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>
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>
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>
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>
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>
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>
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
...
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>
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>
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>
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>
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>
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>
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
...
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>
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>
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>
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
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>
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>
Mark Brown
Thomas Gleixner
Takashi Iwai
Lars-Peter Clausen
Linus Torvalds
Eldad Zack
Masanari Iida
Daniel Mack
Yacine Belkadi
Pierre-Louis Bossart
David Howells
Randy Dunlap
Mark Brown
Dimitris Papastamos
Ola Lilja