1. 05 Feb, 2007 1 commit
    • Andy Wingo's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c · 451ff2f9
      Andy Wingo authored
      Original commit message from CVS:
      2007-02-05  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/audio/gstbaseaudiosink.c
      (gst_base_audio_sink_callback): Update basesink->offset so that we
      pull monotonically increasing offsets instead of, um, seeking back
      to 0 each time. Fixes alsasrc ! alsasink!
      451ff2f9
  2. 12 Jan, 2007 1 commit
    • Andy Wingo's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c · d853b238
      Andy Wingo authored
      Original commit message from CVS:
      2007-01-12  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/audio/gstbaseaudiosink.c
      (gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
      (gst_base_audio_sink_activate_pull): Remove the handwavey nego
      stuff, as the base class handles this now. Actually tell the ring
      buffer to start.
      (gst_base_audio_sink_callback): Cast the ring buffer correctly.
      How did this work before? Maybe I'm not as awesome a programmer as
      I think.
      
      * gst-libs/gst/audio/gstbaseaudiosrc.c
      (gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
      of a pad function.
      d853b238
  3. 06 Jan, 2007 1 commit
    • Andy Wingo's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init) · 85aee8e2
      Andy Wingo authored
      Original commit message from CVS:
      2007-01-06  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/audio/gstbaseaudiosink.c
      (gst_base_audio_sink_class_init)
      (gst_base_audio_sink_init):
      (gst_base_audio_sink_activate_pull): Add an activate_pull function
      to baseaudiosink, and tell basesink that we can work in pull mode.
      This way the ring buffer thread drives the pipeline directly, if
      pull mode is possible. There is some lingering nastiness regarding
      capsnego, however.
      (gst_base_audio_sink_callback): Implement the callback to pull
      data. This interface is a bit light, though -- it should get a
      GstFlowReturn return value at least.
      85aee8e2
  4. 13 Nov, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate... · 0990cbf2
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_event), (gst_base_audio_sink_render):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Make the clock sync code more accurate wrt resampling and playback
      at different rates.
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
      * gst-libs/gst/audio/gstringbuffer.h:
      Use better algorithm to interpolate sample rates.
      0990cbf2
  5. 18 Oct, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event. · 1166abbc
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_event), (gst_base_audio_sink_render):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Extract rate from the NEWSEGMENT event.
      Use commit_full to also take rate adjustment into account when writing
      samples to the ringbuffer.
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
      (gst_ring_buffer_read):
      * gst-libs/gst/audio/gstringbuffer.h:
      Added _commit_full() to also take rate into account.
      Use simple interpolation algorithm to resample audio.
      API: gst_ring_buffer_commit_full()
      * tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
      * tests/examples/seek/seek.c: (segment_done):
      Don't try to seek with 0.0 rate, just pause instead.
      Remove bogus debug line.
      1166abbc
  6. 13 Oct, 2006 1 commit
    • Ville Syrjala's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created. · 9b139e41
      Ville Syrjala authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_setcaps):
      Don't crash when ringbuffer is not yet created.
      Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
      Fixes #361634.
      * gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
      * gst/playback/gststreamselector.c:
      (gst_stream_selector_request_new_pad):
      Activate pads befre adding them to running elements.
      9b139e41
  7. 05 Oct, 2006 1 commit
    • Tim-Philipp Müller's avatar
      Printf format fixes. · 9e107d67
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * ext/alsa/gstalsadeviceprobe.c:
      (gst_alsa_device_property_probe_get_values):
      * ext/alsa/gstalsasink.c: (set_hwparams):
      * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
      (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
      * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
      (gst_ogg_mux_process_best_pad):
      * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
      (gst_ogg_parse_chain):
      * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
      * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
      * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
      (gst_vorbis_enc_buffer_check_discontinuous):
      * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      * gst-libs/gst/cdda/gstcddabasesrc.c:
      (gst_cdda_base_src_handle_track_seek):
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_push_full):
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
      * gst/audioresample/resample.c: (resample_input_pushthrough):
      * gst/playback/gstplaybasebin.c: (queue_out_of_data):
      * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
      * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
      (wavpack_type_find):
      * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
      * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
      (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
      * tests/check/elements/volume.c: (GST_START_TEST):
      Printf format fixes.
      9e107d67
  8. 28 Sep, 2006 1 commit
  9. 27 Sep, 2006 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING. · 1980f167
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      Add some more info in a WARNING.
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_create):
      Handle PAUSE in create function, use new -core addition to
      wait for playing. Fixes pausing and resuming capture from an
      audiosrc.
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
      (gst_ring_buffer_read):
      Constify some more.
      Caller supports interrupted reads now.
      1980f167
    • Wim Taymans's avatar
      Added docs for the audio libs. · 73677225
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * gst-libs/gst/audio/gstaudioclock.c:
      * gst-libs/gst/audio/gstaudioclock.h:
      * gst-libs/gst/audio/gstaudiosink.c:
      * gst-libs/gst/audio/gstaudiosink.h:
      * gst-libs/gst/audio/gstaudiosrc.c:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      * gst-libs/gst/audio/gstringbuffer.h:
      Added docs for the audio libs.
      73677225
  10. 15 Sep, 2006 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes... · 65b1938b
      Wim Taymans authored
      gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
      (gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
      (gst_base_audio_src_create), (gst_base_audio_src_change_state):
      Do the delay calculation in the source/sink base classes as this is
      specific for the capture/playback mode.
      Try to fixate a bit better, like round depth up to a multiple of 8
      bigger than width.
      Handle underruns correctly by marking DISCONT on buffers and adjusting
      timestamps to handle the gap.
      Set offset/offset_end correctly on buffers.
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
      (gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
      (gst_ring_buffer_read):
      Remove resync and underrun recovery from the ringbuffer.
      Fix ringbuffer read code on under/overrun.
      65b1938b
    • Wim Taymans's avatar
      configure.ac: We require 0.10.10.1 now because of _wait_preroll(). · 557b3672
      Wim Taymans authored
      Original commit message from CVS:
      * configure.ac:
      We require 0.10.10.1 now because of _wait_preroll().
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      Use gst_base_sink_wait_preroll().
      557b3672
  11. 24 Jul, 2006 2 commits
  12. 06 Jul, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass) · fa5dacc9
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init),
      (gst_base_audio_sink_provide_clock):
      Use gobject_class instead of G_OBJECT_CLASS (klass)
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init), (gst_base_audio_src_init),
      (gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
      (gst_base_audio_src_get_time),
      (gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
      (gst_base_audio_src_create_ringbuffer):
      Fix latency and buffer-time constants and properties ala basesink.
      Implement pull based scheduling. Fixes #346527.
      Set default blocksize in GstBaseSrc to 0, we default to pushing out
      one segment.
      Refuse slaving to another clock instead of silently not working.
      Only provide a clock when we are actually able to do so.
      Various small cleanups and compiler hints.
      fa5dacc9
  13. 03 Jun, 2006 1 commit
    • Jan Schmidt's avatar
      gst-libs/gst/audio/: Document better the fact that latency_time and... · 45e06fe7
      Jan Schmidt authored
      gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
      (gst_ring_buffer_samples_done):
      * gst-libs/gst/audio/gstringbuffer.h:
      Document better the fact that latency_time and buffer_time are values
      stored in microseconds, and not the usual GStreamer nanoseconds.
      Change the variables (compatibly) that store them from GstClockTime
      to guint64 to make it more clear that they're not storing clock times.
      Also, remove the bogus property description that says the user can
      specify -1 to get the default value, since that's never been the case.
      When computing the default segment size for the ring buffer, make it
      an integer number of samples.
      When the sub-class indicates a delay greater than the number of
      samples we've written return 0 from the audio sink get_time method.
      45e06fe7
  14. 28 Apr, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more... · 102b79e4
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
      
