1. 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
  2. 16 Jun, 2006 1 commit
    • Stefan Kost's avatar
      docs/libs/: add remaining symbols into correct setions · cade7911
      Stefan Kost authored
      Original commit message from CVS:
      * docs/libs/Makefile.am:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * docs/libs/gst-plugins-base-libs.types:
      add remaining symbols into correct setions
      * gst-libs/gst/audio/gstringbuffer.c:
      fix incomplete docs
      * gst-libs/gst/audio/gstringbuffer.h:
      comment out not yet implemented function
      * gst-libs/gst/floatcast/floatcast.h:
      * gst-libs/gst/netbuffer/gstnetbuffer.c:
      add short descriptions
      * gst-libs/gst/interfaces/propertyprobe.c:
      fix return value docs
      * gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
      simplify debug logging
      * gst-libs/gst/riff/riff-read.h:
      sync function prototype and docs
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      remove left over symbol
      cade7911
  3. 07 Jun, 2006 1 commit
    • Thomas Vander Stichele's avatar
      move last template doc snippets to source code and delete them · 51ca8fe3
      Thomas Vander Stichele authored
      Original commit message from CVS:
      * docs/libs/tmpl/gstaudio.sgml:
      * docs/libs/tmpl/gstcolorbalance.sgml:
      * docs/libs/tmpl/gstmixer.sgml:
      * docs/libs/tmpl/gstringbuffer.sgml:
      * docs/libs/tmpl/gsttuner.sgml:
      * docs/libs/tmpl/gstxoverlay.sgml:
      * gst-libs/gst/audio/audio.c:
      * gst-libs/gst/audio/gstringbuffer.c:
      * gst-libs/gst/interfaces/colorbalance.c:
      * gst-libs/gst/interfaces/mixer.c:
      * gst-libs/gst/interfaces/tuner.c:
      * gst-libs/gst/interfaces/xoverlay.c:
      move last template doc snippets to source code and delete them
      51ca8fe3
  4. 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
  5. 01 Jun, 2006 1 commit
    • Stefan Kost's avatar
      Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass · 131fb86b
      Stefan Kost authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixerelement.h:
      * ext/alsa/gstalsamixeroptions.h:
      * ext/alsa/gstalsamixertrack.h:
      * ext/gnomevfs/gstgnomevfssink.h:
      * ext/gnomevfs/gstgnomevfssrc.h:
      * ext/theora/gsttheoradec.h:
      * ext/theora/gsttheoraenc.h:
      * ext/theora/gsttheoraparse.h:
      * ext/vorbis/vorbisparse.h:
      * gst-libs/gst/audio/gstaudioclock.h:
      * gst-libs/gst/audio/gstaudiofilter.h:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.h:
      * gst/audioconvert/gstaudioconvert.h:
      * gst/audioresample/gstaudioresample.h:
      * gst/audiotestsrc/gstaudiotestsrc.h:
      * gst/ffmpegcolorspace/gstffmpegcolorspace.h:
      * gst/playback/gststreamselector.h:
      * gst/tcp/gstmultifdsink.h:
      * gst/tcp/gsttcpclientsink.h:
      * gst/tcp/gsttcpclientsrc.h:
      * gst/tcp/gsttcpserversink.h:
      * gst/tcp/gsttcpserversrc.h:
      * gst/videorate/gstvideorate.h:
      * gst/videoscale/gstvideoscale.h:
      * gst/videotestsrc/gstvideotestsrc.h:
      * gst/volume/gstvolume.h:
      * sys/v4l/gstv4ljpegsrc.h:
      * sys/v4l/gstv4lmjpegsink.h:
      * sys/v4l/gstv4lmjpegsrc.h:
      * sys/v4l/gstv4lsrc.h:
      * sys/ximage/ximagesink.h:
      * sys/xvimage/xvimagesink.h:
      * tests/old/testsuite/alsa/sinesrc.h:
      Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
      131fb86b
  6. 16 May, 2006 1 commit
    • Tim-Philipp Müller's avatar
      gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and... · 10d35563
      Tim-Philipp Müller authored
      gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/multichannel.c:
      (gst_audio_check_channel_positions):
      It's okay to have caps with channels=1 and a channel position
      different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
      (deinterleavers might want to keep the position in the caps,
      so that they can be re-interleaved again properly later).
