aboutsummaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2014-09-02ALSA: hda - Fix COEF setups for ALC1150 codecTakashi Iwai
ALC1150 codec seems to need the COEF- and PLL-setups just like its compatible ALC882 codec. Some machines (e.g. SunMicro X10SAT) show the problem like too low output volumes unless the COEF setup is applied. Reported-and-tested-by: Dana Goyette <danagoyette@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-09-01ALSA: hda - Fix digital mic on Acer Aspire 3830TGTakashi Iwai
Acer Aspire 3830TG with CX20588 codec has a digital built-in mic that has the same problem like many others, the inverted signal in stereo. Apply the same fixup to this machine, too. Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-29ALSA: firewire-lib/dice: add arrangements of PCM pointer and interrupts for ↵Takashi Sakamoto
Dice quirk In IEC 61883-6, one data block transfers one event. In ALSA, the event equals one PCM frame, hence one data block transfers one PCM frame. But Dice has a quirk at higher sampling rate (176.4/192.0 kHz) that one data block transfers two PCM frames. Commit 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") moved some codes related to this quirk into Dice driver. But the commit forgot to add arrangements for PCM period interrupts and DMA pointer updates. As a result, Dice driver cannot work correctly at higher sampling rate. This commit adds 'double_pcm_frames' parameter to amdtp structure for this quirk. When this parameter is set, PCM period interrupts and DMA pointer updates occur at double speed than in IEC 61883-6. Reported-by: Daniel Robbins <drobbins@funtoo.org> Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.16 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-29ALSA: dice: fix wrong channel mappping at higher sampling rateTakashi Sakamoto
The channel mapping is initialized by amdtp_stream_set_parameters(), however Dice driver set it before calling this function. Furthermore, the setting is wrong because the index is the value of array, and vice versa. This commit moves codes for channel mapping after the function and set it correctly. Reported-by: Daniel Robbins <drobbins@funtoo.org> Fixes: 10550bea44a8 ("ALSA: dice/firewire-lib: Keep dualwire mode but obsolete CIP_HI_DUALWIRE") Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Cc: <stable@vger.kernel.org> # 3.16 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-27ALSA: hda - Set up initial pins for Acer Aspire V5Takashi Iwai
Acer Aspire V5 doesn't set up the pins correctly at the cold boot while the pins are corrected after the warm reboot. This patch gives the proper pin configs statically in the driver as a workaround. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=81561 Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22ALSA: pcm: Fix the silence data for DSD formatsTakashi Iwai
Right now we set 0 as the silence data for DSD_U8 and DSD_U16 formats, but this is actually wrong. 0 is rather the most negative value. Alternatively, we may take the repeating 0x69 pattern like ffmpeg deploys. Reference: https://ffmpeg.org/pipermail/ffmpeg-cvslog/2014-April/076427.html Suggested-by: Alexander E. Patrakov <patrakov@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22ALSA: ctxfi: ct20k1reg: Fix typo in include guardRasmus Villemoes
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22ALSA: hda: ca0132_regs.h: Fix typo in include guardRasmus Villemoes
Signed-off-by: Rasmus Villemoes <linux@rasmusvillemoes.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-22ALSA: core: fix buffer overflow in snd_info_get_line()Clemens Ladisch
snd_info_get_line() documents that its last parameter must be one less than the buffer size, but this API design guarantees that (literally) every caller gets it wrong. Just change this parameter to have its obvious meaning. Reported-by: Tommi Rantala <tt.rantala@gmail.com> Cc: <stable@vger.kernel.org> # v2.2.26+ Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-19ALSA: hda/hdmi - apply Valleyview fix-ups to Cherryview display codecLibin Yang
Valleyview and Cherryview have the same behavior on display audio. So this patch defines is_valleyview_plus() to include codecs for both Valleyview and its successor Cherryview, and apply Valleyview fix-ups to Cherryview. Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-19ALSA: hda/hdmi - set depop_delay for haswell plusLibin Yang
Both Haswell and Broadwell need set depop_delay to 0. So apply this setting to haswell plus. Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-19ALSA: hda - restore the gpio led after resumeHui Wang
On some HP laptops, the mute led is controlled by codec gpio. When some machine resume from s3/s4, the codec gpio data will be cleared to 0 by BIOS: Before suspend: IO[3]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 After resume: IO[3]: enable=1, dir=1, wake=0, sticky=0, data=0, unsol=0 To skip the AFG node to enter D3 can't fix this problem. A workaround is to restore the gpio data when the system resume back from s3/s4. It is safe even on the machines without this problem. BugLink: https://bugs.launchpad.net/bugs/1358116 Tested-by: Franz Hsieh <franz.hsieh@canonical.com> Cc: stable@vger.kernel.org Signed-off-by: Hui Wang <hui.wang@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-16ALSA: hda/realtek - Avoid setting wrong COEF on ALC269 & coTakashi Iwai
ALC269 & co have many vendor-specific setups with COEF verbs. However, some verbs seem specific to some codec versions and they result in the codec stalling. Typically, such a case can be avoided by checking the return value from reading a COEF. If the return value is -1, it implies that the COEF is invalid, thus it shouldn't be written. This patch adds the invalid COEF checks in appropriate places accessing ALC269 and its variants. The patch actually fixes the resume problem on Acer AO725 laptop. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181 Tested-by: Francesco Muzio <muziofg@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-16Merge tag 'asoc-v3.17-rc1' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Fixes for v3.17 Nothing too exciting here, a bunch of driver fixes that came along since the initial pull request but none that really stand our and a warning fix in the core.
