Commit Graph

134 Commits

Author SHA1 Message Date
Pierre-Louis Bossart 2d52a5abd5 ALSA: core: remove .wall_clock
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>
2015-02-20 17:30:07 +01:00
Pierre-Louis Bossart 3179f62001 ALSA: core: add .get_time_info
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>
2015-02-20 17:30:05 +01:00
Pierre-Louis Bossart 229d043096 ALSA: core: selection of audio_tstamp type and accuracy reports
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>
2015-02-20 17:30:03 +01:00
Pierre-Louis Bossart 2b79d7a6bf ALSA: pcm: allow for trigger_tstamp snapshot in .trigger
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>
2015-02-09 16:01:53 +01:00
Takashi Iwai a3ae255e37 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.
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Merge tag 'asoc-v3.20-2' of 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
2015-02-05 07:08:35 +01:00
Mark Brown 7d590e4623 Merge remote-tracking branch 'asoc/topic/pcm512x' into asoc-next 2015-02-04 20:57:06 +00:00
Takashi Iwai 3fe9cf390f Merge branch 'topic/snd-device' into for-next 2015-02-03 17:57:16 +01:00
Takashi Iwai ef46c7af93 ALSA: pcm: Embed struct device
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>
2015-02-02 14:42:42 +01:00
Peter Rosin f66f898e95 ALSA: pcm: Add snd_interval_ranges() and snd_pcm_hw_constraint_ranges()
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>
2015-01-28 19:27:15 +00:00
Lars-Peter Clausen adc2ae0bb6 ALSA: pcm: Remove unused functions declaration
Remove function declarations for functions that don't have a matching
implementation.

