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* ALSA: core: remove .wall_clockPierre-Louis Bossart2015-02-201-2/+0
| | | | | | | can be removed without breaking git-bisect now Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: core: add .get_time_infoPierre-Louis Bossart2015-02-201-0/+4
| | | | | | | | | | | Introduce more generic .get_time_info to retrieve system timestamp and audio timestamp in single routine. Backwards compatibility is preserved with same functionality as with .wall_clock method (to be removed in following commits to avoid breaking git bisect) Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: core: selection of audio_tstamp type and accuracy reportsPierre-Louis Bossart2015-02-201-0/+60
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Audio timestamps can be extracted from sample counters, wall clocks, PHC clocks (Ethernet AVB), on-demand synchronized snapshots. This patch provides the ability to report timestamping capabilities, select timestamp types and retrieve timestamp accuracy, if supported. Details can be found in Documentations/sound/alsa/timestamping.txt This functionality is introduced by reclaiming the reserved_aligned field introduced by commit9c7066aef4a5eb8e4063de28f06c508bf6f2963a in snd_pcm_status to provide userspace with selection/query capabilities. Additional driver_tstamp and audio_tstamp_accuracy fields are also added. snd_pcm_mmap_status remains a read-only structure with only the audio timestamp value accessible from user space. The selection of audio timestamp type is done through snd_pcm_status only This commit does not impact ABI and does not impact the default behavior. By default audio timestamp is aligned with hw_pointer and reports the DMA position. Backwards compatibility is handled by using the HDAudio wall clock for playback and the hw_ptr for all other cases. For timestamp selection a new STATUS_EXT ioctl is introduced with read/write parameters. Alsa-lib will be modified to make use of STATUS_EXT. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: allow for trigger_tstamp snapshot in .triggerPierre-Louis Bossart2015-02-091-0/+1
| | | | | | | | | Don't use generic snapshot of trigger_tstamp if low-level driver or hardware can get a more precise value for better audio/system time synchronization. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-v3.20-2' of ↵Takashi Iwai2015-02-051-0/+12
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for v3.20 More updates for v3.20: - Lots of refactoring from Lars-Peter Clausen, moving drivers to more data driven initialization and rationalizing a lot of DAPM usage. - Much improved handling of CDCLK clocks on Samsung I2S controllers. - Lots of driver specific cleanups and feature improvements. - CODEC support for TI PCM514x and TLV320AIC3104 devices. - Board support for Tegra systems with Realtek RT5677. Conflicts: sound/soc/intel/sst-mfld-platform-pcm.c
| * Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-nextMark Brown2015-02-041-0/+12
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| | * ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges()Peter Rosin2015-01-281-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add helper functions to allow drivers to specify several disjoint ranges for a variable. In particular, there is a codec (PCM512x) that has a hole in its supported range of rates, due to PLL and divider restrictions. This is like snd_pcm_hw_constraint_list(), but for ranges instead of points. Signed-off-by: Peter Rosin <peda@axentia.se> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
* | | Merge branch 'topic/snd-device' into for-nextTakashi Iwai2015-02-031-1/+1
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| * | | ALSA: pcm: Embed struct deviceTakashi Iwai2015-02-021-1/+1
| |/ / | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Like previous patches, at this time we embed the struct device into PCM object. However, this needs a bit more caution: struct snd_pcm doesn't own one device but two, for both playback and capture! Thus not struct snd_pcm but struct snd_pcm_str object contains the device. Along with this change, pcm->dev field is dropped for avoiding confusion. It was meant to point to a non-standard parent. But, since now we can touch each struct device directly, we can manipulate the parent field easily there, too. Reviewed-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: pcm: Remove unused functions declarationLars-Peter Clausen2015-01-031-7/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Remove function declarations for functions that don't have a matching implementation. For snd_pcm_build_linear_format the implementation was removed in 64d27f96cb719cf8 ("[ALSA] Support 3-bytes 24bit format in PCM OSS emulation"). All the others never had one (as far as git history goes). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: pcm: add SNDRV_PCM_TRIGGER_DRAIN triggerLibin Yang2014-12-311-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add SNDRV_PCM_TRIGGER_DRAIN trigger for pcm drain. Some audio devices require notification of drain events in order to properly drain and shutdown an audio stream. Signed-off-by: Libin Yang <libin.yang@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | ALSA: pcm: Remove unused SNDRV_PCM_IOCTL1_{FALSE,TRUE} definesLars-Peter Clausen2014-12-301-3/+0
|/ / | | | | | | | | | | | | | | Both SNDRV_PCM_IOCTL1_FALSE and SNDRV_PCM_IOCTL1_TRUE are unused and have in fact never been used (at least as far as the git history goes). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* / ALSA: pcm: Fix kerneldoc for params_*() functionsLars-Peter Clausen2014-12-301-5/+5
|/ | | | | | | | Fix a copy and paste error in the kernel doc description for the params_*() functions. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' into for-nextTakashi Iwai2014-11-281-0/+2
|\ | | | | | | | | | | The commit [7a2e9ddc: ALSA: usb-audio: Add native DSD support for Denon/Marantz DACs] requires the new format definition that has landed only in for-next branch.