2014-08-15Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/fsl', ↵Mark Brown
'asoc/fix/fsl-esai', 'asoc/fix/intel', 'asoc/fix/mcasp' and 'asoc/fix/pxa' into asoc-linus
2014-08-15Merge remote-tracking branch 'asoc/fix/pcm512x' into asoc-linusMark Brown
2014-08-15Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linusMark Brown
2014-08-15Merge tag 'asoc-v3.17' into asoc-linusMark Brown
ASoC: Updates for v3.17 This has been a pretty exciting release in terms of the framework, we've finally got support for multiple CODECs attached to a single DAI link which has been something there's been interest in as long as I've been working on ASoC. A big thanks to Benoit and Misael for their work on this. Otherwise it's been a fairly standard release for development, including more componentisation work from Lars-Peter and a good selection of both CODEC and CPU drivers. - Support for multiple CODECs attached to a single DAI, enabling systems with for example multiple DAC/speaker drivers on a single link, contributed by Benoit Cousson based on work from Misael Lopez Cruz. - Support for byte controls larger than 256 bytes based on the use of TLVs contributed by Omair Mohammed Abdullah. - More componentisation work from Lars-Peter Clausen. - The remainder of the conversions of CODEC drivers to params_width() - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments TAS2552. - Lots of updates and fixes, especially to the DaVinci, Intel, Freescale, Realtek, and rcar drivers. # gpg: Signature made Mon 04 Aug 2014 17:13:21 BST using RSA key ID 7EA229BD # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
2014-08-15Merge tag 'asoc-v3.16-rc5' into asoc-linusMark Brown
ASoC: Fixes for v3.16 A bigger batch of changes than I would like as I didn't send any for a few weeks without noticing how many had built up. They are almost all driver specific though, larger changes are: - Fixes to the newly added Baytrail/MAX98090 which look like some QA was missed on the microphone detection. - Deletion of some erroniously listed audio formats for Haswell. - Fix debugfs creation in the core so that we don't try to generate multiple directories with the same name, relatively large textually but simple to inspect by eye and test. - A couple of bugfixes for the rcar driver one of which which involves a bit of code motion to move initailisation of some hardware out of common paths into device specific ones. - Ensure both channels are powered up for mono outputs on Arizona devices, involving some simple data tables listing the outputs and a loop over them. - A couple of fixes to save and restore information on suspended and idle Samsung I2S controllers. # gpg: Signature made Tue 22 Jul 2014 00:52:53 BST using RSA key ID 7EA229BD # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
2014-08-15Merge tag 'asoc-v3.16-rc1' into asoc-linusMark Brown
ASoC: Fixes for v3.16 Quite a few build coverage fixes in here among the usual small driver fixes includling the sigmadsp change from Lars - moving the driver to separate modules per bus (which is basically just code motion) avoids issues with some combinations of buses being enabled. # gpg: Signature made Thu 19 Jun 2014 11:57:31 BST using RSA key ID 7EA229BD # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
2014-08-14ALSA: hda - Set TLV_DB_SCALE_MUTE bit for cx5051 vmasterTakashi Iwai
Conexnat HD-audio driver has a workaround for cx5051 (aka CX20561) chip to add fake mute controls to each amp (commit 3868137e). This implies the minimum-as-mute TLV bit in TLV for each corresponding control. Meanwhile we build the virtual master from these, but the TLV bit is missing, even though the slaves have it. This patch simply adds the missing TLV_DB_SCALE_MUTE bit for vmaster, as already done in patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-13ASoC: pxa-ssp: drop SNDRV_PCM_FMTBIT_S24_LEDaniel Mack
This mode is unsupported, as the DMA controller can't do zero-padding of samples. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-by: Johannes Stezenbach <js@sig21.net> Signed-off-by: Mark Brown <broonie@linaro.org> Cc: stable@vger.kernel.org
2014-08-13ASoC: fsl-esai: Revert .xlate_tdm_slot_mask() supportShengjiu Wang
