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2012-01-09ENGR00160605 Update ALSA Headset detect codeMahesh Mahadevan
Update Headset detect code incase no dedicated pin and detect is always activated Signed-off-by: Mahesh Mahadevan <r9aadq@freescale.com>
2012-01-09ENGR00156745 MX6Q ESAI: Playback and record can't start up concurrentlyLionel Xu
Proper flag setting and placement should be used to avoid function hw_param called multiple times when playback and record startup concurrently. Signed-off-by: Lionel Xu <Lionel.Xu@freescale.com>
2012-01-09SAUCE: set correct rates before registering SPDIF codec DAIEric Miao
BugLink: http://bugs.launchpad.net/bugs/855281 Playback/capture rates should be configured before the SPDIF codec DAI is registered, according to the parameters that passed in by the platform data. And this caused pulseaudio not working with the SPDIF sound card. Signed-off-by: Eric Miao <eric.miao@linaro.org>
2012-01-09ENGR00158176 SGTL5000: I/O Error appeared when recording mono wav through HWLionel Xu
When recording mono wav, SSI's network mode should be closed, or it will influence the internal freq config, making recording fail. Signed-off-by: Lionel Xu <Lionel.Xu@freescale.com>
2012-01-09ENGR00156813 MX53 ALSA: Recording no soundLionel Xu
There is no sound in the recorded wav, to enable recording, the VAG should be powered up, and the mic bias resistor should be setup with proper value. Signed-off-by: Lionel Xu <Lionel.Xu@freescale.com>
2012-01-09ENGR00139255 MX6Q_BSP ESAI: Add esai recording supportLionel Xu
Add ESAI recording to mx6q platform. To differentiate mx6q and mx53 in codec machine layer code. Signed-off-by: Lionel Xu <R63889@freescale.com>
2012-01-09ENGR00139255-2 MX6Q_BSP ESAI: Add esai recording supportLionel Xu
Add ESAI recording to mx6q platform. Note: since there is pad conflict between esai record and fec, add a boot argument esai_record to deal with it. This argument is required to enable the record functionality. Signed-off-by: Lionel Xu <R63889@freescale.com>
2012-01-09ENGR00154650-2 ESAI: add mx53 playback/record supportGary Zhang
add driver codes for mx53 ard. close esai clk when not used. add delay when power on cs42888 to avoid noise Signed-off-by: Gary Zhang <b13634@freescale.com>
2012-01-09ENGR00154727 ESAI: Resolve i2c access problem when there is no audio cardLionel Xu
I2c device should not probe successfully when there is no such device on the bus. This will make i2c access failure later. Signed-off-by: Lionel Xu <R63889@freescale.com>
2012-01-09ENGR00153793-1 ESAI: Resolve esai codec i2c suspend/resume problemLionel Xu
1)Resolve esai codec i2c suspend/resume problem; 2)Remove imx pcm operating function which already defined in imx-ssi.c Signed-off-by: Lionel Xu <R63889@freescale.com>
2012-01-09ENGR00153740-6 imx-cs42888: remove asrc codeDong Aisheng
The asrc code in imx-cs42888 driver will cause building fail. Further more, the current asrc driver does not support p2p mode, so remove it first. Signed-off-by: Dong Aisheng <b29396@freescale.com>
2012-01-09ENGR00153651-2 ESAI: Add esai/cs42888 audio codec support on mx6q platformLionel Xu
Add ESAI and related audio codec cs42888 support on mx6q platform. Signed-off-by: Lionel Xu <R63889@freescale.com>
2012-01-09ENGR00139461-2 mxc alsa soc spdif driverAlan Tull
S/PDIF tx and rx using ASoC layer. Signed-off-by: Alan Tull <alan.tull@freescale.com>
2012-01-09ENGR00142436-2 SGTL5000: Audio not work in 2.6.38 kernelZeng Zhaoming
If platform not provide headphone status detect gpio, should not failed the audio subsystem. Fix it by complains with a warning. Signed-off-by: Zeng Zhaoming <b32542@freescale.com>
2012-01-09ENGR00137340-1 ASOC: add headphone detection support for imx-sgtl5000.cZhang Jiejing
add a asoc headphone detection, it's a generic way using by asoc area. Signed-off-by: Zhang Jiejing <jiejing.zhang@freescale.com>
2012-01-09ENGR00141651-2 Audio: audio not work when ipmux2 clock disabled in idleZeng Zhaoming
When SOC enter wait mode, ipmux2_clk will disable to save power, if using ssi1 for audio transfer, audio will not work correctly. To around this, using ssi2. Signed-off-by: Zeng Zhaoming <b32542@freescale.com>
2012-01-09ENGR00141647 AUDIO,SGTL5000: Fix codec error after resetZeng Zhaoming
sgtl5000 codec not work after board reset, this is caused by sgtl5000 using register address step is 2, and snd-soc-core can't handle this as we expect, so we have to fill the register cache by reading register out when initialization instead of providing a default value array. Signed-off-by: Zeng Zhaoming <b32542@freescale.com>
2012-01-09ENGR00141217-8 ASoC: SGTL5000: Add DAP support for sgtl5000 codecFrank Li
