1. 09 Mar, 2009 2 commits
    • Jochen Keil's avatar
      Initial support for embedded cue sheets found in flac files · 706112bb
      Jochen Keil authored
      So far only seekpoints are supported, so no proper tagging yet
      except for track number and track length.
      Tagging should be done by parsing the cue sheet which
      is often embedded as vorbis comment in flac files.
      Furthermore the pathname should be configurable like "%A - %t - %T",
      where %A means Artist, %t track number and %T Title or so.
      706112bb
    • Jochen Keil's avatar
      decoder_plugin: added method container_scan() · ab3d89f4
      Jochen Keil authored
      [mk: fixed whitespace errors; use delete_song() instead of
      songvec_delete()]
      ab3d89f4
  2. 08 Mar, 2009 4 commits
    • Max Kellermann's avatar
      music_chunk: added assertions on the audio format · 94d1a87d
      Max Kellermann authored
      In !NDEBUG, remember which audio_format is stored in every chunk and
      every pipe.  Check the audio_format of every new data block appended
      to the music_chunk, and the format of every new chunk appended to the
      music_pipe.
      94d1a87d
    • Max Kellermann's avatar
      359f9871
    • Max Kellermann's avatar
      alsa: determine buffer_time if not already known · ab656a52
      Max Kellermann authored
      This patch fixes a theoretical (but practically impossible) flaw: the
      variable "buffer_time" may be uninitialized when it is used.
      Initialize the variable with snd_pcm_hw_params_get_buffer_time().
      ab656a52
    • Max Kellermann's avatar
      alsa: better period_time default value for high sample rates · 554a34fb
      Max Kellermann authored
      The default values for buffer_time and period_time were both capped by
      the hardware limits on practically all chips.  The result was a
      period_time which was half as big as the buffer_time.  On some chips,
      this led to lots of underruns when using a high sample rate (192 kHz),
      because MPD had very little time to send new samples to ALSA.
      
      A period time which is one fourth of the buffer time turned out to be
      much better.  If no period_time is configured, see how much
      buffer_time the hardware accepts, and try to configure one fourth of
      it as period_time, instead of hard-coding the default period_time
      value.
      
      This is yet another attempt to provide a solution which is valid for
      all sound chips.  Using the SND_PCM_NONBLOCK flag also seemed to solve
      the underruns, but put a lot more CPU load to MPD.
      554a34fb
  3. 07 Mar, 2009 17 commits
  4. 06 Mar, 2009 17 commits