Commit Graph

144 Commits

Author SHA1 Message Date
Liam Girdwood
c147c0e17b ASoC: topology: Add topology UAPI header
The ASoC topology UAPI header defines the structures
 required to define any DSP firmware audio topology and control objects from
 userspace.

The following objects are supported :-
 o kcontrols including TLV controls.
 o DAPM widgets and graph elements
 o Vendor bespoke objects.
 o Coefficient data
 o FE PCM capabilities and config.
 o BE link capabilities and config.
 o Codec <-> codec link capabilities and config.
 o Topology object manifest.

The file format is simple and divided into blocks for each object type and
each block has a header that defines it's size and type. Blocks can be in
any order of type and can either all be in a single file or spread across
more than one file. Blocks also have a group identifier ID so that they can
be loaded and unloaded by ID.

Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
2015-06-03 21:01:01 +01:00
Vinod Koul
43c499dc27 ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN
we have defined SNDRV_CTL_ELEM_ID_NAME_MAXLEN as size of name array so use
this define instead of numeric value

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-04-18 09:05:55 +02:00
Masanari Iida
1a6ab46fa9 ALSA: Fix spelling typo in Documentation/DocBook/alsa-driver-api.xml
This patch fix spelling typo found in alsa-driver-api.xml.
It is because this file is generated from comments in source files,
I have to fix source files.

Signed-off-by: Masanari Iida <standby24x7@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-03-04 12:12:59 +01:00
Takashi Iwai
72f770c6ac Merge branch 'topic/timestamp' into for-next 2015-02-23 09:15:02 +01:00
Takashi Iwai
88cacc57e8 Merge branch 'topic/uapi-fix' into for-next 2015-02-23 09:14:03 +01:00
Pierre-Louis Bossart
c72638bdaa ALSA: bump PCM protocol to 2.0.13
Bump PCM protocol to enable use of STATUS_EXT ioctls, older
apps will still use STATUS and audio timestamp configuration
is not supported (backwards compatible behavior).

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
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
Mikko Rapeli
b9956409c2 include/uapi/sound/emu10k1.h: include sound/asound.h
Fixes userspace compilation errors like:
error: field ‘id’ has incomplete type
struct snd_ctl_elem_id id;  /* full control ID definition */

Signed-off-by: Mikko Rapeli <mikko.rapeli@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-02-17 07:45:23 +01:00
Mikko Rapeli
bbf91c1c5b include/uapi/sound/asequencer.h: include sound/asound.h
Fixes userspace compilation error:
error: unknown type name ‘snd_seq_client_type_t’
snd_seq_client_type_t type; /* client type */

Signed-off-by: Mikko Rapeli <mikko.rapeli@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-02-17 07:44:06 +01:00
Mikko Rapeli
4bebf7091a include/uapi/sound/asound.h: include stdlib.h in userspace
Fixes compiler errors like:
error: field ‘trigger_tstamp’ has incomplete type
error: invalid application of ‘sizeof’ to incomplete t
ype ‘struct timespec’

Signed-off-by: Mikko Rapeli <mikko.rapeli@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-02-17 07:43:54 +01:00
Mikko Rapeli
76a3aeac2f hdspm.h: include stdint.h in userspace
Fixes compilation error:

sound/hdspm.h:43:2: error: unknown type name ‘uint32_t’

Signed-off-by: Mikko Rapeli <mikko.rapeli@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-02-17 07:43:25 +01:00
Takashi Iwai
5da7f924a4 ALSA: usx2y: Move UAPI definition into include/uapi/sound/usb_stream.h
The user-space API definition for usb_stream stuff should be moved
to include/uapi/sound to be exposed publicly.

While we're at it, add the missing ifdef guard for double inclusion,
too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2015-01-28 17:33:49 +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
Takashi Sakamoto
8985f4ac1c ALSA: oxfw: Add hwdep interface
This interface is designed for mixer/control application. By using this
interface, an application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-12-10 10:50:00 +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
ddcecf6b6a ALSA: Fix invalid kerneldoc markers
They are no real kerneldoc comments, so drop such markers.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-11-11 09:39:17 +01:00
Takashi Iwai
f533ccb61e ALSA: doc: Fix uapi/sound/compress_offload.h kerneldoc comments
so that make htmldocs works properly.
Since kerneldoc can't handle noname enum properly, name enum
sndrv_compress_encoder.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-10-29 08:22:05 +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
589008106b ALSA: pcm: Introduce protocol version field to sw_params
For controlling the new fields more strictly, add sw_params.proto
field indicating the protocol version of the user-space.  User-space
should fill the SNDRV_PCM_VERSION value it's built with, then kernel
can know whether the new fields should be evaluated or not.

