28 Jun, 2013

1 commit


18 Apr, 2013

1 commit

  • 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
    Signed-off-by: Takashi Iwai

    Daniel Mack
     

14 Feb, 2013

1 commit

  • 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

    Jeeja KP
     

26 Nov, 2012

1 commit


28 Oct, 2012

1 commit

  • 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

    Clemens Ladisch
     

23 Oct, 2012

1 commit

  • 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

    Pierre-Louis Bossart
     

09 Oct, 2012

1 commit


03 Oct, 2012

1 commit