| * ALSA: pcm: Add big-endian DSD sample formats and fix XMOS DSD sample formatJussi Laako2014-11-211-0/+2
| | | | | | | | | | | | | | | | | | This patch fixes XMOS DSD sample format to DSD_U32_BE and also adds DSD_U16_BE and DSD_U32_BE sample formats. Signed-off-by: Jussi Laako <jussi@sonarnerd.net> Acked-by: Jurgen Kramer <gtmkramer@xs4all.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Add snd_pcm_stop_xrun() helperTakashi Iwai2014-11-091-0/+1
| | | | | | | | | | | | | | | | | | | | | | Add a new helper function snd_pcm_stop_xrun() to the standard sequnce lock/snd_pcm_stop(XRUN)/unlock by a single call, and replace the existing open codes with this helper. The function checks the PCM running state to prevent setting the wrong state, too, for more safety. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Add xrun_injection proc entryTakashi Iwai2014-11-041-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | This patch adds a new proc entry for PCM substreams to inject an XRUN. When a PCM substream is running and any value is written to its xrun_injection proc file, the driver triggers XRUN. This is a useful feature for debugging XRUN and error handling code paths. Note that this entry is enabled only when CONFIG_SND_PCM_XRUN_DEBUG is set. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Convert params_*() with static inline functionsTakashi Iwai2014-10-301-12/+53
| | | | | | | | | | | | | | | | ... and add proper kerneldoc comments. There is no big reason to keep them as macros. Static inline functions are safer in general, and suitable for kerneldoc, too. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: More kerneldoc updatesTakashi Iwai2014-10-301-7/+153
| | | | | | | | | | | | Add proper kerneldoc comments to the exported functions. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Use static inline for snd_pcm_lib_alloc_vmalloc_buffer()Takashi Iwai2014-10-301-12/+13
|/ | | | | | ... instead of #if 0 hack. It's more straightforward and obvious. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: add new DSD sampleformat for native DSD playback on XMOS based ↵Jurgen Kramer2014-09-081-0/+1
| | | | | | | | | | | | | | devices XMOS based USB DACs with native DSD support expose this feature via a USB alternate setting. The audio format is either 32-bit raw or a 32-bit PCM format. To utilize this feature on linux this patch introduces a new 32-bit DSD sampleformat DSD_U32_LE. A follow up patch will add a quirk for XMOS based devices to utilize the new format. Further patches will add support to alsa-lib. Signed-off-by: Jurgen Kramer <gtmkramer@xs4all.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: Uninline snd_pcm_stream_lock() and _unlock()Takashi Iwai2014-09-031-69/+12
| | | | | | | | | | | The previous commit for the non-atomic PCM ops added more codes to snd_pcm_stream_lock() and its variants. Since they are inlined functions, it resulted in a significant code size bloat. For reducing the size bloat, this patch changes the inline functions to the normal function calls. The export of rwlock and rwsem are removed as well, since they are referred only in pcm_native.c now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: Allow nonatomic trigger operationsTakashi Iwai2014-09-031-12/+46
| | | | | | | | | | | | | | | | | | | | | | | | | | Currently, many PCM operations are performed in a critical section protected by spinlock, typically the trigger and pointer callbacks are assumed to be atomic. This is basically because some trigger action (e.g. PCM stop after drain or xrun) is done in the interrupt handler. If a driver runs in a threaded irq, however, this doesn't have to be atomic. And many devices want to handle trigger in a non-atomic context due to lengthy communications. This patch tries all PCM calls operational in non-atomic context. What it does is very simple: replaces the substream spinlock with the corresponding substream mutex when pcm->nonatomic flag is set. The driver that wants to use the non-atomic PCM ops just needs to set the flag and keep the rest as is. (Of course, it must not handle any PCM ops in irq context.) Note that the code doesn't check whether it's atomic-safe or not, but trust in 100% that the driver sets pcm->nonatomic correctly. One possible problem is the case where linked PCM substreams have inconsistent nonatomic states. For avoiding this, snd_pcm_link() returns an error if one tries to link an inconsistent PCM substream. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Provide a CLOCK_MONOTONIC_RAW timestamp typeMark Brown2014-07-101-2/+9