This reverts commit a603c8ee526f5ea9ad9b40710308766299ad8a69. fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask(). fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled bit to 0. For esai when the bit value is 1, the slot is enabled, when the bit value is 0, the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will work abnormally. So revert this patch, make the esai use default function. Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com> Acked-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-12ASoC: mcasp: Fix implicit BLCK divider settingJyri Sarha
The implicit BLCK divider setting was broken by "ASoC: mcasp: don't override bclk divider if it was provided by the machine"-patch. After the BCLK divider is implicitly set for the first time the mcasp->bclk_div gets a non zero value and the implicit setting is "turned off". Signed-off-by: Jyri Sarha <jsarha@ti.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-12ASoC: arizona: Fix TDM slot length handling in arizona_hw_paramsNikesh Oswal
TDM slot length was set same as word length, regardless of the value received in set_tdm_slot. This patch sets the TDM slot length correctly as received in set_tdm_slot DAI callback Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-12ASoC: pcm512x: Correct Digital Playback control namesMark Brown
The source type should come before the direction specifier according to ControlNames.txt. Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-11ASoC: dapm: Fix uninitialized variable in snd_soc_dapm_get_enum_double()Geert Uytterhoeven
If soc_dapm_read() fails, reg_val will be uninitialized, and bogus values will be written later: sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double': sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized] unsigned int reg_val, val; ^ Return early on error to fix this. Introduced by commit ce0fc93ae56e2ba50ff8c220d69e4e860e889320 ("ASoC: Add DAPM support at the component level"). Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be> Signed-off-by: Mark Brown <broonie@linaro.org> Acked-by: Lars-Peter Clausen <lars@metafoo.de>
2014-08-11ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in resetJarkko Nikula
There is no need to restore and restart PCM streams in case ADSP didn't reach reset and power off state during system suspend/resume cycle. In that case stream is still active but paused and firmware doesn't allow allocating a new stream before paused stream is freed. ADSP remains active in case suspend sequence didn't go to suspend_late stage. This can happen when either suspend sequence is aborted by a wakeup or by letting only devices suspend by "echo devices >/sys/power/pm_test". Currently stream restoring fails in these suspend cases. Fix this by adding a flag that indicates is complete stream reinitialization needed or is it enough to resume paused stream. Flag is set when we know that ADSP reached suspend_late. Initial fix to this issue came from Fang Yang. I modified it a little and forward ported it to top of two other suspend/resume patches from me. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Tested-by: Borun Fu <borun.fu@intel.com> Cc: yang fang <yang.a.fang@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-11ASoC: Intel: Wait Baytrail ADSP boot at resume_early stageJarkko Nikula
Remove sst_byt_pcm_dev_resume() and move waiting of firmware boot into sst_byt_pcm_dev_resume_early(). Now suspend_late and resume_early phases are in sync with each other so that we know that ADSP was put into reset and was unpowered after suspend_late and is ready to resume IO after resume_early during resume stage in sst_byt_pcm_trigger(). Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Tested-by: Borun Fu <borun.fu@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-11ASoC: Intel: Merge Baytrail ADSP suspend_noirq into suspend_lateJarkko Nikula
Merge DSP reset and cleanup sequence in sst_byt_pcm_dev_suspend_noirq() into sst_byt_pcm_dev_suspend_late(). First their order was wrong by first unloading firmware modules in suspend_late and then taking DSP into reset in suspend_noirq. Second ACPI has put device into OFF state already during suspend_late so trying to reset the DSP is a no-op at suspend_noirq stage. Fix these by moving DSP reset and cleanup into sst_byt_pcm_dev_suspend_late() before firmware unloading. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Tested-by: Borun Fu <borun.fu@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-10ALSA: hda/ca0132 - Don't try loading firmware at resume when already failedTakashi Iwai