      Leave check for unexpected 2-channel layouts intact for now.
      10d35563
  7. 28 Apr, 2006 4 commits
    • Stefan Kost's avatar
      make GstElementDetails const · e972defd
      Stefan Kost authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixerelement.c:
      * ext/alsa/gstalsasrc.c:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      * ext/gnomevfs/gstgnomevfssink.c:
      * ext/gnomevfs/gstgnomevfssrc.c:
      * ext/ogg/gstoggdemux.c:
      * ext/ogg/gstoggmux.c:
      * ext/ogg/gstoggparse.c:
      * ext/ogg/gstogmparse.c:
      * ext/pango/gstclockoverlay.c:
      * ext/pango/gsttextoverlay.c:
      * ext/pango/gsttextrender.c:
      * ext/pango/gsttimeoverlay.c:
      * ext/theora/theoradec.c:
      * ext/theora/theoraenc.c:
      * ext/vorbis/vorbisdec.c:
      * ext/vorbis/vorbisenc.c:
      * gst-libs/gst/audio/gstaudiofilter.c:
      * gst-libs/gst/audio/gstaudiofiltertemplate.c:
      * gst/audioconvert/gstaudioconvert.c:
      * gst/audiorate/gstaudiorate.c:
      * gst/audioresample/gstaudioresample.c:
      * gst/audiotestsrc/gstaudiotestsrc.c:
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      * gst/playback/gstdecodebin.c:
      * gst/playback/gstplaybin.c:
      * gst/playback/gststreamselector.c:
      * gst/subparse/gstsubparse.c:
      * gst/tcp/gstmultifdsink.c:
      * gst/tcp/gsttcpclientsink.c:
      * gst/tcp/gsttcpclientsrc.c:
      * gst/tcp/gsttcpserversink.c:
      * gst/tcp/gsttcpserversrc.c:
      * gst/typefind/gsttypefindfunctions.c: (plugin_init):
      * gst/videorate/gstvideorate.c:
      * gst/videoscale/gstvideoscale.c:
      * gst/videotestsrc/gstvideotestsrc.c:
      * gst/volume/gstvolume.c:
      * sys/v4l/gstv4ljpegsrc.c:
      * sys/v4l/gstv4lmjpegsink.c:
      * sys/v4l/gstv4lmjpegsrc.c:
      * sys/v4l/gstv4lsrc.c:
      * sys/ximage/ximagesink.c:
      * sys/xvimage/xvimagesink.c:
      * tests/check/libs/cddabasesrc.c:
      make GstElementDetails const
      e972defd
    • 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
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing. · 04754176
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
      (gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
      (gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
      (gst_ring_buffer_pause), (gst_ring_buffer_stop),
      (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
      (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
      (gst_ring_buffer_commit), (gst_ring_buffer_read),
      (gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
      (gst_ring_buffer_clear), (gst_ring_buffer_may_start):
      Check arguments passed to public functions instead of
      crashing.
      04754176
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work. · c068425b
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
      (gst_base_audio_src_get_time), (gst_base_audio_src_create):
      GstBaseAudioSrc must be live or it does not work.
      * gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
      Don't set live to TRUE as this is the default in the parentclass.