| | | | | | | | | | | | | | | | | For applications which need to synchronise with external timebases such as broadcast TV applications the kernel monotonic time is not optimal as it includes adjustments from NTP and so may still include discontinuities due to that. A raw monotonic time which does not include any adjustments is available in the kernel from getrawmonotonic() so provide userspace with a new timestamp type SNDRV_PCM_TSTAMP_TYPE_MONOTONIC_RAW which provides timestamps based on this as an option. [dropped tstamp_type assignment code, as it's no longer needed -- tiwai] Reported-by: Daniel Thompson <daniel.thompson@linaro.org> Signed-off-by: Mark Brown <broonie@linaro.org> Acked-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: core: Use ktime_get_ts()Thomas Gleixner2014-06-121-1/+1
| | | | | | | | do_posix_clock_monotonic_gettime() is a leftover from the initial posix timer implementation which maps to ktime_get_ts(). Signed-off-by: Thomas Gleixner <tglx@linutronix.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: Use standard printk helpersTakashi Iwai2014-02-141-0/+8
| | | | | | | | | Use dev_err() & co as much as possible. If not available (no device assigned at the calling point), use pr_xxx() helpers instead. For simplicity, introduce new helpers for pcm stream, pcm_err(), etc. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwaiMark Brown2014-01-161-0/+2
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| * ALSA: Add helper function for intersecting two rate masksLars-Peter Clausen2014-01-141-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | A bit of special care is necessary when creating the intersection of two rate masks. This comes from the special meaning of the SNDRV_PCM_RATE_CONTINUOUS and SNDRV_PCM_RATE_KNOT bits, which needs special handling when intersecting two rate masks. SNDRV_PCM_RATE_CONTINUOUS means the hardware supports all rates in a specific interval. SNDRV_PCM_RATE_KNOT means the hardware supports a set of discrete rates specified by a list constraint. For all other cases the supported rates are specified directly in the rate mask. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org>
* | ALSA: Remove memory reservation code from memalloc helperTakashi Iwai2014-01-091-1/+0
|/ | | | | | | | | | | Nowadays we have CMA for obtaining the contiguous memory pages efficiently. Let's kill the old kludge for reserving the memory pages for large buffers. It was rarely useful (only for preserving pages among module reloading or a little help by an early boot scripting), used only by a couple of drivers, and yet it gives too much ugliness than its benefit. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: Constify the snd_pcm_substream struct ops fieldLars-Peter Clausen2013-05-241-2/+3
| | | | | | | | | The ops field of the snd_pcm_substream struct is never modified inside the ALSA core. Making it const allows drivers to declare their snd_pcm_ops struct as const. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'sound-3.10' of ↵Linus Torvalds2013-05-031-9/+22
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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: pcm_format_to_bits strong-typed conversionEldad Zack2013-04-291-0/+6
| | | | | | | | | | | | | | | | | | | | | | 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: add DSD formatsDaniel Mack2013-04-181-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds two formats for Direct Stream Digital (DSD), a pulse-density encoding format which is described here: https://en.wikipedia.org/wiki/Direct_Stream_Digital DSD operates on 2.8, 5.6 or 11.2MHz sample rates and as a 1-bit stream. The two new types added by this patch describe streams that are capable of handling DSD samples in DOP format as 8-bit or in 16-bit (or at a x8 or x16 data rate, respectively). DSD itself specifies samples in *bit*, while DOP and ALSA handle them as *bytes*. Hence, a factor of 8 or 16 has to be applied for the sample rare configuration, according to the following table: configured hardware 176.4KHz 352.8kHz 705.6KHz <---- sample rate 8-bit 2.8MHz 5.6MHz 16-bit 2.8Mhz 5.6MHz 11.2MHz `-----------------------------' actual DSD sample rates Signed-off-by: Daniel Mack <zonque@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: add/change some comments describing function return valuesYacine Belkadi2013-03-121-9/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | script/kernel-doc reports the following type of warnings (when run in verbose mode): Warning(sound/core/init.c:152): No description found for return value of 'snd_card_create' To fix that: - add missing descriptions of function return values - use "Return:" sections to describe those return values Along the way: - complete some descriptions - fix some typos Signed-off-by: Yacine Belkadi <yacine.belkadi.1@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | treewide: Fix typo in printk and commentsMasanari Iida2013-04-241-1/+1