CA0132 driver tries to reload the firmware at resume. Usually this works since the firmware loader core caches the firmware contents by itself. However, if the driver failed to load the firmwares (e.g. missing files), reloading the firmware at resume goes through the actual file loading code path, and triggers a kernel WARNING like: WARNING: CPU: 10 PID:11371 at drivers/base/firmware_class.c:1105 _request_firmware+0x9ab/0x9d0() For avoiding this situation, this patch makes CA0132 skipping the f/w loading at resume when it failed at probe time. Reported-and-tested-by: Janek Kozicki <cosurgi@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10ALSA: hda - Fix pop noises on reboot for Dell XPS 13 9333Gabriele Mazzotta
If nid 0x15 (Headphone Playback Switch) is in D3 and headphones are plugged in when the laptop reboots, a pop noise is generated. Prevent this by keeping nid 0x15 in D0 when headphones are plugged in. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611 Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10ALSA: hda - Set internal mic as default input source on Dell XPS 13 9333Gabriele Mazzotta
If the laptop is powered on with a jack plugged in, independently on what is plugged, the jack is treated as a microphone jack. Initialize the capture source so that by default jacks are treated as headphones jacks. This will also prevent pop noises on boot in case headphones are plugged in since setting/unsetting mic-in as input source causes a pop noise. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=76611 Signed-off-by: Gabriele Mazzotta <gabriele.mzt@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-10ALSA: usb-audio: fix BOSS ME-25 MIDI regressionClemens Ladisch
The BOSS ME-25 turns out not to have any useful descriptors in its MIDI interface, so its needs a quirk entry after all. Reported-and-tested-by: Kees van Veen <kees.vanveen@gmail.com> Fixes: 8e5ced83dd1c ("ALSA: usb-audio: remove superfluous Roland quirks") Cc: <stable@vger.kernel.org> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-08MAINTAINERS: Add i.MX maintainers and paths to Freescale ASoC entryMark Brown
There's several new i.MX specific controllers, try to help make sure they get reviewed by the people working on them. Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-07ALSA: hda - Fix parsing of CMI8888 codecTakashi Iwai
CMI8888 codec chip has a boost amp (only) on the headphone pin, and this confuses the generic parser, which tends to pick up the most outside amp. This results in the wrong volume setup, as the driver complains like: hda_codec: Mismatching dB step for vmaster slave (-100!=1000) For avoiding this problem, rule out the amp on NID 0x10 and create "Headphone Amp" volume control manually instead. Note that this patch still doesn't fix all problems yet. The sound output from the line out seems still too low. It will be fixed in another patch (hopefully). Reported-and-tested-by: Vincent Lejeune <vljn@ovi.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-07ALSA: hda - Fix probing and stuttering on CMI8888 HD-audio controllerTakashi Iwai
ASUS Phoebus with CMI8888 HD-audio chip (PCI id 13f6:5011) doesn't work with HD-audio driver as is because of some weird nature. For making DMA properly working, we need to disable MSI. The position report buffer doesn't work, thus we need to force reading LPIB instead. And yet, the codec CORB/RIRB communication gives errors unless we disable the snooping (caching). In this patch, all these workarounds are added as a quirk for the device. The HD-audio *codec* chip needs yet another workaround, but it'll be provided in the succeeding patch. Reported-and-tested-by: Vincent Lejeune <vljn@ovi.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-07ALSA: hda/realtek - Fixed ALC286/ALC288 recording delay for Headset MicKailang Yang
It will be recording voice delay for resume back recording for Headset Mic. This alc286 will quickly open Headset Mic, to prevent avoid recording files are missing. The issue was fixed. This is follow ALC286 programing guide. [fix build error, add static and renamed the function by tiwai] Signed-off-by: Kailang Yang <kailang@realtek.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-06ASoC: Intel: Update Baytrail ADSP firmware nameJarkko Nikula
Update the initial Baytrail ADSP firmware file name with the one that is now in linux-firmware.git. Please see linux-firmware.git commit 7551a3a78453 ("fw_sst_0f28: Add firmware for Intel Baytrail SST DSP"). Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-06sound: oss: Remove typedefs wanc_info and wavnc_port_infoHimangi Saraogi