And now tstamp_type field is evaluated only when the valid value is
set there.  This avoids the wrong override of tstamp_type to zero,
which is SNDRV_PCM_TSTAMP_TYPE_GETTIMEOFDAY.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-21 09:07:46 +02:00
Takashi Iwai
5646eda585 ALSA: pcm: Add timestamp type to sw_params
For allowing adjusting the timestamp type on the fly, add it to
sw_params.  The existing ioctl is still kept for compatibility.

Along with this, increment the PCM protocol version.

The extension was suggested by Clemens Ladisch.

Acked-by: Jaroslav Kysela <perex@perex.cz>
Reviewed-by: Mark Brown <broonie@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-07-10 16:58:15 +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
Vinod Koul
2da38e0c94 ALSA: compress: fix the struct alignment to 4 bytes
In 64bit systems the compiler can default align to 8bytes causing mis-match with
32bit usermode. Avoid this is future by ensuring all the structures shared with
usermode are packed and aligned to 4 bytes irrespective of arch used

[coding style fixes by tiwai]

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-23 12:15:33 +02:00
Wang, Xiaoming
2bd0ae464a ALSA: compress: Cancel the optimization of compiler and fix the size of struct for all platform.
Cancel the optimization of compiler for struct snd_compr_avail
which size will be 0x1c in 32bit kernel while 0x20 in 64bit
kernel under the optimizer. That will make compaction between
32bit and 64bit. So add packed to fix the size of struct
snd_compr_avail to 0x1c for all platform.

Signed-off-by: Zhang Dongxing <dongxing.zhang@intel.com>
Signed-off-by: xiaoming wang <xiaoming.wang@intel.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-06-12 11:55:41 +02:00
Takashi Sakamoto
618eabeae7 ALSA: bebob: Add hwdep interface
This interface is designed for mixer/control application. By using hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:31:03 +02:00
Takashi Sakamoto
555e8a8f7f ALSA: fireworks: Add command/response functionality into hwdep interface
This commit adds two functionality for hwdep interface, adds two parameters for
this driver, add a node for proc interface.

To receive responses from devices, this driver already allocate own callback
into initial memory space in host controller. This means no one can allocate
its own callback to the address. So this driver must give a way for user
applications to receive responses.

This commit adds a functionality to receive responses via hwdep interface. The
application can receive responses to read from this interface. To achieve this,
this commit adds a buffer to queue responses. The default size of this buffer is
1024 bytes. This size can be changed to give preferrable size to
'resp_buf_size' parameter for this driver. The application should notice rest
of space in this buffer because this driver don't push responses when this
buffer has no space.

Additionaly, this commit adds a functionality to transmit commands via hwdep
interface. The application can transmit commands to write into this interface.
I note that the application can transmit one command at once, but can receive
as many responses as possible untill the user-buffer is full.

When using these interfaces, the application must keep maximum number of
sequence number in command within the number in firewire.h because this driver
uses this number to distinguish the response is against the command by the
application or this driver.

Usually responses against commands which the application transmits are pushed
into this buffer. But to enable 'resp_buf_debug' parameter for this driver, all
responses are pushed into the buffer. When using this mode, I reccomend to
expand the size of buffer.

Finally this commit adds a new node into proc interface to output status of the
buffer.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:58 +02:00
Takashi Sakamoto
594ddced82 ALSA: fireworks: Add hwdep interface
This interface is designed for mixer/control application. To use hwdep
interface, the application can get information about firewire node, can
lock/unlock kernel streaming and can get notification at starting/stopping
kernel streaming.

Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-05-26 14:28:41 +02:00
Vinod Koul
929559be6d ALSA: compress: add num_sample_rates in snd_codec_desc
this gives ability to convey the valid values of supported rates in
sample_rates array

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-01-07 18:33:40 +01:00
Vinod Koul
b8bab04829 ALSA: compress: update struct snd_codec_desc for sample rate
Now that we don't use SNDRV_PCM_RATE_xxx bit fields for sample rate, we need to
change the description to an array for describing the sample rates supported by
the sink/source

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-01-05 11:58:27 +01:00
Vinod Koul
d9afee6904 ALSA: compress: update comment for sample rate in snd_codec
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2014-01-05 11:58:18 +01:00
Vinod Koul
f0e9c08065 ALSA: compress: change the way sample rates are sent to kernel
The usage of SNDRV_RATES is not effective as we can have rates like 12000 or
some other ones used by decoders. This change the usage of this to use the raw
Hz values to be sent to kernel

Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-16 15:54:28 +01:00
Takashi Iwai
6733cf572a ALSA: compress: Fix 64bit ABI incompatibility
snd_pcm_uframes_t is defined as unsigned long so it would take
different sizes depending on 32 or 64bit architectures.  As we don't
want this ABI incompatibility, and there is no real 64bit user yet,
let's make it the fixed size with __u32.

Also bump the protocol version number to 0.1.2.

Acked-by: Vinod Koul <vinod.koul@intel.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-12-10 15:32:44 +01:00
Chen Gang
3b098eb486 ALSA: include/uapi/sound/firewire.h: use "_UAPI" instead of "UAPI"
When installing, "scripts/headers_install.sh" will strip guard macro'
"_UAPI" to prevent from appearing it to users. And also, all another
files which need uapi prefix always use "_UAPI", not "UAPI".

So use "_UAPI" instead of "UAPI" on the guard macro, and also give a
comment for "#endif".

Signed-off-by: Chen Gang <gang.chen@asianux.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-11-07 10:28:54 +01:00
Takashi Iwai
861e66d341 Merge branch 'dice-driver-playback-only' of git://git.alsa-project.org/alsa-kprivate into for-next 2013-10-22 10:02:57 +02:00
Clemens Ladisch
82fbb4f7b4 ALSA: add DICE driver
As a start point for further development, this is an incomplete driver
for DICE devices:
- only playback (so no clock source except the bus clock)
- only 44.1 kHz
- no MIDI
- recovery after bus reset is slow
- hwdep device is created, but not actually implemented

Contains compilation fixes by Stefan Richter.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-10-17 21:18:32 +02:00
Adrian Knoth
b43dd416be ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC
Use struct hdspm_ltc to query the LTC, using a mixer struct is just
plain wrong.

Due to the wrong struct, this ioctl was never working, so we're free to
fix it without breaking userspace compatibility.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-08-19 20:09:09 +02:00
Takashi Iwai
975cc02a90 ALSA: Replace the magic number 44 with const
The char arrays with size 44 are for the name string of
snd_ctl_elem_id.  Define the constant and replace the raw numbers with
it for clarifying better.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-28 12:14:44 +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
Jeeja KP
9727b490e5 ALSA: compress: add support for gapless playback
this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track

Also bump the compress API version

Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-02-14 12:30:22 +01:00
Takashi Iwai
7cc17a31ff ALSA: Extend chmap definitions for UAC2
USB audio class 2 has more channel map positions than we currently
have.  Let's add missing definitions.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-11-26 16:18:59 +01:00
Clemens Ladisch
9c7066aef4 ALSA: core: fix 64-bit SNDRV_PCM_IOCTL_STATUS ABI breakage
Commit 4eeaaeaea (ALSA: core: add hooks for audio timestamps) added the
new audio_tstamp field to struct snd_pcm_status.  However, struct
timespec requires 64-bit alignment, so the 64-bit compiler would insert
32 bits of padding before this field, which broke SNDRV_PCM_IOCTL_STATUS
with error messages like this:

  kernel: unknown ioctl = 0x80984120

To solve this, insert the padding explicitly so that it can be taken
into account when calculating the ABI structure size.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2012-10-28 09:52:37 +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
David Howells
674e95ca44 UAPI: (Scripted) Disintegrate include/sound
Signed-off-by: David Howells <dhowells@redhat.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Acked-by: Thomas Gleixner <tglx@linutronix.de>
Acked-by: Michael Kerrisk <mtk.manpages@gmail.com>
Acked-by: Paul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: Dave Jones <davej@redhat.com>
2012-10-09 09:49:13 +01:00
David Howells
4413e16d9d UAPI: (Scripted) Set up UAPI Kbuild files
Set up empty UAPI Kbuild files to be populated by the header splitter.

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 18:01:35 +01:00