upgrade to 2.6.38 kernel Add Digital Audio Process(DAP) for sgtl5000 codec Audio: imx53,loco,audio: make Loco sgtl5000 codec work config audmux when ssi probed, and fix sdma watermark settings Signed-off-by: Zeng Zhaoming <b32542@freescale.com>
2011-12-21ASoC: core: Don't schedule deferred_resume_work twiceStephen Warren
commit 82e14e8bdd88b69018fe757192b01dd98582905e upstream. For cards that have two or more DAIs, snd_soc_resume's loop over all DAIs ends up calling schedule_work(deferred_resume_work) once per DAI. Since this is the same work item each time, the 2nd and subsequent calls return 0 (work item already queued), and trigger the dev_err message below stating that a work item may have been lost. Solve this by adjusting the loop to simply calculate whether to run the resume work immediately or defer it, and then call schedule work (or not) one time based on that. Note: This has not been tested in mainline, but only in chromeos-2.6.38; mainline doesn't support suspend/resume on Tegra, nor does the mainline Tegra ASoC driver contain multiple DAIs. It has been compile-checked in mainline. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-21ASoC: Provide a more complete DMA driver stubMark Brown
commit cefcc03ffc9527dde56807339edb1719c8dbae5f upstream. Allow userspace applications to do more parameter setting by providing a more complete stub DMA driver specifying a wildcard set of formats and channels and essentially random values for the DMA parameters. This is required for useful runtime operation of the dummy DMA driver until we are able to figure out how to power up links and do hw_params() from DAPM. Sending to stable as without this the dummy driver is not terribly useful. Reported-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com> Tested-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-21ALSA: hda/realtek - Fix Oops in alc_mux_select()Takashi Iwai
commit cce4aa378a049f4275416ee6302dd24f37b289df upstream. When no imux is available (e.g. a single capture source), alc_auto_init_input_src() may trigger an Oops due to the access to -1. Add a proper zero-check to avoid it. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-21ALSA: sis7019 - give slow codecs more time to resetDavid Dillow
commit fc084e0b930d546872ab23667052499f7daf0fed upstream. There are some AC97 codec and board combinations that have been observed to take a very long time to respond after the cold reset has completed. In one case, more than 350 ms was required. To allow users to have sound on those platforms, we'll wait up to 500ms for the codec to become ready. As a board may have multiple codecs, with some faster than others to reset, we add a module parameter to inform the driver which codecs should be present. Reported-by: KotCzarny <tjosko@yahoo.com> Signed-off-by: David Dillow <dave@thedillows.org> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-09ASoC: Ensure WM8731 register cache is synced when resuming from disabledMark Brown
commit ed3e80c4c991a52f9fce3421536a78e331ae0949 upstream. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-09ASoC: wm8753: Skip noop reconfiguration of DAI modeTimo Juhani Lindfors
commit 2391a0e06789a3f1718dee30b282562f7ed28c87 upstream. This patch makes it possible to set DAI mode to its currently applied value even if codec is active. This is necessary to allow aplay -t raw -r 44100 -f S16_LE -c 2 < /dev/urandom & alsactl store -f backup.state alsactl restore -f backup.state to work without returning errors. This patch is based on a patch sent by Klaus Kurzmann <mok@fluxnetz.de>. Signed-off-by: Timo Juhani Lindfors <timo.lindfors@iki.fi> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-09ASoC: fsl_ssi: properly initialize the sysfs attribute objectTimur Tabi
commit 0f768a7235d3dfb6f4833030a95a06419df089cb upstream. Commit 6992f533 ("sysfs: Use one lockdep class per sysfs attribute") requires 'struct attribute' objects to be initialized with sysfs_attr_init(). Signed-off-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-12-09ALSA: lx6464es - fix device communication via command busTim Blechmann
commit a29878553a9a7b4c06f93c7e383527cf014d4ceb upstream. commit 6175ddf06b6172046a329e3abfd9c901a43efd2e optimized the mem*io functions that have been used to send commands to the device. these optimizations somehow corrupted the communication with the lx6464es, that resulted the device to be unusable with kernels after 2.6.33. this patch emulates the memcpy_*_io functions via a loop to avoid these problems. Signed-off-by: Tim Blechmann <tim@klingt.org> LKML-Reference: <4ECB5257.4040600@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: usb-audio - Fix the missing volume quirks at delayed initTakashi Iwai
