1. 14 Apr, 2009 3 commits
    • Wim Taymans's avatar
      baseaudiosrc: adjust the internal timestamp · dffd1bcc
      Wim Taymans authored
      Adjust the internal timestamp before comparing it against the adjusted clock
      time.
      Fixes #578506
      dffd1bcc
    • Wim Taymans's avatar
      baseaudiosink: use new clock time methods · 0c4c1410
      Wim Taymans authored
      Use the unadjusted internal clock times to calculate the internal/external
      offset when calibrating the clock.
      
      When going to NULL, unparent and free the ringbuffer, like we do in the source
      element.
      See #578506
      0c4c1410
    • Wim Taymans's avatar
      audioclock: add methods for the internal offset · 4231d548
      Wim Taymans authored
      Add two methods for getting the unadjusted time of the clock and one for
      adjusting an internal time. We will need these methods for correctly handling
      the time after a gst_audio_clock_reset().
      
      Add a debug category and some debug lines to the audio clock.
      
      API: gst_audio_clock_get_time()
      API: gst_audio_clock_adjust()
      API: GST_AUDIO_CLOCK_CAST()
      4231d548
  2. 10 Apr, 2009 1 commit
  3. 09 Apr, 2009 1 commit
  4. 08 Apr, 2009 1 commit
    • Wim Taymans's avatar
      baseaudiosink: fix a small glitch after pause · cae2981f
      Wim Taymans authored
      After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
      the amount of output samples we consumed. We can't do this reliably with the
      current API when we are doing trick modes but we can do the right thing for
      normal playback.
      cae2981f
  5. 07 Apr, 2009 1 commit
  6. 25 Mar, 2009 1 commit
  7. 26 Feb, 2009 1 commit
  8. 02 Feb, 2009 1 commit
  9. 31 Jan, 2009 1 commit
  10. 23 Jan, 2009 1 commit
  11. 06 Jan, 2009 1 commit
    • José Alburquerque's avatar
      gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar*... · 74317892
      José Alburquerque authored
      gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
      