      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_render), (gst_base_audio_sink_change_state):
      patch to make timestamp checking more tollerant to rounding
      errors given that real discontinuities are to be marked on
      buffers. Fixes some asf files and #338778.
      Also avoid some crashers when we receive an event in the
      NULL state.
      102b79e4
  15. 10 Apr, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not... · 35058f78
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_event):
      Starting the ringbuffer when we did not acquire it can cause
      a deadlock, is pointless and causes nasty things for
      subclasses.
      Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
      35058f78
  16. 23 Mar, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to... · 227474e4
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
      (gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
      Implement new async_play vmethod to start slaving and allow
      playback start in case of async PLAY state changes.
      * gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
      Enable QoS with new method in base class.
      227474e4
  17. 22 Mar, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we... · 747d560f
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_dispose):
      Since we _parent the ringbuffer, we also need to
      _unparent instead of a plain _unref.
      747d560f
  18. 17 Mar, 2006 1 commit
  19. 28 Feb, 2006 1 commit
  20. 17 Feb, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/TODO: Update TODO · 454618e9
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/TODO:
      Update TODO
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_get_offset):
      When trying to play samples ASAP and we don't have a
      previous sample, try to play at position 0 instead of
      an invalid position.
      454618e9
  21. 13 Feb, 2006 1 commit
  22. 12 Feb, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible. · 0be7d56e
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
      (gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
      (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
      Use scale functions when possible.
      Fix error messages.
      Free clockid when after waiting for EOS.
      Use G_(UN_)LIKLY when it makes sense.
      Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
      0be7d56e
  23. 02 Feb, 2006 3 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess... · 260b5295
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_provide_clock):
      Ugh.. getting late I guess...
      260b5295
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we... · c78a5d7e
      Wim Taymans authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_provide_clock),
      (gst_base_audio_sink_set_property),
      (gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
      Don't try to provide a clock when we are not negotiated since
      we might not be able to make it run.
      c78a5d7e
    • Wim Taymans's avatar
      gst-libs/gst/audio/TODO: Updated. · 416c011f
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/TODO:
      Updated.
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_drain), (gst_base_audio_sink_event):
      On EOS, wait till the last sample is played before posting EOS.
      416c011f
  24. 30 Jan, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion. · a169abc6
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiosink.c:
      (gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
      (gst_audioringbuffer_pause):
      Implement pause that does not wait for completion.
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
      Don't drop buffers when going to PAUSED but perform preroll on
      remaining samples now that core base class supports this.
      
