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* Merge tag 'sound-3.10' of ↵Linus Torvalds2013-05-1012-48/+51
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "This contains small fixes since the previous pull request: - A few regression fixes and small updates of HD-audio - Yet another fix for Haswell HDMI audio - A copule of trivial fixes in ASoC McASP, DPAM and WM8994" * tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: Revert "ALSA: hda - Don't set up active streams twice" ALSA: Add comment for control TLV API ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecs ALSA: HDA: Fix Oops caused by dereference NULL pointer ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata() ALSA: mips/hal2: Remove redundant platform_set_drvdata() ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecs sound: Fix make allmodconfig on MIPS ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllers ALSA: atmel: Remove redundant platform_set_drvdata() ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode. ASoC: wm8994: missing break in wm8994_aif3_hw_params() ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare
| * Revert "ALSA: hda - Don't set up active streams twice"Takashi Iwai2013-05-101-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | This reverts commit affdb62b815b38261f09f9d4ec210a35c7ffb1f3. The commit introduced a regression with AD codecs where the stream is always clean up. Since the patch is just a minor optimization and reverting the commit fixes the issue, let's just revert it. Reported-and-tested-by: Michael Burian <michael.burian@sbg.at> Cc: <stable@vger.kernel.org> [v3.9+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Apply pin-enablement workaround to all Haswell HDMI codecsTakashi Iwai2013-05-081-32/+22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is a revised patch based on Mengdong Lin's fix patch, which is a supplement to a previous patch [1611a9c9: ALSA: hda - Add fixup for Haswell to enable all pin and convertor widgets]. Some Haswell BIOS will disable the 2nd and 3rd pin/covertor widgets when the HD-A controller changes state from D3 to D0. So when the controller resumes after a system or runtime suspend, these widgets are disabled and programming these widgets to D0 will cause H/W error and codec will not respond. In addition, we found out that some BIOS disables the pins at S3 although it shows up at boot. This confuses the driver utterly, and the hardware falls into the fatal communication error like the above. So in this patch, we apply intel_haswell_enable_all_pins() not only as a fixup to a certain device (with 8086:2010) but to all Haswell machines. The codec driver basically assumes that all pins are exposed, so it's anyway better to see them from the beginning. Even if all pins and converters are shown by this call, there should be no regression in practice: the pin default configurations are still kept, thus the disabled pins are handled as disabled by the driver properly. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: HDA: Fix Oops caused by dereference NULL pointerWang YanQing2013-05-071-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The interrupt handler azx_interrupt will call azx_update_rirb, which may call snd_hda_queue_unsol_event, snd_hda_queue_unsol_event will dereference chip->bus pointer. The problem is we alloc chip->bus in azx_codec_create which will be called after we enable IRQ and enable unsolicited event in azx_probe. This will cause Oops due dereference NULL pointer. I meet it, good luck:) [Rearranged the NULL check before the tracepoint and added another NULL check of bus->workq -- tiwai] Signed-off-by: Wang YanQing <udknight@gmail.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: mips/sgio2audio: Remove redundant platform_set_drvdata()Sachin Kamat2013-05-061-1/+0
| | | | | | | | | | | | | | | | | | | | Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Cc: Vivien Chappelier <vivien.chappelier@linux-mips.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: mips/hal2: Remove redundant platform_set_drvdata()Sachin Kamat2013-05-061-1/+0
| | | | | | | | | | | | | | | | | | | | Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Cc: Thomas Bogendoerfer <tsbogend@alpha.fanken.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix 3.9 regression of EAPD init on Conexant codecsTakashi Iwai2013-05-041-3/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | The older Conexant codecs have up to two EAPDs and these are supposed to be rather statically turned on. The new generic parser code assumes the dynamic on/off per path usage, thus it resulted in the silent output on some machines. This patch fixes the problem by simply assuming the static EAPD on for such old Conexant codecs as we did until 3.8 kernel. Reported-and-tested-by: Christopher K. <c.krooss@gmail.com> Cc: <stable@vger.kernel.org> [v3.9] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * sound: Fix make allmodconfig on MIPSTakashi Iwai2013-05-031-0/+1
| | | | | | | | | | | | | | | | | | | | The compile of soundcard.c is broken on MIPS when allmodconfig is used because of the missing MAX_DMA_CHANNELS definition. As a simple workaround, just add a Kconfig dependency. Reported-by: Andrew Morton <akpm@linux-foundation.org> Cc: Ralf Baechle <ralf@linux-mips.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: hda - Fix system panic when DMA > 40 bits for Nvidia audio controllersMike Travis2013-05-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | The audio driver mistakenly allows 64 bit addresses to be created for the audio driver on Nvidia GPUs. Unfortunately, the hardware normally only supports up to 40 bits of DMA. This can cause system panics as well as misdirected data when the address is > 40 bits as the upper part the address is truncated. Signed-off-by: Mike Travis <travis@sgi.com> Reviewed-by: Mike Habeck <habeck@sgi.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: atmel: Remove redundant platform_set_drvdata()Sachin Kamat2013-05-032-4/+0
| | | | | | | | | | | | | | | | | | | | | | Commit 0998d06310 (device-core: Ensure drvdata = NULL when no driver is bound) removes the need to set driver data field to NULL. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no> Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge tag 'asoc-v3.10-4' of ↵Takashi Iwai2013-05-033-4/+8
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.10 A few more bug fixes, the DAPM clock fix is actually a driver specific one since currently there's only one user of the clock support due to the problems relying on the clock API.