For snd_pcm_build_linear_format the implementation was removed in
64d27f96cb ("[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>
2015-01-03 20:59:57 +01:00
Libin Yang 48d882978e ALSA: pcm: add SNDRV_PCM_TRIGGER_DRAIN trigger
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>
2014-12-31 17:10:08 +01:00
Lars-Peter Clausen b3a8c862cd ALSA: pcm: Remove unused SNDRV_PCM_IOCTL1_{FALSE,TRUE} defines
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>
2014-12-30 16:41:54 +01:00
Lars-Peter Clausen 62f64a880a ALSA: pcm: Fix kerneldoc for params_*() functions
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>
2014-12-30 16:41:11 +01:00
Takashi Iwai 5031466387 Merge branch 'for-linus' into for-next
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.
2014-11-28 18:30:19 +01:00
Jussi Laako d42472ecff ALSA: pcm: Add big-endian DSD sample formats and fix XMOS DSD sample format
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>
2014-11-21 15:13:28 +01:00
Takashi Iwai 1fb8510cdb ALSA: pcm: Add snd_pcm_stop_xrun() helper
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>
2014-11-09 18:20:40 +01:00
Takashi Iwai 2b30d411db ALSA: pcm: Add xrun_injection proc entry
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>
2014-11-04 14:09:16 +01:00
Takashi Iwai 85926e0fe8 ALSA: pcm: Convert params_*() with static inline functions
... 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>
2014-10-30 15:42:32 +01:00
Takashi Iwai 30b771cf8c ALSA: pcm: More kerneldoc updates
Add proper kerneldoc comments to the exported functions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-10-30 15:42:27 +01:00
Takashi Iwai f213d8f79a ALSA: pcm: Use static inline for snd_pcm_lib_alloc_vmalloc_buffer()
... instead of #if 0 hack.  It's more straightforward and obvious.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-10-30 13:48:29 +01:00
Jurgen Kramer d4288d3fac ALSA: pcm: add new DSD sampleformat for native DSD playback on XMOS based 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>
2014-09-08 17:11:00 +02:00
Takashi Iwai 7af142f752 ALSA: pcm: Uninline snd_pcm_stream_lock() and _unlock()
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>
2014-09-03 14:04:18 +02:00
Takashi Iwai 257f8cce5d ALSA: pcm: Allow nonatomic trigger operations
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>
2014-09-03 14:04:08 +02:00
Mark Brown 0ac8a52d45 ALSA: Provide a CLOCK_MONOTONIC_RAW timestamp type
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>
2014-07-10 16:58:13 +02:00
Thomas Gleixner 26204e048d ALSA: core: Use ktime_get_ts()
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>
2014-06-12 12:58:16 +02:00
Takashi Iwai 09e56df8b3 ALSA: pcm: Use standard printk helpers
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>
2014-02-14 08:14:15 +01:00
Mark Brown 2f43a23ab9 Merge remote-tracking branch 'asoc/topic/pcm' into for-tiwai 2014-01-16 12:42:57 +00:00
Lars-Peter Clausen e3a9269f87 ALSA: Add helper function for intersecting two rate masks
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>
2014-01-14 20:42:34 +00:00
Takashi Iwai 47d98c026e ALSA: Remove memory reservation code from memalloc helper
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>
2014-01-09 07:32:10 +01:00
Lars-Peter Clausen e6c2e7eb27 ALSA: Constify the snd_pcm_substream struct ops field
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>
2013-05-24 15:41:44 +02:00
Linus Torvalds 9992ba7232 sound updates for v3.10-rc1
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
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Merge tag 'sound-3.10' of 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
  ...
2013-05-03 09:10:23 -07:00
Eldad Zack 74c34ca1cc ALSA: pcm_format_to_bits strong-typed conversion
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>
2013-04-29 13:36:15 +02:00
Masanari Iida b23f7a09f9 treewide: Fix typo in printk and comments
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>
2013-04-24 16:43:00 +02:00
Daniel Mack ef7a4f979b ALSA: add DSD formats
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>
2013-04-18 10:02:33 +02:00
Yacine Belkadi eb7c06e8e9 ALSA: add/change some comments describing function return values
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>
2013-03-12 08:32:53 +01:00
Pierre-Louis Bossart 4eeaaeaea1 ALSA: core: add hooks for audio timestamps
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>
2012-10-23 16:13:48 +02:00
Pierre-Louis Bossart 0e8014d772 ALSA: core: keep track of boundary wrap-around
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>
2012-10-23 16:13:41 +02:00
Linus Torvalds f5a246eab9 Sound updates for 3.7-rc1
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.
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Merge tag 'sound-3.7' of 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
  ...
2012-10-09 07:07:14 +09:00
David Howells a1ce39288e UAPI: (Scripted) Convert #include "..." to #include <path/...> in kernel 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>
2012-10-02 18:01:25 +01:00
Takashi Iwai 9d069dc00b ALSA: Make snd_sgbuf_get_{ptr|addr}() available for non-SG cases
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>
2012-09-23 11:24:42 +02:00
Takashi Iwai 2d3391ec0e ALSA: PCM: channel mapping API implementation
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>
2012-09-06 18:01:16 +02:00
Randy Dunlap 8513915acc ALSA: fix pcm.h kernel-doc warning and notation
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>
2012-08-20 10:26:47 +02:00
Takashi Iwai 4609ed6b1f 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
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Merge tag 'asoc-3.6' of 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
2012-07-19 08:03:20 +02:00
Mark Brown 1464189f8c ALSA: pcm: Make constraints lists const
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>
2012-07-05 14:19:39 +02:00
Dimitris Papastamos 4be77a530b ALSA: pcm: Add snd_pcm_rate_bit_to_rate()
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>
2012-06-18 09:38:58 +02:00
Ola Lilja 1aad779fcc ALSA: pcm: Add debug-print helper function
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>
2012-06-03 13:06:38 +01:00
Takashi Iwai cb3f2adc03 Merge branch 'topic/asoc' into for-linus 2012-03-18 18:22:37 +01:00
Mark Brown 4af87a939e ALSA: pcm: Constify the list in snd_pcm_hw_constraint_list
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>
2012-03-15 07:35:17 +01:00
Liam Girdwood 945e503845 ALSA: PCM - Add PCM creation API for internal PCMs.
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>
2012-02-09 09:20:22 +01:00
Linus Torvalds 68d99b2c8e Merge branch 'for-linus' of 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
  ...
2011-10-28 14:25:01 -07:00