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
      (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
      (gst_ring_buffer_commit):
      Pause should not signal waiters.
      Implement return value of _commit correctly.
      a169abc6
  25. 29 Jan, 2006 1 commit
    • Sébastien Moutte's avatar
      gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES) · dc46970c
      Sébastien Moutte authored
      Original commit message from CVS:
      * gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
      * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
      use of gst_guint64_to_gdouble to be compliant with vs6
      * gst/playback/gstdecodebin.c: (try_to_link_1)
      * gst/videorate/videorate.c: (gst_video_rate_blank_data)
      use of G_GINT64_CONSTANT for int64 constants
      * win32/common/libgstinterfaces.def:
      export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
      * win32/vs6:
      update and add new project files
      dc46970c
  26. 25 Jan, 2006 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause. · ccd05fa0
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
      Undo previous commit, it breaks resume after pause.
      ccd05fa0
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging. · 2bc5ca17
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
      (gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
      Improve debugging.
      Post error when caps cannot be parsed.
      Resync on discontinuity in the stream.
      Clip samples to segment boundaries.
      return WRONG_STATE sooner when we are flushing.
      
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
      (gst_base_audio_src_get_time), (gst_base_audio_src_create):
      Make audiosrc operate in TIME.
      Set TIMESTAMP and DURATION on buffers.
      2bc5ca17
  27. 17 Jan, 2006 1 commit
    • Jan Schmidt's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised... · 04333a56
      Jan Schmidt authored
      gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_render):
      Fix playback of non-synchronised streams by assuming a rate
      of 1.0 instead of a random one.
      
      Makes this work again:
      
      gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
      endianness=(int)4321, signed=(boolean)true, width=(int)16,
      depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
      audioresample ! alsasink
      04333a56
  28. 20 Dec, 2005 2 commits
    • Thomas Vander Stichele's avatar
      stop making fun of older compilers · 01bc68f9
      Thomas Vander Stichele authored
      Original commit message from CVS:
      stop making fun of older compilers
      01bc68f9
    • Thomas Vander Stichele's avatar
      gst-libs/gst/audio/: update strings, values are in microseconds change the... · b4b2b62a
      Thomas Vander Stichele authored
      gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
      
      Original commit message from CVS:
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      update strings, values are in microseconds
      change the default sink buffer time to something that is smaller
      (to help software volume mixing have a slightly lower delay) but
      still be acceptable on Wim's laptop
      b4b2b62a
  29. 06 Dec, 2005 1 commit
  30. 28 Nov, 2005 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/TODO: Updated TODO · 3f05db18
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/TODO:
      Updated TODO
      
      * gst-libs/gst/audio/gstaudiosink.c:
      (gst_audioringbuffer_open_device),
      (gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
      (gst_audioringbuffer_release):
      Small cleanups.
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
      (gst_base_audio_sink_change_state):
      Slave to the master clock when going to PLAYING and unslave when
      going to PAUSED.
      
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
      (gst_ring_buffer_acquire), (gst_ring_buffer_release),
      (gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
      (gst_ring_buffer_clear_all), (wait_segment),
      (gst_ring_buffer_commit), (gst_ring_buffer_read),
      (gst_ring_buffer_advance):
      * gst-libs/gst/audio/gstringbuffer.h:
      Add some docs and cleanups.
      3f05db18
  31. 23 Nov, 2005 1 commit
  32. 22 Nov, 2005 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosink.c: And we provide a clock by default, of course... · af2acb6e
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
      (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
      And we provide a clock by default, of course...
      af2acb6e
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstaudioclock.c: This clock can be slaved to a master clock now. · a3cb4d49
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
      This clock can be slaved to a master clock now.
      
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
      (gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
      (gst_base_audio_sink_set_clock),
      (gst_base_audio_sink_set_property),
      (gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
      (gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      Handle slaving the internal clock to the clock selected in the
      pipeline.
      Add property to make the basesink not provide a clock.
      
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
      (gst_base_rtp_depayload_wait):
      * gst-libs/gst/rtp/gstbasertpdepayload.h:
      We can use the clock in GstElement, no need to store it ourselves.
      a3cb4d49