| | * Merge remote-tracking branch 'asoc/fix/wm8994' into asoc-linusMark Brown2013-05-031-0/+1
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| | | * ASoC: wm8994: missing break in wm8994_aif3_hw_params()Dan Carpenter2013-04-301-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The missing break here means that we always return early and the function is a no-op. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * | Merge remote-tracking branch 'asoc/fix/davinci' into asoc-linusMark Brown2013-05-031-2/+5
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| | | * | ASoC: McASP: Fix receive clock polarity in DAIFMT_NB_NF mode.Marek Belisko2013-05-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | According documentation bit ACLKRPOL is set to 0 (receiver samples data on falling edge) and when set to 1 (receiver samples data on rising edge). I2S data are always sampled on falling edge and valid during rising edge of bit clock. So in case of capture data transmitter sample data on falling edge and macsp must read then on rising edge. Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | ASoC: McASP: Add pins output direction for rx clocks when configured in ↵Marek Belisko2013-04-301-1/+4
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | CBS_CFS format When McASP is bit clock and frame clock master enable pin output for rx clocks. Signed-off-by: Marek Belisko <marek.belisko@streamunlimited.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * / ASoC: dapm: use clk_prepare_enable and clk_disable_unprepareFabio Baltieri2013-04-301-2/+2
| | |/ | | | | | | | | | | | | | | | | | | | | | Update dapm_clock_event to use clk_prepare_enable and clk_disable_unprepare. Signed-off-by: Fabio Baltieri <fabio.baltieri@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | aio: don't include aio.h in sched.hKent Overstreet2013-05-071-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Faster kernel compiles by way of fewer unnecessary includes. [akpm@linux-foundation.org: fix fallout] [akpm@linux-foundation.org: fix build] Signed-off-by: Kent Overstreet <koverstreet@google.com> Cc: Zach Brown <zab@redhat.com> Cc: Felipe Balbi <balbi@ti.com> Cc: Greg Kroah-Hartman <gregkh@linuxfoundation.org> Cc: Mark Fasheh <mfasheh@suse.com> Cc: Joel Becker <jlbec@evilplan.org> Cc: Rusty Russell <rusty@rustcorp.com.au> Cc: Jens Axboe <axboe@kernel.dk> Cc: Asai Thambi S P <asamymuthupa@micron.com> Cc: Selvan Mani <smani@micron.com> Cc: Sam Bradshaw <sbradshaw@micron.com> Cc: Jeff Moyer <jmoyer@redhat.com> Cc: Al Viro <viro@zeniv.linux.org.uk> Cc: Benjamin LaHaise <bcrl@kvack.org> Reviewed-by: "Theodore Ts'o" <tytso@mit.edu> Signed-off-by: Andrew Morton <akpm@linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | | Merge tag 'sound-3.10' of ↵Linus Torvalds2013-05-03241-3622/+7059
|\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "Mostly many small changes spread as seen in diffstat in sound/* directory by this update. A significant change in the subsystem level is the introduction of snd_soc_component, which will help more generic handling of SoC and off-SoC components. Also, snd_BUG_ON() macro is enabled unconditionally now due to its misuses, so people might hit kernel warnings (it's a good thing for us). - compress-offload: support for capture by Charles Keepax - HD-audio: codec delay support by Dylan Reid - HD-audio: improvements/fixes in generic parser: better headphone mic and headset mic support, jack_modes hint consolidation, proper beep attach/detachment, generalized power filter controls by David Henningsson, et al - HD-audio: Improved management of HDMI codec pins/converters - HD-audio: Better pin/DAC assignment for VIA codecs - HD-audio: Haswell HDMI workarounds - HD-audio: ALC268 codec support, a few new quirks for Chromebooks - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency fix by Clemens Ladisch - USB: support for DSD formats by Daniel Mack - USB: A few UAC2 device endian/cock fixes by Eldad Zack - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller, Yamaha THRxx devices - HDSPM: updates for TCO controls by Adrian Knoth - ASoC: Add a snd_soc_component object type for generic handling of SoC and off-SoC components by Kuninori Morimoto, - dmaengine: a large set of cleanups and conversions by Lars-Peter Clausen - ASoC DAPM: performance optimizations from Ryo Tsutsui - ASoC DAPM: support for mixer control sharing by Stephen Warren - ASoC: multiplatform ARM cleanups from Arnd Bergmann - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack" * tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits) ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats ALSA: pcm_format_to_bits strong-typed conversion ALSA: compress: fix the states to check for allowing read ALSA: hda - Move Thinkpad X220 to use auto parser ALSA: USB: adjust for changed 3.8 USB API ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources sound: oss/dmabuf: use dma_map_single ALSA: ali5451: use mdelay instead of large udelay constants ALSA: hda - Add the support for ALC286 codec ALSA: usb-audio: USB quirk for Yamaha THR10C ALSA: usb-audio: USB quirk for Yamaha THR5A ALSA: usb-audio: USB quirk for Yamaha THR10 ALSA: usb-audio: Fix autopm error during probing ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT ALSA: sound kconfig typo ALSA: emu10k1: Fix dock firmware loading ASoC: ux500: forward declare msp_i2s_platform_data ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers ...
| * | ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatchEldad Zack2013-04-301-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively (neglecting the index): be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Daniel Mack confirmed that the driver works on little endian CPUs, leading to the conclusion that the device-side structure is actually little endian. This changes the code to use le16_to_cpu(). Caught by sparse. Acked-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: asihpi: add format support check in snd_card_asihpi_capture_formatsEldad Zack2013-04-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some Asihpi formats are not supported or invalid, and their mapping to ALSA format is set to -1. Before performing the format conversion into ALSA bitwise formats, add a consistency check for the requested format, as done in snd_card_asihpi_playback_formats(). Compile tested only. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: pcm_format_to_bits strong-typed conversionEldad Zack2013-04-297-11/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add a function to handle conversion from snd_pcm_format_t to bitwise with proper typing. Change such conversions to use this function and silence sparse warnings. Signed-off-by: Eldad Zack <eldad@fogrefinery.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: compress: fix the states to check for allowing readVinod Koul2013-04-291-2/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul@intel.com> Cc: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Move Thinkpad X220 to use auto parserDavid Henningsson2013-04-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This enables better volume controls than the current model parser. Also, because the original quirk for X220 was added to fix docking station support, add the TP410 fixup instead. Reported-by: Willian Jon McCann <william.jon.mccann@gmail.com> Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: USB: adjust for changed 3.8 USB APIClemens Ladisch2013-04-297-12/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The recent changes in the USB API ("implement new semantics for URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the default, and changed this flag to mean that URBs can be delayed. This is not the behaviour wanted by any of the audio drivers because it leads to discontinuous playback with very small period sizes. Therefore, our URBs need to be submitted without this flag. Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org> Cc: <stable@vger.kernel.org> # 3.8 only Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sourcesDavid Henningsson2013-04-261-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Scarlett 2i2 seems to take almost 500 ms to set the sample rate, even if the clock is currently set to that value. This patch speeds up prepare of the device, by avoiding setting the clock to something it already is. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | sound: oss/dmabuf: use dma_map_singleArnd Bergmann2013-04-261-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The virt_to_bus/bus_to_virt functions have been deprecated for as long as I can remember, and they are used in very few remaining instances, usually in obscure ISA device drivers. The OSS sound drivers are the only ones that are still used on the ARM architecture, and only on some of the earliest StrongARM machines. The problem for converting the OSS subsystem to use dma_map_single instead is that the caller of virt_to_bus does not have a device pointer, since the subsystem has never been ported to use the common device infrastructure. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: ali5451: use mdelay instead of large udelay constantsArnd Bergmann2013-04-261-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | ARM cannot handle udelay for more than 2 miliseconds, so we should use mdelay instead for those. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge tag 'asoc-v3.10-3' of ↵Takashi Iwai2013-04-2510-166/+64
| |\ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: More updates for v3.10 A few more fixes, nothing too major though the DMA changes fix modular builds.
