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2013-04-11ALSA: hda - Handle Headphone Mic jack more genericDavid Henningsson
Now that we have a flag for headphone mics, we can use that flag in the jack creation instead of creating the jack manually. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-11ALSA: hda - add some small convenience functions to auto parserDavid Henningsson
I never liked that we move our speaker and hp pins to line out if there are not any line outs; but now that we do, add some convenience functions to find hp and speaker pins even if they have been moved. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-11ALSA: hda - allow "Headphone Mic" parser flagDavid Henningsson
This allows a specific mic to get the "Headphone Mic" name, in addition to the existing "Headset Mic" name. Also, it allows for a special mark: if the sequence number is set to 0xc, that's an indication to prefer it for headset mic, and if it's set to 0xd, that's an indication to prefer it for headphone mic. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-11ALSA: at73c213: Use dev_pm_opsLars-Peter Clausen
Use dev_pm_ops instead of the deprecated legacy suspend/resume callbacks. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-10ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locationsDaniel Mack
It turns out the devices from Playback Design need the delay quirk after usb_set_interface from clocks.c as well. Make it a proper quirks function and factor out the code to quirks.c. Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-09ASoC: wm5102: Correct lookup of arizona struct in SYSCLK eventMark Brown
Reported-by: Ryo Tsutsui <Ryo.Tsutsui@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-04-09ASoC: wm8903: Fix the bypass to HP/LINEOUT when no DAC or ADC is runningAlban Bedel
The Charge Pump needs the DSP clock to work properly, without it the bypass to HP/LINEOUT is not working properly. This requirement is not mentioned in the datasheet but has been confirmed by Mark Brown from Wolfson. Signed-off-by: Alban Bedel <alban.bedel@avionic-design.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
2013-04-09ASoC: wm8994: Remove duplicate revision cacheMark Brown
There's already a device revision stored in the core data structure, don't duplicate it in the CODEC driver. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-09ALSA: hda - Apply codec delay to wallclock.Dylan Reid
For playback add the codec-side delay to the timestamp, for capture subtract it. This brings the timestamps in line with the time that was recently added to the delay reporting. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-08ASoC: Ux500: remove test for undefined Kconfig macroPaul Bolle
A test for CONFIG_SND_SOC_UX500_AB5500 was added in v3.5. But there never was a corresponding Kconfig symbol so this test has always evaluated to true. And since AB5500 support was removed in v3.5 it appears safe to remove this test and a few lines of code. Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-07ASoC: ux500_pcm: Use the same snd_pcm_hardware for playback and captureLars-Peter Clausen
The snd_pcm_hardware structs for playback and capture in the ux500 PCM are identical, so remove one of them and use the same snd_pcm_hardware struct for both playback and capture. Also move the defines used to initialize the snd_pcm_hardware fields from ux500_pcm.h to ux500_pcm.c since that's the only place where they are used. Also drop the assignment of the snd_pcm_hardware struct to runtime->hw since that is what the call to snd_soc_set_runtime_hwparams() right above it already does, so the second assignment is redundant. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-07ALSA: hda/realtek - Add a quirk for AC700 Chromebook.Dylan Reid
Correct pin configs for the Acer AC700. Most importantly indicate that SPDIF is connected, it routes to HDMI out. Similar to Aspire models, chain in the DMIC fixup and allow it to be applied to this codec (ALC269VB) as well. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-05ASoC: davinci-mcasp: don't overwrite DIT settingsYegor Yefremov
Channel size settings will be made at the end of davinci_mcasp_hw_params() routine and thus overwrite frame format settings made for DIT mode. This patch fixes this issue by taking op_mode into account. Tested with official PSP 3.2 kernel and sii9022a HDMI transmitter. Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com> Tested-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-05ASoC: davinci-mcasp: don't configure AFSX direction in DIT modeYegor Yefremov
AFSX won't be used in DIT mode. The related pins are AHCLKX and the data pins. Signed-off-by: Yegor Yefremov <yegorslists@googlemail.com> Acked-by: Vaibhav Bedia <vaibhav.bedia@ti.com> Tested-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-05ASoC: ep93xx_pcm: Fix compile errorLars-Peter Clausen
Commit 453807f3 ("ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params") introduced a small compile error by not updating the name of the 'dma_port' field to 'port'. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-05ASoC: fsl: Use common DAI DMA data structLars-Peter Clausen
Use the common DAI DMA data struct for fsl/imx, this allows us to use the common helper function to configure the DMA slave config based on the DAI DMA data. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Shawn Guo <shawn.guo@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-05ALSA: hda/cirrus - Add a quirk for Stumpy ChromeBox.Dylan Reid
The Stumpy ChromeBox needs its pin configs fixed up. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-05ALSA: hda/ca0132 - Update latency based on DSP state.Dylan Reid
The DSP in the CA0132 codec adds a variable latency to audio depending on what processing is being done. Add a new patch op to return that latency for capture and playback streams. The latency is determined by which blocks are enabled and knowing how much latency is added by each block. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-05ALSA: hda - Introduce get_delay codec PCM opsTakashi Iwai
