brown2012-12-15
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| | | * | ASoC: tpa6130a2: Use devm_* APIsSachin Kamat2012-12-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Converted to use devm_gpio_request and devm_regulator_get APIs. These are device managed and make error handling and cleanup a bit simpler. Cc: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | Merge remote-tracking branch 'asoc/topic/log' into asoc-nextMark Brown2012-12-15
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| | | * | | ASoC: core: Fix splitting of log messagesMark Brown2012-12-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Don't wrap log messages over multiple lines, it makes them hard to grep for. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge remote-tracking branch 'asoc/topic/cs42l73' into asoc-nextMark Brown2012-12-15
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| | | * | | | ASoC: cs42l73: Change VSPIN/VSPOUT to VSPINOUTPaul Handrigan2012-12-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since VSP only has one power bit, only provide one DAPM widget. Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: cs42l73: Add DAPM events for power down.Paul Handrigan2012-12-09
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add power down delays between setting PDN and MCLKDIS for spk amp, spklo amp, and ear amp. Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: cs42l73: Add DMIC's as DAPM inputs.Paul Handrigan2012-12-09
| | | | |/ / | | | |/| | | | | | | | | | | | | | | | | | | | Signed-off-by: Paul Handrigan <Paul.Handrigan@cirrus.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | Merge remote-tracking branch 'asoc/topic/core' into asoc-nextMark Brown2012-12-15
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| | | * | | | ASoC: Prevent pop_wait overwriteMisael Lopez Cruz2012-12-15
| | | |/ / / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | pop_wait is used to determine if a deferred playback close needs to be cancelled when the a PCM is open or if after the power-down delay expires it needs to run. pop_wait is associated with the CODEC DAI, so the CODEC DAI must be unique. This holds true for most CODECs, except for the dummy CODEC and its DAI. In DAI links with non-unique dummy CODECs (e.g. front-ends), pop_wait can be overwritten by another DAI link using also a dummy CODEC. Failure to cancel a deferred close can cause mute due to the DAPM STOP event sent in the deferred work. One scenario where pop_wait is overwritten and causing mute is below (where hw:0,0 and hw:0,1 are two front-ends with default pmdown_time = 5 secs): aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1 sleep 1 aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 & aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE Since CODECs may not be unique, pop_wait is moved to the PCM runtime structure. Creating separate dummy CODECs for each DAI link can also solve the problem, but at this point it's only pop_wait variable in the CODEC DAI that has negative effects by not being unique. Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>