linux-toradex.git/include/uapi/sound, branch v3.13-rc2

ALSA: include/uapi/sound/firewire.h: use "_UAPI" instead of "UAPI"

2013-11-07T09:28:54+00:00

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 
Acked-by: Clemens Ladisch 
Signed-off-by: Takashi Iwai

Merge branch 'dice-driver-playback-only' of git://git.alsa-project.org/alsa-kprivate into for-next

2013-10-22T08:02:57+00:00

ALSA: add DICE driver

2013-10-17T19:18:32+00:00

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

ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC

2013-08-19T18:09:09+00:00

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 
Signed-off-by: Takashi Iwai

ALSA: Replace the magic number 44 with const

2013-06-28T10:14:44+00:00

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

ALSA: add DSD formats

2013-04-18T08:02:33+00:00

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 
Signed-off-by: Takashi Iwai

ALSA: compress: add support for gapless playback

2013-02-14T11:30:22+00:00

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 
Signed-off-by: Vinod Koul 
Signed-off-by: Takashi Iwai

ALSA: Extend chmap definitions for UAC2

2012-11-26T15:18:59+00:00

USB audio class 2 has more channel map positions than we currently
have.  Let's add missing definitions.

Signed-off-by: Takashi Iwai

ALSA: core: fix 64-bit SNDRV_PCM_IOCTL_STATUS ABI breakage

2012-10-28T08:52:37+00:00

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 
Signed-off-by: Takashi Iwai

ALSA: core: add hooks for audio timestamps

2012-10-23T14:13:48+00:00

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 
Signed-off-by: Takashi Iwai