Add a new codec PCM ops, get_delay(), to obtain the codec/stream- specific PCM delay count. When it's NULL, nothing changes. This new feature was requested for CA0132, which has significant delays in the path depending on the running DSP code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-05ALSA: hda/generic - fix uninitialized variableJiri Slaby
changed is not initialized in path_power_down_sync, but it is expected to be false in case no change happened in the loop. So set it to false. Signed-off-by: Jiri Slaby <jslaby@suse.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04Revert "ALSA: hda - Allow power_save_controller option override DCAPS"Takashi Iwai
This reverts commit 6ab317419c62850a71e2adfd1573e5ee87d8774f. The commit [6ab317419c: ALSA: hda - Allow power_save_controller option override DCAPS] changed the behavior of power_save_controller so that it can override the driver capability. This assumed that this option is rarely changed dynamically unlike power_save option. Too naive. It turned out that the user-space power-management tool tries to set power_save_controller option to 1 together with power_save option without knowing what's actually doing. This enabled forcibly the runtime PM of the controller, which is known to be broken om many chips thus disabled as default. So, the only sane fix is to revert this commit again. It was intended to ease debugging/testing for runtime PM enablement, but obviously we need another way for it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=56171 Reported-and-tested-by: Nikita Tsukanov <keks9n@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: hda - fix typo in proc outputDavid Henningsson
Rename "Digitial In" to "Digital In". This function is only used for proc output, so should not cause any problems to change. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: hda - Enabling Realtek ALC 671 codecRainer Koenig
* Added the device ID to the modalias list and assinged ALC662 patches for it * Added 4 port support for the device ID 0671 in alc662_parse_auto_config Signed-off-by: Rainer Koenig <Rainer.Koenig@ts.fujitsu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: support read-only freq controlEldad Zack
Some clocks might be read-only, e.g., external clocks (see also UAC2 4.7.2.1). In this case, setting the sample frequency will always fail (even if the rate is equal to the current clock rate), therefore do not write, but read the value and compare to the requested rate. If the clock is read only, avoid reading it twice. If it doesn't match, return -ENXIO since the clock is invalid for this configuration. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: show err in set_sample_rate_v2 debugEldad Zack
Show the error code returned from the USB subsystem in the debug messages. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: auto clock selection module paramEldad Zack
Add a module param to disable auto clock selection. This is provided for users that expect the audio stream to fail when the clock source is invalid (e.g., the word clock was unintentionally disconnected). Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: try to find and switch to valid clockEldad Zack
If a selector is available on a device, it may be pointing to a clock source which is currently invalid. If there is a valid clock source which can be selected, switch to it. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: UAC2: do clock validity check earlierEldad Zack
Move the check that parse_audio_format_rates_v2() do after receiving the clock source entity ID directly into the find function and add a validation flag to the function. This patch does not introduce any logic flow change. It is provided to allow introducing automatic clock switching easier later. By moving this uac_clock_source_is_valid callsite, 2 additional callsites can be avoided. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: use endianness macrosEldad Zack
Replace the endianness conversions with the kernel-wide swabbing macros in get/set_sample_rate_v2. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: spelling correctionEldad Zack
Correct spelling of snd_usb_endpoint_implict_feedback_sink in all occurances. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: neaten EXPORT_SYMBOLS placementEldad Zack
Put EXPORT_SYMBOLS directly under the exported function. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: neaten MODULE_DEVICE_TABLE placementEldad Zack
Minor style fix, following a general code style in the kernel. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04ALSA: usb-audio: convert list_for_each to entry variantEldad Zack
Change occurances of list_for_each into list_for_each_entry where applicable. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03ASoC: core: Merge Samsung fixes to avoid trivial context conflictMark Brown
Merge branch 'fix/samsung' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-component to resolve trivial conflict Conflicts: sound/soc/samsung/i2s.c
2013-04-03Merge branch 'fix/samsung' of ↵Mark Brown
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-samsung
2013-04-03ASoC: tegra: Use common DAI DMA data structLars-Peter Clausen
Use the common DAI DMA data struct for tegra, this allows us to use the common helper function to configure the DMA slave config based on the DAI DMA data. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Stephen Warren <swarren@nvidia.com> Tested-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ASoC: omap: Use common DAI DMA dataLars-Peter Clausen
Use the common DAI DMA data struct for omap, this allows us to use the common helper function to configure the DMA slave config based on the DAI DMA data. For omap-dmic and omap-mcpdm also move the DMA data from a global variable to the driver state struct. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03Merge branch 'topic/omap' of ↵Mark Brown