|/ | | | | | | Fix typo in printk and comments within various drivers. Signed-off-by: Masanari Iida <standby24x7@gmail.com> Signed-off-by: Jiri Kosina <jkosina@suse.cz>
* ALSA: core: add hooks for audio timestampsPierre-Louis Bossart2012-10-231-0/+2
| | | | | | | | | | | | | | | | | | | ALSA did not provide any direct means to infer the audio time for A/V sync and system/audio time correlations (eg. PulseAudio). Applications had to track the number of samples read/written and add/subtract the number of samples queued in the ring buffer. This accounting led to small errors, typically several samples, due to the two-step process. Computing the audio time in the kernel is more direct, as all the information is available in the same routines. Also add new .audio_wallclock routine to enable fine-grain synchronization between monotonic system time and audio hardware time. Using the wallclock, if supported in hardware, allows for a much better sub-microsecond precision and a common drift tracking for all devices sharing the same wall clock (master clock). Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: core: keep track of boundary wrap-aroundPierre-Louis Bossart2012-10-231-0/+1
| | | | | | | | | | Keep track of boundary crossing when hw_ptr exceeds boundary limit and wraps-around. This will help keep track of total number of frames played/received at the kernel level Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'sound-3.7' of ↵Linus Torvalds2012-10-091-25/+62
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This contains pretty many small commits covering fairly large range of files in sound/ directory. Partly because of additional API support and partly because of constantly developed ASoC and ARM stuff. Some highlights: - Introduced the helper function and documentation for exposing the channel map via control API, as discussed in Plumbers; most of PCI drivers are covered, will follow more drivers later - Most of drivers have been replaced with the new PM callbacks (if the bus is supported) - HD-audio controller got the support of runtime PM and the support of D3 clock-stop. Also changing the power_save option in sysfs kicks off immediately to enable / disable the power-save mode. - Another significant code change in HD-audio is the rewrite of firmware loading code. Other than that, most of changes in HD-audio are continued cleanups and standardization for the generic auto parser and bug fixes (HBR, device-specific fixups), in addition to the support of channel-map API. - Addition of ASoC bindings for the compressed API, used by the mid-x86 drivers. - Lots of cleanups and API refreshes for ASoC codec drivers and DaVinci. - Conversion of OMAP to dmaengine. - New machine driver for Wolfson Microelectronics Bells. - New CODEC driver for Wolfson Microelectronics WM0010. - Enhancements to the ux500 and wm2000 drivers - A new driver for DA9055 and the support for regulator bypass mode." Fix up various arm soc header file reorg conflicts. * tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits) ALSA: hda - Add new codec ALC283 ALC290 support ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls ALSA: hda - fix indices on boost volume on Conexant ALSA: aloop - add locking to timer access ALSA: hda - Fix hang caused by race during suspend. sound: Remove unnecessary semicolon ALSA: hda/realtek - Fix detection of ALC271X codec ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310 ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event ALSA: hda - make a generic unsol event handler ASoC: codecs: Add DA9055 codec driver ASoC: eukrea-tlv320: Convert it to platform driver ALSA: ASoC: add DT bindings for CS4271 ASoC: wm_hubs: Ensure volume updates are handled during class W startup ASoC: wm5110: Adding missing volume update bits ASoC: wm5110: Add OUT3R support ASoC: wm5110: Add AEC loopback support ASoC: wm5110: Rename EPOUT to HPOUT3 ASoC: arizona: Add more clock rates ASoC: arizona: Add more DSP options for mixer input muxes ...
