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* 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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| * | | | | | | vm: convert snd_pcm_lib_mmap_iomem() to vm_iomap_memory() helperLinus Torvalds2013-04-191-10/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This is my example conversion of a few existing mmap users. The pcm mmap case is one of the more straightforward ones. Acked-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
* | | | | | | | 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>
* | | | | | | | ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINTDaniel Mack2013-04-251-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually stuffed directly after the standard USB endpoint descriptor, and this is where the driver currently expects it to be. There are, however, devices in the wild that have it the other way around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes *before* the standard enpoint. Devices known to implement it that way are "Sennheiser BTD-500" and Plantronics USB headsets. When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to change sample rates, as the bitmask for the validity of this command is storen in bmAttributes of that descriptor. Fix this by searching the entire interface instead of just the extra bytes of the first endpoint, in case the latter fails. Signed-off-by: Daniel Mack <zonque@gmail.com> Reported-and-tested-by: Torstein Hegge <hegge@resisty.net> Reported-and-tested-by: Yves G <alsa-user@vivigatt.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: sound kconfig typoPavel Machek2013-04-241-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix english in sound/drivers/Kconfig. Signed-off-by: Pavel Machek <pavel@ucw.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: emu10k1: Fix dock firmware loadingTakashi Iwai2013-04-241-6/+33
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit [b209c4df: ALSA: emu10k1: cache emu1010 firmware] broke the firmware loading of the dock, just (mistakenly) ignoring a different firmware for docks on some models. This patch revives them again. Bugzilla: https://bugs.archlinux.org/task/34865 Reported-and-tested-by: Tobias Powalowski <tobias.powalowski@googlemail.com> Cc: <stable@vger.kernel.org> [v3.8+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: hda - Limit internal mic boost for a few Asus machinesDavid Henningsson2013-04-221-1/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These are being reported as being so noisy at high mic boost levels, so they are unusable in practice. Therefore artificially limit the boosts. BugLink: https://bugs.launchpad.net/bugs/1089795 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 msDaniel Schürmann2013-04-221-1/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Set the timeout for USB control set messages according to the USB 2 spec, using the macros from include/linux/usb.h. The get timout becomes 5000 ms even though it is 500 ms in the spec. This patch is required to run the Hercules RMX2 which needs a timeout of 1240 ms. More notes from author: I still distinguish between set and get but as long both are 5000 ms GCC will remove it anyway. IMHO this is more easy read and there is no need to explain why we use a get timeout for set messages. Signed-off-by: Daniel Schürmann <daschuer@mixxx.org> Acked-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress: Use kzalloc() for ioctls writing back dataTakashi Iwai2013-04-221-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Like the previous patch by Dan, we should clear the data to be returned from certain compress ioctls, namely, snd_compr_get_codec_caps() and snd_compr_get_params(). This time, we can simply replace kmalloc() with kzalloc(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress: info leak in snd_compr_get_caps()Dan Carpenter2013-04-221-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If the ->get_caps() function doesn't clear the buffer then there would stack information leaked to userspace. For example, soc_compr_get_caps() can return success without clearing the buffer. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress_core: Rework writes to use cumulative valuesCharles Keepax2013-04-211-7/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch reworks the writes to use cumulative values thus making the app_pointer unecessary and removing it. Only tested as far as build. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress_core: Remove unused hw_pointerCharles Keepax2013-04-211-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Only tested as far as build. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ASoC: soc-compress: Deduce stream directionCharles Keepax2013-04-211-1/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Previously we just hard coded all streams as playback streams, this patch checks the DAI to see if it is a capture or playback stream. It is worth noting that at this time only unidirectional streams are supported. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress_core: Add support for capture streamsCharles Keepax2013-04-211-2/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress_core: Deconstify copy callback bufferCharles Keepax2013-04-212-4/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The buffer passed to the copy callback should not be const because the copy callback can be used for capture and playback. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress_core: Calculate avail correctly for capture streamsCharles Keepax2013-04-211-11/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | ALSA: compress_core: Update calc_avail to use cumulative valuesCharles Keepax2013-04-211-18/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The app_pointer is managed locally by the compress core for memory mapped DSPs but for DSPs that are not memory mapped this would have to be manually updated from within the DSP driver itself, which is hardly very idiomatic. This patch switches to using the cumulative values to calculate the available buffer space because these are already gracefully passed out of the DSP driver to the compress core and otherwise should be functionally equivalent. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | Merge tag 'asoc-v3.10-2' of ↵Takashi Iwai2013-04-1826-457/+539
|\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: More updates for v3.10 The main additional change here is Lars-Peter's DMA work plus the platform conversions which have been tested - getting this in mainline will make life easier for development after the merge window. These factor a large chunk of code out of the drivers for the platforms using dmaengine, greatly simplifying development.
| * \ \ \ \ \ \ \ Merge remote-tracking branch 'asoc/topic/max98088' into asoc-nextMark Brown2013-04-181-7/+25
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| | * | | | | | | ASoC: max98088: Fix logging of hardware revision.Dylan Reid2013-04-171-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The hardware revision of the codec is based at 0x40. Subtract that before convering to ASCII. The same as it is done for 98095. Signed-off-by: Dylan Reid <dgreid@chromium.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| * | | | | | | | Merge remote-tracking branch 'asoc/topic/fsl' into asoc-nextMark Brown2013-04-185-5/+12
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| | * | | | | | | ASoC: fsl-ssi: Add SACNT definitionsMarkus Pargmann2013-04-161-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add definitions for AC97 control register. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Acked-by: Timur Tabi <timur@tabi.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | | | | | | Merge remote-tracking branch 'asoc/topic/dma' into asoc-nextMark Brown2013-04-1858-1079/+808
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