1. 23 May, 2011 1 commit
  2. 08 Apr, 2011 2 commits
  3. 21 Dec, 2010 2 commits
  4. 06 Sep, 2010 1 commit
  5. 09 Apr, 2010 1 commit
  6. 19 Mar, 2010 1 commit
  7. 21 Jan, 2010 2 commits
  8. 19 Jan, 2010 1 commit
  9. 05 Jan, 2010 2 commits
  10. 16 Oct, 2009 2 commits
  11. 10 Sep, 2009 1 commit
  12. 03 Sep, 2009 3 commits
  13. 01 Sep, 2009 1 commit
  14. 19 Jun, 2009 1 commit
  15. 11 Jun, 2009 1 commit
  16. 12 May, 2009 1 commit
  17. 13 Oct, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY · 4ae82906
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
      (gst_base_rtp_depayload_set_gst_timestamp),
      (gst_base_rtp_depayload_change_state):
      * gst-libs/gst/rtp/gstbasertpdepayload.h:
      Add some more G_LIKELY
      Fail when the setcaps function was not called.
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_set_outcaps):
      Propagate return value of setcaps.
      4ae82906
  18. 06 Oct, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp... · a2eb0536
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_change_state):
      Configure the next seqnum and timestamp in the state change so that they
      can be queried soon after.
      a2eb0536
  19. 30 May, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into... · 11309247
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
      (gst_basertppayload_change_state):
      Simply converting the running time into an RTP timestamp by scaling it
      based on the clock-rate is good enough for making an RTP timestamp. This
      has the added benefit that we can later on expose a property with the
      RTP timestamp of running time 0, as is needed for RTSP servers to
      generate the response of the PLAY request.
      11309247
  20. 02 May, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function... · c6389eec
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
      (gst_basertppayload_sink_setcaps),
      (gst_basertppayload_sink_getcaps):
      Rename the setcaps/getcaps function internally to make it clear that
      they are called for the sink pad.
      c6389eec
  21. 22 Mar, 2008 1 commit
    • Sebastian Dröge's avatar
      Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings... · 49deb0c0
      Sebastian Dröge authored
      Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
      
      Original commit message from CVS:
      * configure.ac:
      * ext/alsa/gstalsamixerelement.c:
      (gst_alsa_mixer_element_class_init):
      * ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
      * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
      * ext/cdparanoia/gstcdparanoiasrc.c:
      (gst_cd_paranoia_src_class_init):
      * ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
      * ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
      * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
      * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
      * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
      * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
      * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
      * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
      * ext/pango/gsttextrender.c: (gst_text_render_class_init):
      * ext/theora/theoradec.c: (gst_theora_dec_class_init):
      * ext/theora/theoraenc.c: (gst_theora_enc_class_init):
      * ext/theora/theoraparse.c: (gst_theora_parse_class_init):
      * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
      * gst-libs/gst/audio/gstaudiofiltertemplate.c:
      (gst_audio_filter_template_class_init):
      * gst-libs/gst/audio/gstbaseaudiosink.c:
      (gst_base_audio_sink_class_init):
      * gst-libs/gst/audio/gstbaseaudiosrc.c:
      (gst_base_audio_src_class_init):
      * gst-libs/gst/cdda/gstcddabasesrc.c:
      (gst_cdda_base_src_class_init):
      * gst-libs/gst/interfaces/mixertrack.c:
      (gst_mixer_track_class_init):
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_class_init):
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_class_init):
      * gst/audioconvert/gstaudioconvert.c:
      (gst_audio_convert_class_init):
      * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
      * gst/audioresample/gstaudioresample.c:
      (gst_audioresample_class_init):
      * gst/audiotestsrc/gstaudiotestsrc.c:
      (gst_audio_test_src_class_init):
      * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
      * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
      * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
      (preroll_unlinked):
      * gst/playback/gstplaybin.c: (gst_play_bin_class_init):
      * gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
      * gst/playback/gstplaysink.c: (gst_play_sink_class_init):
      * gst/playback/gstqueue2.c: (gst_queue_class_init):
      * gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
      * gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
      (gst_stream_selector_class_init):
      * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
      * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
      * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
      * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
      * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
      * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
      * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
      * gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
      * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
      * gst/videotestsrc/gstvideotestsrc.c:
      (gst_video_test_src_class_init):
      * gst/volume/gstvolume.c: (gst_volume_class_init):
      * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
      * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
      * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
      * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
      * sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
      Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
      static strings (i.e. all). This gives us less memory usage,
      fewer allocations and thus less memory defragmentation. Depend
      on core CVS for this. Fixes bug #523806.
      49deb0c0
  22. 03 Mar, 2008 1 commit
    • Sebastian Dröge's avatar
      Correct all relevant warnings found by the sparse semantic code analyzer. This... · ec7afb6f
      Sebastian Dröge authored
      Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
      