      Original commit message from CVS:
      Patch by: José Alburquerque <jaalburqu svn gnome org>
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
      * gst-libs/gst/audio/gstaudioclock.h:
      Make gst_audio_clock_new use const gchar* to ease the wrapping of
      C++ bindings. Fixes #566723.
      74317892
  12. 05 Jan, 2009 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when... · 0a4c1bc6
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_change_state):
      Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
      take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
      this because the async_play method is deprecated and usually not called
      anymore.
      0a4c1bc6
  13. 31 Dec, 2008 1 commit
    • Edward Hervey's avatar
      Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in... · e2fcc716
      Edward Hervey authored
      Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/Makefile.am:
      * gst-libs/gst/audio/audio.c:
      * gst-libs/gst/audio/multichannel.h:
      * gst-libs/gst/audio/testchannels.c:
      * win32/MANIFEST:
      * win32/common/audio-enumtypes.c:
      (gst_audio_channel_position_get_type),
      (gst_ring_buffer_state_get_type),
      (gst_ring_buffer_seg_state_get_type),
      (gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
      * win32/common/audio-enumtypes.h:
      * win32/common/multichannel-enumtypes.c:
      * win32/common/multichannel-enumtypes.h:
      * win32/vs6/grammar.dsp:
      * win32/vs6/libgstaudio.dsp:
      * win32/vs7/libgstaudio.vcproj:
      * win32/vs8/libgstaudio.vcproj:
      Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
      audio- in order to wrap all enums declarations of that library.
      This modification should not matter since that header file is not a
      public header (it will be included by public headers).
      Modify win32 crap^Wfiles accordingly.
      e2fcc716
  14. 30 Dec, 2008 1 commit
    • Edward Hervey's avatar
      gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting... · 20adaa13
      Edward Hervey authored
      gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Complete Sebastien's commit from the 13th by exporting the
      _slave_method_get_type() methods.
      20adaa13
  15. 20 Dec, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before... · a579eba7
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_setcaps):
      Pause the write thread before deactivating and releasing the ringbuffer
      to avoid a deadlock when we do gapless playback with different sample
      rates in playbin2.  Fixes #564929.
      a579eba7
  16. 19 Dec, 2008 1 commit
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type()... · 4ed1f5d6
      Sebastian Dröge authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      Make GstAudioSrcSlaveMethod get_type() function non-static
      as it's public now.
      * win32/common/libgstaudio.def:
      * win32/common/libgstnetbuffer.def:
      Add some missing functions to the list of exported symbols.
      4ed1f5d6
  17. 13 Dec, 2008 1 commit
    • Sebastian Dröge's avatar
      gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to... · 04d9ff9a
      Sebastian Dröge authored
      gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_slave_method_get_type),
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_slave_method_get_type),
      (gst_base_audio_src_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
      public API. This is needed for the C++ bindings to be able
      to use this base classes. Fixes bug #564200, #564206.
      04d9ff9a
  18. 27 Nov, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12... · af354dbe
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_get_offset), (gst_base_audio_src_create):
      Avoid nasty int overflows after about 12 hours and 25 minutes when these
      code paths are triggered.
      A free beer to Håvard Graff for finding this!
      af354dbe
  19. 25 Nov, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by... · 6983c1c8
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_drain), (gst_base_audio_sink_event),
      (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
      (gst_base_audio_sink_change_state):
      Really fix audiosink drain handling by keeping track of the running_time
      of the last sample.
      6983c1c8
  20. 24 Nov, 2008 2 commits
    • Stefan Kost's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove... · a8264f66
      Stefan Kost authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      Time is already in running_time. Remove base_time handling. Fixes
      audiosinks not draining and thus chopping some audio in the end.
      a8264f66
    • Stefan Kost's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for... · 7f937c99
      Stefan Kost authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      Add one log message to check for audio_drained. Sync one log message
      with the condition. Send EOS after draining audio in pull mode.
      7f937c99
  21. 10 Nov, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait... · e701e640
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
      (gst_base_audio_sink_callback):
      Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
      for the latency to expire, fixes #559567.
      e701e640
  22. 20 Oct, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to... · 6eed8ca2
      Wim Taymans authored
      gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiosink.c:
      (gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
      (gst_audioringbuffer_activate), (gst_audioringbuffer_release),
      (gst_audioringbuffer_stop):
      Implement a separate activate functions to start monitoring the segments
      or, in pull mode, pulling in data.
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
      (gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
      (gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
      (gst_base_audio_sink_activate_pull),
      (gst_base_audio_sink_async_play),
      (gst_base_audio_sink_change_state):
      Implement pad and element convert query function.
      Activate the ringbuffer.
      Use the segment last_stop value as the offset to pull.
      Use new basesink _do_preroll() method to preroll in the pulling thread.
      Take appropriate locking in the pulling thread.
      * gst-libs/gst/audio/gstringbuffer.h:
      Update some docs.
      6eed8ca2
  23. 17 Oct, 2008 1 commit
    • Wim Taymans's avatar
      Add methods to more accuratly control the pulling thread of a ringbuffer. · a6b78893
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
      (gst_ring_buffer_activate), (gst_ring_buffer_is_active):
      * gst-libs/gst/audio/gstringbuffer.h:
      Add methods to more accuratly control the pulling thread of a
      ringbuffer.
      Add format conversion helper code to the ringbuffer.
      API: GstRingBuffer:gst_ring_buffer_activate()
      API: GstRingBuffer:gst_ring_buffer_is_active()
      API: GstRingBuffer:gst_ring_buffer_convert()
      a6b78893
  24. 16 Oct, 2008 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we... · 92799960
      Wim Taymans authored
      gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
      (gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
      (gst_audioringbuffer_stop):
      Signal thread startup earlier so that we can immediatly go into pull
      mode when we have to and block on preroll.
      92799960
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to... · 7bd29abb
      Wim Taymans authored
      gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_prepare_read):
      In pull mode we want the callback to prepull a buffer we can preroll on
      even when we are not yet playing.
      7bd29abb
  25. 08 Oct, 2008 3 commits
  26. 13 Sep, 2008 1 commit
  27. 04 Sep, 2008 1 commit
  28. 26 Aug, 2008 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this... · da76d5e7
      Wim Taymans authored
      gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
      Since we now call stop, we trigger this code path that causes a deadlock
      is apparently not needed.
      da76d5e7
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer... · 44043261
      Wim Taymans authored
      gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
      (gst_ring_buffer_stop):
      Also allow the case where the ringbuffer was paused when we try to stop
      it so that the basesrc stop function is still called.
      44043261
  29. 13 Aug, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also... · 510a5bef
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      When not slaved to another clock also subtract the base_time from our
      internal clock time to get the running time.
      510a5bef
  30. 11 Aug, 2008 4 commits