| | * \ Merge remote-tracking branch 'asoc/topic/ux500' into asoc-nextMark Brown2013-04-231-0/+1
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| | | * | ASoC: ux500: forward declare msp_i2s_platform_dataArnd Bergmann2013-04-231-0/+1
| | | |/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We get a lot of build warnings from the msp driver like: In file included from sound/soc/ux500/ux500_msp_dai.h:21:0, from sound/soc/ux500/mop500.c:25: sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: 'struct msp_i2s_platform_data' declared inside parameter list [enabled by default] struct msp_i2s_platform_data *platform_data); ^ sound/soc/ux500/ux500_msp_i2s.h:546:11: warning: its scope is only this definition or declaration, which is probably not what you want [enabled by default] The easiest solution is to add a declaration of the struct name. Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | Merge remote-tracking branch 'asoc/topic/max98088' into asoc-nextMark Brown2013-04-231-1/+1
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| | * \ \ Merge remote-tracking branch 'asoc/topic/fsl' into asoc-nextMark Brown2013-04-231-0/+8
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| | * \ \ \ Merge remote-tracking branch 'asoc/topic/dma' into asoc-nextMark Brown2013-04-2323-579/+532
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| | | * | | | ASoC: generic-dmaengine-pcm: call dma_request_slave_channel()Shawn Guo2013-04-231-6/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | dma_request_slave_channel() is a more appropriate API for dmaengine clients that adopt generic DMA bindings to call. Let's use it instead of of_dma_request_slave_channel() to save <linux/of_dma.h> include. Signed-off-by: Shawn Guo <shawn.guo@linaro.org> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: generic-dmaengine-pcm: use a more common dma nameShawn Guo2013-04-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The examples in Documentation/devicetree/bindings/dma/dma.txt recommends the name for dma channel doing both RX and TX to be "rx-tx". This becomes a common pattern that has been adopted by platforms that converts to generic DMA bindings. Let's follow this common pattern in generic-dmaengine-pcm. Signed-off-by: Shawn Guo <shawn.guo@linaro.org> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | ASoC: mxs: Use generic dmaengine PCMLars-Peter Clausen2013-04-222-126/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use the generic dmaengine PCM driver instead of a custom implementation. 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>
| | | * | | | ASoC: mxs: Setup dma data in DAI probeLars-Peter Clausen2013-04-221-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This allows us to access the DAI DMA data when we create the PCM. We'll use this when converting mxs to generic DMA engine PCM driver. 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>
| | | * | | | ASoC: mxs-pcm: Set SNDRV_PCM_INFO_HALF_DUPLEXLars-Peter Clausen2013-04-221-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The MXS SAIF is only half-duplex so set the SNDRV_PCM_INFO_HALF_DUPLEX flag for the PCM in order to prevent playback and capture from running at the same time. 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>
| | | * | | | ASoC: generic-dmaengine-pcm: Add support for half-duplexLars-Peter Clausen2013-04-221-13/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some platforms which are half-duplex share the same DMA channel between the playback and capture stream. Add support for this to the generic dmaengine PCM driver. 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>
| | * | | | | Merge remote-tracking branch 'asoc/topic/davinci' into asoc-nextMark Brown2013-04-23143-1479/+3189
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| | | * | | | | ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modesMichal Bachraty2013-04-232-4/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For TDM mode, BCLK-to-LCLK ratio is computed as (tdm_slots) x (word_length). I2S mode is only subset of TDM mode with specific tdm_slots = 2 channels. Also bclk_lrclk_ratio can be greater than 255, therefore u16 need to be used. Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com> Acked-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | | * | | | | ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializersMichal Bachraty2013-04-233-10/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As pointed of by Vaibhav, commit message: "ASoC: davinci-mcasp: Add support for multichannel playback" number of active serializers can be hidden into fifo_level variable, which is set in davimci-mcasp. Signed-off-by: Michal Bachraty <michal.bachraty@streamunlimited.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * | | | | | Merge remote-tracking branch 'asoc/topic/cs4271' into asoc-nextMark Brown2013-04-231-70/+96
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| | * \ \ \ \ \ \ Merge remote-tracking branch 'asoc/topic/core' into asoc-nextMark Brown2013-04-235-20/+42
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| * | | | | | | | ALSA: hda - Add the support for ALC286 codecKailang Yang2013-04-251-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | It's yet another ALC269-variant. Signed-off-by: Kailang Yang <kailang@realtek.com> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | ALSA: usb-audio: USB quirk for Yamaha THR10CTrulan Martin2013-04-251-0/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds a USB quirk for the Yamaha THR10C amp. Signed-off-by: Trulan Martin <trulanm@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | ALSA: usb-audio: USB quirk for Yamaha THR5ATrulan Martin2013-04-251-0/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds a USB quirk for the Yamaha THR5A amp. Signed-off-by: Trulan Martin <trulanm@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | ALSA: usb-audio: USB quirk for Yamaha THR10Trulan Martin2013-04-251-0/+26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds a USB quirk for the Yamaha THR10 amp. Signed-off-by: Trulan Martin <trulanm@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | | | ALSA: usb-audio: Fix autopm error during probingTakashi Iwai2013-04-251-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We've got strange errors in get_ctl_value() in mixer.c during probing, e.g. on Hercules RMX2 DJ Controller: ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 .... It turned out that the culprit is autopm: snd_usb_autoresume() returns -ENODEV when called during card->probing = 1. Since the call itself during card->probing = 1 is valid, let's fix the return value of snd_usb_autoresume() as success. Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org> Cc: <stable@vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
generated by cgit v1.2.3 (git 2.39.1) at 2024-04-28 02:35:40 +0000