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into HEAD
2013-04-03ASoC: dmaengine-pcm: Add a common DAI DMA data structLars-Peter Clausen
This patch adds a common DMA data struct which can be used by DAI drivers to communicate their DMA configuration requirements to the DMA pcm driver. Having a common data structure for this allows us to implement common functions on top of them, which can be used by multiple platforms. This patch also introduces a new function to initialize certain fields of a dma_slave_config struct from the common DAI DMA data struct. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Stephen Warren <swarren@nvidia.com> Tested-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ASoC: dmaengine_pcm: Setup device_fc in snd_hwparams_to_dma_slave_configLars-Peter Clausen
Usually device_fc should be set to false for audio DMAs. Initialize it in a common place so drivers don't have to do this manually. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ASoC: tegra: Don't claim to support PCM pause and resumeLars-Peter Clausen
The tegra dmaengine driver does not support pausing and resuming a DMA stream. The tegra PCM driver still claims to support pause and resume though and implements them by stopping and restarting the stream. This is not what an application using pause/resume would expect. Usually applications have support for working around PCMs which do not support suspend and resume, so don't set the SNDRV_PCM_INFO_PAUSE and SNDRV_PCM_INFO_RESUME flags for the tegra PCM and use the default snd_dmaengine_pcm_trigger callback. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Stephen Warren <swarren@nvidia.com> Tested-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ALSA: usb-audio: Clean up the code in set_sample_rate_v2()Takashi Iwai
Just for cleaning up, introduce a new function get_sample_rate_v2() for replacing two identical calls in set_sample_rate_v2(). No functional change. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-03ASoC: Samsung: set drvdata before adding secondary devicePrathyush K
Currently, a new platform device is created for secondary device by calling platform_device_register_resndata and then the drvdata is set for this device. The following patch has been added to driver core: "driver core: fix possible missing of device probe". This results in the added device getting probed immediately but the drvdata for the secondary device is not yet set. This patch removes the platform_device_register_resndata call and instead calls platform_device_alloc, platform_set_drvdata and platform_device_add which fixes the above issue. Signed-off-by: Prathyush K <prathyush.k@samsung.com> Signed-off-by: Padmavathi Venna <padma.v@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03ASoC: Samsung: return error if drvdata is not setPrathyush K
This patch fixes a possible crash in case drvdata for the secondary device is not set. Signed-off-by: Prathyush K <prathyush.k@samsung.com> Signed-off-by: Padmavathi Venna <padma.v@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-04-03Merge branch 'for-linus' into for-nextTakashi Iwai
Back-merge for cleaning up usb-audio code the recent commit modified, and further UAC2 autoclock patches.
2013-04-03ALSA: usb: Work around CM6631 sample rate change bugTorstein Hegge
The C-Media CM6631 USB receiver doesn't respond to changes in sample rate while the interface is active. The same behavior is observed in other UAC2 hardware like the VIA VT1731. Reset the interface after setting the sampling frequency on sample rate changes, to ensure that the sample rate set by snd_usb_init_sample_rate() is used. Otherwise, the device will try to use the sample rate of the previous stream, causing distorted sound on sample rate changes. The reset is performed for all UAC2 devices, as it should not affect a standards compliant device, but it is only necessary for C-Media CM6631, VIA VT1731 and possibly others. Failure to read sample rate from the device is not handled as an error in set_sample_rate_v2(), as (permanent or intermittent) failure to read sample rate isn't essential for a successful sample rate set. Signed-off-by: Torstein Hegge <hegge@resisty.net> Acked-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda - bug fix on HDMI ELD debug messageMengdong Lin
This patch let ELD debug message show 'pin_eld->monitor_present' which reflects the real pin response to verb GET_PIN_SENSE. 'eld->monitor_present' should not be used here because 'eld' is a temp structure now and so its "monitor_present" is not set. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda - bug fix on return value when getting HDMI ELD infoMengdong Lin
In function snd_hdmi_get_eld(), the variable 'ret' should be initialized to 0. Otherwise it will be returned uninitialized as non-zero after ELD info is got successfully. Thus hdmi_present_sense() will always assume ELD info is invalid by mistake, and /proc file system cannot show the proper ELD info. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Cc: stable@vger.kernel.org Acked-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02ALSA: hda/ca0132 - Delay HP amp turnon.Chih-Chung Chang
Turing on the headphone amp interferes with the impedance measurement used to detect a TRRS style headset microphone. Delay the HP turn on until 500ms after the jack is detected, allowing the mic detection state machine to run to completion. Signed-off-by: Chih-Chung Chang <chihchung@chromium.org> Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-02sound: oss: sb_common: Used kmemdup instead of kmalloc and memcpyAlexandru Gheorghiu
Used kmemdup instead of replicating it's behaviour with kmalloc followed by memcpy. Patch found using coccinelle. Signed-off-by: Alexandru Gheorghiu <gheorghiuandru@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>