The Linux kernel coding style guidelines suggest not using typedefs for structure types. This patch gets rid of the typedefs for wanc_info and wavnc_port_info. A simplified version of the Coccinelle semantic patch that finds the case is: @tn@ identifier i; type td; @@ -typedef struct i { ... } -td ; @@ type tn.td; identifier tn.i; @@ -td + struct i Signed-off-by: Himangi Saraogi <himangi774@gmail.com> Acked-by: Julia Lawall <julia.lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-06sound: oss: uart401: Remove typedef uart401_devcHimangi Saraogi
The Linux kernel coding style guidelines suggest not using typedefs for structure types. This patch gets rid of the typedef for uart401_devc. The following Coccinelle semantic patch detects the case. @tn@ identifier i; type td; @@ -typedef struct i { ... } -td ; @@ type tn.td; identifier tn.i; @@ -td + struct i Signed-off-by: Himangi Saraogi <himangi774@gmail.com> Acked-by: Julia Lawall <julia.lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05ALSA: usb-audio: Whitespace cleanups for sound/usb/midi.*Adam Goode
Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05ALSA: usb-audio: Respond to suspend and resume callbacks for MIDI inputAdam Goode
sound/usb/card.c registers USB suspend and resume but did not previously kill the input URBs. This means that USB MIDI devices left open across suspend/resume had non-functional input (output still usually worked, but it looks like that is another issue). Before this change, we would get ESHUTDOWN for each of the input URBs at suspend time, killing input. Signed-off-by: Adam Goode <agoode@google.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05sound/oss/pss: Remove typedefs pss_mixerdata and pss_confdataHimangi Saraogi
The Linux kernel coding style guidelines suggest not using typedefs for structure types. This patch gets rid of the typedefs for pss_mixerdata and pss_confdata. The following Coccinelle semantic patch is used to make the change. @tn@ identifier i; type td; @@ -typedef struct i { ... } -td ; @@ type tn.td; identifier tn.i; @@ -td + struct i Signed-off-by: Himangi Saraogi <himangi774@gmail.com> Acked-by: Julia Lawall <julia.lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05sound/oss/opl3: Remove typedef opl_devinfoHimangi Saraogi
This typedef is unnecessary and should just be removed as they are never used. The following Coccinelle semantic patch detects the case. @tn@ identifier i; type td; @@ -typedef struct i { ... } -td ; @@ type tn.td; identifier tn.i; @@ -td + struct i Signed-off-by: Himangi Saraogi <himangi774@gmail.com> Acked-by: Julia Lawall <julia.lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-05ALSA: fireworks: fix specifiers in format strings for propper outputTakashi Sakamoto
Use %d for loop counter and %X for device capabilities. This is a supplemental patch for Hans Wennborg's patch. Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-08-04Merge tag 'asoc-v3.17' of ↵Takashi Iwai
git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: Updates for v3.17 This has been a pretty exciting release in terms of the framework, we've finally got support for multiple CODECs attached to a single DAI link which has been something there's been interest in as long as I've been working on ASoC. A big thanks to Benoit and Misael for their work on this. Otherwise it's been a fairly standard release for development, including more componentisation work from Lars-Peter and a good selection of both CODEC and CPU drivers. - Support for multiple CODECs attached to a single DAI, enabling systems with for example multiple DAC/speaker drivers on a single link, contributed by Benoit Cousson based on work from Misael Lopez Cruz. - Support for byte controls larger than 256 bytes based on the use of TLVs contributed by Omair Mohammed Abdullah. - More componentisation work from Lars-Peter Clausen. - The remainder of the conversions of CODEC drivers to params_width() - Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments TAS2552. - Lots of updates and fixes, especially to the DaVinci, Intel, Freescale, Realtek, and rcar drivers.
2014-08-04Merge remote-tracking branch 'asoc/topic/wm8985' into asoc-nextMark Brown
2014-08-04Merge remote-tracking branches 'asoc/topic/tlv320aic3x', 'asoc/topic/width', ↵Mark Brown
'asoc/topic/wm0010', 'asoc/topic/wm8904' and 'asoc/topic/wm8962' into asoc-next
2014-08-04Merge remote-tracking branches 'asoc/topic/tlv', 'asoc/topic/tlv320aic23', ↵Mark Brown
'asoc/topic/tlv320aic31xx' and 'asoc/topic/tlv320aic32x4' into asoc-next