      c068425b
  8. 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
  9. 08 Apr, 2006 2 commits
    • Stefan Kost's avatar
      Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) · 0afac375
      Stefan Kost authored
      Original commit message from CVS:
      * ext/alsa/gstalsamixeroptions.c:
      (gst_alsa_mixer_options_class_init):
      * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
      * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
      * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
      * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
      * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
      * gst-libs/gst/audio/gstaudiofilter.c:
      (gst_audio_filter_class_init):
      * gst-libs/gst/audio/gstaudiosink.c:
      (gst_audioringbuffer_class_init):
      * gst-libs/gst/audio/gstaudiosrc.c:
      (gst_audioringbuffer_class_init):
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
      * gst-libs/gst/interfaces/colorbalancechannel.c:
      (gst_color_balance_channel_class_init):
      * gst-libs/gst/interfaces/mixeroptions.c:
      (gst_mixer_options_class_init):
      * gst-libs/gst/interfaces/mixertrack.c:
      (gst_mixer_track_class_init):
      * gst-libs/gst/interfaces/tunerchannel.c:
      (gst_tuner_channel_class_init):
      * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
      * gst-libs/gst/netbuffer/gstnetbuffer.c:
      (gst_netbuffer_class_init):
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_class_init):
      * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
      * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
      * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
      * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
      * gst/playback/gststreamselector.c:
      (gst_stream_selector_class_init):
      * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
      * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
      * sys/v4l/gstv4lcolorbalance.c:
      (gst_v4l_color_balance_channel_class_init):
      * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
      * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
      * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
      * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
      (gst_v4l_tuner_norm_class_init):
      * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
      * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
      Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
      0afac375
    • Stefan Kost's avatar
      Fix broken GObject macros · 1a2642a1
      Stefan Kost authored
      Original commit message from CVS:
      * ext/pango/gsttextrender.h:
      * gst-libs/gst/audio/gstaudiosink.h:
      * gst-libs/gst/audio/gstaudiosrc.h:
      * gst-libs/gst/audio/gstbaseaudiosink.h:
      * gst-libs/gst/audio/gstbaseaudiosrc.h:
      * gst-libs/gst/audio/gstringbuffer.h:
      * gst-libs/gst/rtp/gstbasertpdepayload.h:
      * gst-libs/gst/rtp/gstbasertppayload.h:
      * gst-libs/gst/video/gstvideofilter.h:
      * gst-libs/gst/video/gstvideosink.h:
      * gst/playback/gstplaybasebin.h:
      * gst/tcp/gstmultifdsink.h:
      * sys/v4l/gstv4lelement.h:
      Fix broken GObject macros
      1a2642a1
  10. 24 Mar, 2006 1 commit
    • Stefan Kost's avatar
      Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top · 2d826700
      Stefan Kost authored
      Original commit message from CVS:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-base-plugins-docs.sgml:
      * docs/plugins/gst-plugins-base-plugins-sections.txt:
      * ext/cdparanoia/gstcdparanoiasrc.c:
      * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
      (gst_gnome_vfs_sink_class_init):
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
      * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
      * ext/ogg/gstoggmux.c:
      * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
      * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
      (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
      * ext/pango/gsttextoverlay.c:
      * ext/pango/gsttextrender.c:
      * ext/theora/theoradec.c:
      * ext/theora/theoraenc.c:
      * ext/vorbis/vorbisdec.c:
      * ext/vorbis/vorbisenc.c:
      * gst-libs/gst/audio/gstaudiofilter.c:
      (gst_audio_filter_base_init):
      * gst-libs/gst/audio/gstaudiofiltertemplate.c:
      (gst_audio_filter_template_base_init):
      * gst/adder/gstadder.c: (gst_adder_get_type):
      * gst/adder/gstadder.h:
      * gst/audioconvert/gstaudioconvert.c:
      * gst/audiotestsrc/gstaudiotestsrc.c:
      (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
      (gst_audio_test_src_create):
      * gst/ffmpegcolorspace/gstffmpegcolorspace.c:
      * gst/playback/gstdecodebin.c:
      * gst/playback/gstplaybin.c:
      * gst/playback/gststreamselector.c:
      (gst_stream_selector_base_init):
      * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
      * gst/volume/gstvolume.c:
      * sys/v4l/gstv4lmjpegsink.c:
      * sys/v4l/gstv4lmjpegsrc.c:
      * tests/check/libs/cddabasesrc.c:
      * tests/old/examples/gob/gst-identity2.gob:
      Add docs for adder, use GST_ELEMENT_DETAILS macro,
      define GstElementDetails at the top
      2d826700
  11. 23 Mar, 2006 3 commits
  12. 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
  13. 17 Mar, 2006 1 commit
  14. 07 Mar, 2006 2 commits
    • Tim-Philipp Müller's avatar
      gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push(). · ab6f99ab
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
      Don't ignore flow return from gst_pad_push().
      ab6f99ab
    • Christophe Fergeau's avatar
      Don't leak references returned by gst_pad_get_parent() · 8e6d3a5c
      Christophe Fergeau authored
      Original commit message from CVS:
      * ext/libvisual/visual.c: (gst_visual_getcaps),
      (gst_visual_src_setcaps), (gst_visual_sink_setcaps):
      * ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
      * ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
      (gst_vorbisenc_convert_sink):
      * gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
      (gst_audio_duration_from_pad_buffer):
      * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
      (gst_audio_filter_chain):
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_setcaps):
      * gst-libs/gst/video/video.c: (gst_video_frame_rate),
      (gst_video_get_size):
      * gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
      Don't leak references returned by gst_pad_get_parent()
      (#333663, based on patch by: Christophe Fergeau).