commit dcaaf9f2c16b56f8bb316881fcd3f15c18fc71e7 upstream. In the recent usb-audio driver, the initialization of volume ranges may be delayed when the device doesn't respond well at the probing time. But the volume quirks for certain devices are applied only in mixer_ctl_feature_info() thus only at the very first probe and will be missing when the volume range is initialized later. This patch moves the volume quirk code to be always called from the volume-range extraction (get_min_max()), so that the quirks are properly applied in the later init time. Reported-and-tested-by: Alexey Fisher <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: usb-audio - Check the dB-range validity in the later read, tooTakashi Iwai
commit 9fcd0ab130579d9742538340edda3225f2b49a3e upstream. When the initial check of dB-range failed due to the read error, try to check again at the later read, too. When an invalid dB range is found, remove TLV flags and notify the mixer info change. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ASoC: Don't use wm8994->control_data in wm8994_readable_register()Mark Brown
commit 8eeea521d9d0fa6afd62df8c6e6566ee946117fa upstream. The field is no longer initialised so this will crash if running on wm8958. Reported-by: Thomas Abraham <thomas.abraham@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-21ALSA: hda - Don't add elements of other codecs to vmaster slaveTakashi Iwai
commit aeb4b88ec0a948efce8e3a23a8f964d3560a7308 upstream. When a virtual mater control is created, the driver looks for slave elements from the assigned card instance. But this may include the elements of other codecs when multiple codecs are on the same HD-audio bus. This works at the first time, but it'll give Oops when it's once freed and re-created via reconfig sysfs. This patch changes the element-look-up strategy to limit only to the mixer elements of the same codec. Reported-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Ensure the WM8962 oscillator and PLLs start up disabledMark Brown
commit 2af8de8c39cf58e5a5e40a9d5d71332da98e6ba7 upstream. Since there is no current software control for these they would otherwise be left enabled, consuming power. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Ensure WM8962 PLL registers are resetMark Brown
commit 4f4488abc97c1c27ff029f887944e6a6da1f5733 upstream. The WM8962 has a separate software reset for the PLL registers. Ensure that these are reset also on startup. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: WM8904: Set `invert' bit for Capture SwitchHong Xu
commit 5a7c5f26df3c0122814dfa1c13ef6dfbdbffdb86 upstream. Set `invert' bit for Capture Switch. Otherwise analogue is muted when Capture Switch is ON. Signed-off-by: Hong Xu <hong.xu@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Leave input audio data bit length settings untouched in wm8711_set_dai_fmtAxel Lin
commit d558cfc30064a97c2c65dbd2b3a4f5a1dea7ec1b upstream. Current implementation in wm8711_set_dai_fmt always clear BIT[3:2] (the Input Audio Data Bit Length Select) of WM8711_IFACE(07h) register. Input Audio Data Bit Length Select bits are set by wm8711_hw_params, we should leave BIT[3:2] untouched in wm8711_set_dai_fmt. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8711: Fix wrong mask for setting input audio data bit length selectAxel Lin
commit 04c57163c8edfbc50e022737014069998ba4fc5f upstream. The Input Audio Data Bit Length Select is controlled by BIT[3:2] of WM8711_IFACE(07h) register. Current code incorrectly masks BIT[1:0] which is for Audio Data Format Select. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda/realtek - Skip invalid digital out pinsTakashi Iwai
commit 51e4152a969aa6d2306492ebf143932dcb535c9b upstream. Some BIOS report invalid pins as digital output pins. The driver checks the connection but it doesn't do it fully correctly, and it leaves some undefined value as the audio-out widget, which makes the driver spewing warnings. This patch fixes the issue. Reference: https://bugzilla.novell.com/show_bug.cgi?id=727348 Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Add support for 92HD65 / 92HD66 family of codecsCharles Chin
commit ad5d8755116b431f0709c745ee17cb567a478d43 upstream. These codecs have SPDIF-in, which is new to the 92HD83xxx compatible families, so a bit of logic is added to support them. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Disable power-widget control for IDT 92HD83/93 as defaultCharles Chin
commit 35c11777b906042eca9e6f1c03e464726c7faa07 upstream. The power-widget control in patch_stac92hd83xxx() never worked properly, thus it's safer to turn it off as default for now. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: ua101: fix crash when unpluggingClemens Ladisch