      Original commit message from CVS:
      * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
      * ext/alsa/gstalsasink.c: (set_hwparams):
      * ext/alsa/gstalsasrc.c: (set_hwparams):
      * ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
      * ext/ogg/gstoggmux.h:
      * ext/ogg/gstogmparse.c:
      * gst-libs/gst/audio/audio.c:
      * gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
      * gst-libs/gst/pbutils/missing-plugins.c:
      (gst_missing_uri_sink_message_new),
      (gst_missing_element_message_new),
      (gst_missing_decoder_message_new),
      (gst_missing_encoder_message_new):
      * gst-libs/gst/rtp/gstbasertppayload.c:
      * gst-libs/gst/rtp/gstrtcpbuffer.c:
      (gst_rtcp_packet_bye_get_reason):
      * gst/audioconvert/gstaudioconvert.c:
      * gst/audioresample/gstaudioresample.c:
      * gst/ffmpegcolorspace/imgconvert.c:
      * gst/playback/test.c: (gen_video_element), (gen_audio_element):
      * gst/typefind/gsttypefindfunctions.c:
      * gst/videoscale/vs_4tap.c:
      * gst/videoscale/vs_4tap.h:
      * sys/v4l/gstv4lelement.c:
      * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
      * sys/v4l/v4l_calls.c:
      * sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
      (gst_v4lsrc_try_capture):
      * sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
      (gst_ximagesink_ximage_new):
      * sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
      (gst_xvimagesink_xvimage_new):
      * tests/check/elements/audioconvert.c:
      * tests/check/elements/audioresample.c:
      (fail_unless_perfect_stream):
      * tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
      * tests/check/elements/decodebin.c:
      * tests/check/elements/gdpdepay.c: (setup_gdpdepay),
      (setup_gdpdepay_streamheader):
      * tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
      (setup_gdppay_streamheader):
      * tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
      * tests/check/elements/multifdsink.c: (setup_multifdsink):
      * tests/check/elements/textoverlay.c:
      * tests/check/elements/videorate.c: (setup_videorate):
      * tests/check/elements/videotestsrc.c: (setup_videotestsrc):
      * tests/check/elements/volume.c: (setup_volume):
      * tests/check/elements/vorbisdec.c: (setup_vorbisdec):
      * tests/check/elements/vorbistag.c:
      * tests/check/generic/clock-selection.c:
      * tests/check/generic/states.c: (setup), (teardown):
      * tests/check/libs/cddabasesrc.c:
      * tests/check/libs/video.c:
      * tests/check/pipelines/gio.c:
      * tests/check/pipelines/oggmux.c:
      * tests/check/pipelines/simple-launch-lines.c:
      (simple_launch_lines_suite):
      * tests/check/pipelines/streamheader.c:
      * tests/check/pipelines/theoraenc.c:
      * tests/check/pipelines/vorbisdec.c:
      * tests/check/pipelines/vorbisenc.c:
      * tests/examples/seek/scrubby.c:
      * tests/examples/seek/seek.c: (query_positions_elems),
      (query_positions_pads):
      * tests/icles/stress-xoverlay.c: (myclock):
      Correct all relevant warnings found by the sparse semantic code
      analyzer. This include marking several symbols static, using
      NULL instead of 0 for pointers and using "foo (void)" instead
      of "foo ()" for declarations.
      * win32/common/libgstrtp.def:
      Add gst_rtp_buffer_set_extension_data to the symbol definition file.
      ec7afb6f
  23. 12 Feb, 2008 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks. · d8c28a99
      Wim Taymans authored
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_audio_payload_handle_frame_based_buffer),
      (gst_base_rtp_audio_payload_handle_sample_based_buffer):
      Fix potential leaks.
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
      Fix leak when there is no function configured.
      d8c28a99
  24. 03 Dec, 2007 1 commit
  25. 09 Oct, 2007 1 commit
  26. 19 Sep, 2007 1 commit
  27. 16 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST... · 523fd097
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
      (gst_base_rtp_depayload_chain),
      (gst_base_rtp_depayload_handle_sink_event),
      (gst_base_rtp_depayload_push_full),
      (gst_base_rtp_depayload_set_gst_timestamp),
      (gst_base_rtp_depayload_change_state):
      Remove code to deal with RTP to GST time conversion, we now just copy
      the GST timestamp we receive to the outgoing buffers.
      Handle segment and flushes correctly.
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
      When we have no valid input timestamp, use the previous rtp timestamp on
      the outgoing RTP packet instead of the RTP base time.
      523fd097
  28. 15 Sep, 2007 1 commit
  29. 14 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp... · 06ded625
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
      (gst_basertppayload_set_outcaps), (gst_basertppayload_push),
      (gst_basertppayload_change_state):
      Make sure we start our RTP timestamp from the random base RTP
      timestamp even if the buffer timestamp starts from some random value.
      06ded625
  30. 04 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler... · 56e39e7c
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpaudiopayload.c:
      (gst_base_rtp_payload_audio_handle_event):
      Return FALSE from the event handler to let the parent class handle the
      event.
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
      Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
      * gst-libs/gst/rtp/gstbasertppayload.c:
      Bump the MTU to 1400.
      56e39e7c
  31. 03 Sep, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay... · 0cfb3152
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertpdepayload.c:
      (gst_base_rtp_depayload_class_init),
      (gst_base_rtp_depayload_set_gst_timestamp):
      Add some more docs for the queue-delay property and fix a typo in a
      comment.
      * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
      Fix typo.
      0cfb3152
  32. 16 Aug, 2007 1 commit
  33. 12 Aug, 2007 1 commit
    • Wim Taymans's avatar
      gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that... · 3b7071a1
      Wim Taymans authored
      gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
      
      Original commit message from CVS:
      * gst-libs/gst/rtp/gstbasertppayload.c:
      (gst_basertppayload_set_outcaps):
      * gst-libs/gst/rtp/gstbasertppayload.h:
      Improve caps negotiation so that downstream elements can confiure
      certain RTP properties by fixing them on the caps. See #465146.
      Add docs.
      3b7071a1