| * ALSA: Make snd_sgbuf_get_{ptr|addr}() available for non-SG casesTakashi Iwai2012-09-231-25/+14
| | | | | | | | | | | | | | | | | | | | Passing struct snd_dma_buffer pointer instead, so that they work no matter whether real SG buffer is used or not. This is a preliminary work for the HD-audio DSP loader code. Signed-off-by: Ian Minett <ian_minett@creativelabs.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: PCM: channel mapping API implementationTakashi Iwai2012-09-061-0/+48
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch implements the basic data types for the standard channel mapping API handling. - The definitions of the channel positions and the new TLV types are added in sound/asound.h and sound/tlv.h, so that they can be referred from user-space. - Introduced a new helper function snd_pcm_add_chmap_ctls() to create control elements representing the channel maps for each PCM (sub)stream. - Some standard pre-defined channel maps are provided for convenience. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | UAPI: (Scripted) Convert #include "..." to #include <path/...> in kernel ↵David Howells2012-10-021-1/+1
|/ | | | | | | | | | | | system headers Convert #include "..." to #include <path/...> in kernel system headers. Signed-off-by: David Howells <dhowells@redhat.com> Acked-by: Arnd Bergmann <arnd@arndb.de> Acked-by: Thomas Gleixner <tglx@linutronix.de> Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com> Acked-by: Dave Jones <davej@redhat.com>
* ALSA: fix pcm.h kernel-doc warning and notationRandy Dunlap2012-08-201-1/+2
| | | | | | | | | | Fix kernel-doc warning in <sound/pcm.h> and add function name to make the kernel-doc notation complete. Warning(include/sound/pcm.h:1081): No description found for parameter 'substream' Signed-off-by: Randy Dunlap <rdunlap@xenotime.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge tag 'asoc-3.6' of ↵Takashi Iwai2012-07-191-0/+11
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next ASoC: Updates for 3.6 This has been a pretty quiet release - very little activity in framework terms, mostly just a few new drivers and updates: - Added the ability to add and remove DAPM paths dynamically, mostly for reparenting on clock changes. - New machine drivers for Marvell Brownstone, ST-Ericsson Ux500 reference platform and ttc-dkp. - New CPU drivers for Blackfin BF6xx SPORTs in I2S mode, Marvell MMP, Synopsis Designware I2S controllers, and SPEAr DMA and S/PDIF - New CODEC drivers for Dialog DA732x, ST STA529, ST-Ericsson AB8500, TI Isabelle and Wolfson Microelectronics WM5102 and WM5110
| * ALSA: pcm: Add debug-print helper functionOla Lilja2012-06-031-0/+11
| | | | | | | | | | | | | | | | | | Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja <ola.o.lilja@stericsson.com> Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ALSA: pcm: Make constraints lists constMark Brown2012-07-051-1/+1
| | | | | | | | | | | | | | They aren't modified by the core so the drivers can declare them const. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Add snd_pcm_rate_bit_to_rate()Dimitris Papastamos2012-06-181-0/+1
|/ | | | | | | | | | | This is essentially the reverse of snd_pcm_rate_to_rate_bit(). This is generally useful as the Compress API uses the rate bit directly and it helps to be able to map back to the actual sample rate. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'topic/asoc' into for-linusTakashi Iwai2012-03-181-0/+4
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| * ALSA: PCM - Add PCM creation API for internal PCMs.Liam Girdwood2012-02-091-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The new ASoC dynamic PCM core needs to create PCMs and substreams that are for use by internal ASoC drivers only and not visible to userspace for direct IO. These new PCMs are similar to regular PCMs expect they have no device nodes or procfs entries. The ASoC component drivers use them in exactly the same way as regular PCMs for PCM and DAI operations. The intention is that a dynamic PCM based driver will register both regular PCMs and internal PCMs. The regular PCMs will be used for all IO with userspace however the internal PCMs will be used by the driver to route digital audio through numerous back end DAI links (with potentially a DSP providing different hw_params, DAI formats based on the regular front end PCM params) to devices like CODECs, MODEMs, Bluetooth, FM, DMICs, etc This patch adds a new snd_pcm_new_internal() API call to create the internal PCM without device nodes or procfs. It also adds adds a new internal flag to snd_pcm. [fixed minor coding-style issues by tiwai] Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | ALSA: pcm: Constify the list in snd_pcm_hw_constraint_listMark Brown2012-03-151-2/+3
|/ | | | | | | | | Allows the constraint lists to be declared const by drivers which seems reasonable; there's plenty of other constification we could do if we were being complete but this was easy and quick. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' of ↵Linus Torvalds2011-10-281-0/+4
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits) ALSA: hda - Fix ADC input-amp handling for Cx20549 codec ALSA: hda - Keep EAPD turned on for old Conexant chips ALSA: hda/realtek - Fix missing volume controls with ALC260 ASoC: wm8940: Properly set codec->dapm.bias_level ALSA: hda - Fix pin-config for ASUS W90V ALSA: hda - Fix surround/CLFE headphone and speaker pins order ALSA: hda - Fix typo ALSA: Update the sound git tree URL ALSA: HDA: Add new revision for ALC662 ASoC: max98095: Convert codec->hw_write to snd_soc_write ASoC: keep pointer to resource so it can be freed ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2 ASoC: da7210: Add support for line out and DAC ASoC: da7210: Add support for DAPM ALSA: hda/realtek - Fix DAC assignments of multiple speakers ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value ASoC: Set sgtl5000->ldo in ldo_regulator_register ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture ...
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