      8e6d3a5c
  15. 02 Mar, 2006 1 commit
    • Wim Taymans's avatar
      docs/: Added some more docs to libs and plugins. · 1e9f5c43
      Wim Taymans authored
      Original commit message from CVS:
      * docs/libs/gst-plugins-base-libs-docs.sgml:
      * docs/libs/gst-plugins-base-libs-sections.txt:
      * docs/libs/gst-plugins-base-libs.types:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-base-plugins-docs.sgml:
      * docs/plugins/gst-plugins-base-plugins-sections.txt:
      Added some more docs to libs and plugins.
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
      * gst-libs/gst/audio/gstringbuffer.h:
      Document ringbuffer some more.
      * gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
      (gst_video_rate_setcaps), (gst_video_rate_reset),
      (gst_video_rate_init), (gst_video_rate_flush_prev),
      (gst_video_rate_swap_prev), (gst_video_rate_event),
      (gst_video_rate_chain), (gst_video_rate_change_state):
      * gst/videorate/gstvideorate.h:
      Fix videorate to use segments.
      Make it work with 0/1 framerates (closes #331903)
      Handle EOS correctly.
      Added docs.
      1e9f5c43
  16. 28 Feb, 2006 1 commit
  17. 20 Feb, 2006 1 commit
  18. 17 Feb, 2006 2 commits
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.c: Small cleanups. · 3451a818
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
      (gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
      (gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
      (gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
      (gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
      (gst_ring_buffer_pause), (gst_ring_buffer_stop),
      (gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
      (gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
      (gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
      (gst_ring_buffer_clear):
      Small cleanups.
      Added some G_LIKELY.
      3451a818
    • 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
  19. 16 Feb, 2006 2 commits
    • Tim-Philipp Müller's avatar
      gst-libs/gst/audio/multichannel.c: Minor docs fix. · 9490d413
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * gst-libs/gst/audio/multichannel.c:
      Minor docs fix.
      * gst-libs/gst/riff/Makefile.am:
      * gst-libs/gst/riff/riff-ids.h:
      * gst-libs/gst/riff/riff-media.c:
      (gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
      Add support for WAVEFORMATEX, eg. PCM audio with more than two
      channels and a channel layout map.
      9490d413
    • Tim-Philipp Müller's avatar
      gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no... · 5b788a8a
      Tim-Philipp Müller authored
      gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
      
      Original commit message from CVS:
      * gst-libs/gst/audio/multichannel.c:
      (gst_audio_get_channel_positions):
      When we have more than 2 channels, but no channel layout is
      specified in the caps, return some default channel layout
      to the caller and warn about about a possibly buggy element
      (could be buggy filtercaps as well of course) (#317038).
      5b788a8a
  20. 14 Feb, 2006 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help. · 3b457402
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/audio/gstringbuffer.c:
      (gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
      (gst_ring_buffer_samples_done), (wait_segment),
      (gst_ring_buffer_commit), (gst_ring_buffer_clear):
      Add some compiler G_(UN_)LIKELY help.
      SIGNAL the ringbuffer waiters when going to PAUSED as well to
      make sure they can exit their functions. Should fix #330748
      3b457402
  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. 09 Feb, 2006 2 commits
    • Andy Wingo's avatar
      kapowpowpow · 4e0c846f
      Andy Wingo authored
      Original commit message from CVS:
      kapowpowpow
      4e0c846f
    • Andy Wingo's avatar
      gst-libs/gst/audio/gstringbuffer.c · 4ae63e73
      Andy Wingo authored
      Original commit message from CVS:
      2006-02-09  Andy Wingo  <wingo@pobox.com>
      
      * gst-libs/gst/audio/gstringbuffer.c
      (gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
      overflow after 13.5 hours of recording. Kapow!
      
      * ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
      the buffer size -- we don't care about underrun/overrun reporting
      right now, just need to return a useful value.
      4ae63e73
  24. 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
  25. 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
  26. 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
  27. 28 Jan, 2006 1 commit
  28. 25 Jan, 2006 1 commit