commit 862a6244eb9f9f5123fe819454fcfcae0ee1f2f9 upstream. If the device is unplugged while running, it is possible for a PCM device to be closed after the disconnect callback has returned. This means that kill_stream_urb() and disable_iso_interface() would try to access already-invalid or freed USB data structures. The function free_usb_related_resources() was intended to prevent this, but forgot to clear the affected variables. Reported-and-tested-by: Olivier Courtay <olivier@courtay.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Fix ADC input-amp handling for Cx20549 codecTakashi Iwai
commit 6b45214277bec2193ad3ccb8d7aa6100b5a0f1a9 upstream. It seems that Conexant CX20549 chip handle only a single input-amp even though the audio-input widget has multiple sources. This has been never clear, and I implemented in the current way based on the debug information I got at the early time -- the device reacts individual input-amp values for different sources. This is true for another Conexant codec, but it's not applied to CX20549 actually. This patch changes the auto-parser code to handle a single input-amp per audio-in widget for CX20549. After applying this, you'll see only a single "Capture" volume control instead of separate "Mic" or "Line" captures when the device is set up to use a single ADC. We haven't tested 20551 and 20561 codecs yet. If these show the similar behavior like 20549, they need to set spec->single_adc_amp=1, too. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Fix a bug in WM8962 DSP_A and DSP_B settingsSusan Gao
commit fbc7c62a3ff831aef24894b7982cd1adb2b7e070 upstream. Signed-off-by: Susan Gao <sgao@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmico.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: Remove direct register cache accesses from WM8962 driverMark Brown
commit 38f3f31a0a797bdbcc0cdb12553bbecc2f9a91c4 upstream. Also fix return values for speaker switch updates. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 CaptureAxel Lin
commit 35024f4922f7b271e7529673413889aa3d51c5fc upstream. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: ak4535: fixup cache register tableAxel Lin
commit 7c04241acbdaf97f1448dcccd27ea0fcd1a57684 upstream. ak4535_reg should be 8bit, but cache table is defined as 16bit. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: ak4642: fixup cache register tableKuninori Morimoto
commit 19b115e523208a926813751aac8934cf3fc6085e upstream. ak4642 register was 8bit, but cache table was defined as 16bit. ak4642 doesn't work correctry without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8741: Fix setting interface format for DSP modesAxel Lin
commit 3a340104fad6ecbea5ad6792a2ea855f0507a6e0 upstream. According to the datasheet: Format Control (05h) BITS[3:2] FMT[1:0] Audio data format selection 00 = right justified mode 01 = left justified mode 10 = I2S mode 11 = DSP mode BIT[4] LRP Polarity selec for LRCLK/DSP mode select 0 = normal LRCLK poalrity/DSP mode A 1 = inverted LRCLK poarity/DSP mode B For SND_SOC_DAIFMT_DSP_A, we should set 0x000C instead of 0x0003. For SND_SOC_DAIFMT_DSP_B, we should set 0x001C instead of 0x0013. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ASoC: wm8940: Properly set codec->dapm.bias_levelAxel Lin
commit 5927f94700e860ae27ff24e7f3bc9e4f7b9922eb upstream. Reported-by: Chris Paulson-Ellis <chris@edesix.com> Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: HDA: Add new revision for ALC662David Henningsson
commit cc667a72d471e79fd8e5e291ea115923cf44dca0 upstream. The revision 0x100300 was found for ALC662. It seems to work well with patch_alc662. BugLink: http://bugs.launchpad.net/bugs/877373 Tested-by: Shengyao Xue <Shengyao.xue@canonical.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Acked-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-11-11ALSA: hda - Remove bad code for IDT 92HD83 family patchCharles Chin
commit 6c5c04e509b7000617b09d4301f0b9b6d171d1e6 upstream. The purpose of this patch is to remove a section of "bad" code that assigns the last DAC to ports E or F in order to support notebooks with docking in earlier days, around ALSA 1.0.19 - 21. This is not necessary now and actually breaks some configurations that use these ports as other devices. This have been tested on several different configurations to make sure that it is working for different combinations. Signed-off-by: Charles Chin <Charles.Chin@idt.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
2011-10-25ALSA: hda - Add position_fix quirk for Dell Inspiron 1010Takashi Iwai
commit 051a8cb6550d917225ead1cd008b5966350f6d53 upstream. The previous fix for the position-buffer check gives yet another regression on a Dell laptop. The safest fix right now is to add a static quirk for this device (and better to apply it for stable kernels too). Reported-by: Éric Piel <Eric.Piel@tremplin-utc.net> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
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