Commit Graph

17 Commits

Author SHA1 Message Date
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