1. 15 Sep, 2008 2 commits
  2. 13 Sep, 2008 1 commit
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for... · c1647d03
      Wim Taymans authored
      gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...
      
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpbin.c: (create_session),
      (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
      Do not try to adjust the offset of streams for which we have not yet
      seen an SR packet. Avoids large ts-offsets in some cases.
      c1647d03
  3. 10 Sep, 2008 2 commits
    • Michael Smith's avatar
      sys/dshowdecwrapper/: Major rewrite of dshowdecwrapper. Converts code to · 007478f0
      Michael Smith authored
      Original commit message from CVS:
      * sys/dshowdecwrapper/Makefile.am:
      * sys/dshowdecwrapper/gstdshowaudiodec.c:
      * sys/dshowdecwrapper/gstdshowaudiodec.cpp:
      * sys/dshowdecwrapper/gstdshowaudiodec.h:
      * sys/dshowdecwrapper/gstdshowdecwrapper.c:
      * sys/dshowdecwrapper/gstdshowdecwrapper.cpp:
      * sys/dshowdecwrapper/gstdshowdecwrapper.h:
      * sys/dshowdecwrapper/gstdshowfakesrc.cpp:
      * sys/dshowdecwrapper/gstdshowfakesrc.h:
      * sys/dshowdecwrapper/gstdshowutil.cpp:
      * sys/dshowdecwrapper/gstdshowutil.h:
      * sys/dshowdecwrapper/gstdshowvideodec.c:
      * sys/dshowdecwrapper/gstdshowvideodec.cpp:
      * sys/dshowdecwrapper/gstdshowvideodec.h:
      Major rewrite of dshowdecwrapper. Converts code to
      C++, moves to direct use of DirectShow base classes,
      make a lot of code clearer, simplify, etc.
      Fix decode of MP3 on Vista by working around an apparent
      bug in the decoder.
      007478f0
    • Ole Andre Vadla Ravnaas's avatar
      sys/winks/gstksclock.c (gst_ks_clock_worker_thread_func, gst_ks_clock_start): · 61dee512
      Ole Andre Vadla Ravnaas authored
      Original commit message from CVS:
      * sys/winks/gstksclock.c (gst_ks_clock_worker_thread_func,
      gst_ks_clock_start):
      Synchronize KS clock as a single-shot operation for now, there's not
      much point in doing it periodically until we're actually using the
      KS timestamps for anything else than just discarding old frames.
      * sys/winks/gstksvideosrc.c (gst_ks_video_src_open_device):
      Provide the GstClock when opening the device if we already have one.
      61dee512
  4. 09 Sep, 2008 4 commits
    • Ole Andre Vadla Ravnaas's avatar
      sys/winks/gstksvideodevice.c (GST_DEBUG_IS_ENABLED, last_timestamp,... · 0ff4dc30
      Ole Andre Vadla Ravnaas authored
      sys/winks/gstksvideodevice.c (GST_DEBUG_IS_ENABLED, last_timestamp, gst_ks_video_device_prepare_buffers, gst_ks_video...
      
      Original commit message from CVS:
      * sys/winks/gstksvideodevice.c (GST_DEBUG_IS_ENABLED, last_timestamp,
      gst_ks_video_device_prepare_buffers, gst_ks_video_device_create_pin,
      gst_ks_video_device_set_state, gst_ks_video_device_request_frame,
      gst_ks_video_device_read_frame):
      Guard against capturing old frames by keeping track of the last
      timestamp and also zero-fill the buffers before each capture.
      Only assign a master clock if the pin hasn't already got one.
      Actually free buffers on the way down to avoid a huge memory leak,
      as this was previously done when changing state to ACQUIRE downwards
      and we now skip that state on the way down.
      Add some debug.
      * sys/winks/gstksvideosrc.c (DEFAULT_DEVICE_PATH, DEFAULT_DEVICE_NAME,
      DEFAULT_DEVICE_INDEX, KS_WORKER_LOCK, KS_WORKER_UNLOCK,
      KS_WORKER_WAIT, KS_WORKER_NOTIFY, KS_WORKER_WAIT_FOR_RESULT,
      KS_WORKER_NOTIFY_RESULT, KS_WORKER_STATE_STARTING,
      KS_WORKER_STATE_READY, KS_WORKER_STATE_STOPPING,
      KS_WORKER_STATE_ERROR, KsWorkerState, device_path, device_name,
      device_index, running, worker_thread, worker_lock,
      worker_notify_cond, worker_result_cond, worker_state,
      worker_pending_caps, worker_setcaps_result, worker_pending_run,
      worker_run_result, gst_ks_video_src_reset,
      gst_ks_video_src_apply_driver_quirks, gst_ks_video_src_open_device,
      gst_ks_video_src_close_device, gst_ks_video_src_worker_func,
      gst_ks_video_src_start_worker, gst_ks_video_src_stop_worker,
      gst_ks_video_src_change_state, gst_ks_video_src_set_clock,
      gst_ks_video_src_set_caps, gst_ks_video_src_timestamp_buffer,
      gst_ks_video_src_create):
      Remove ENABLE_CLOCK_DEBUG define, it's GST_LEVEL_DEBUG after all.
      Get rid of PROP_ENSLAVE_KSCLOCK and always slave the ks clock to the
      GStreamer clock, it doesn't seem to hurt and matches DirectShow's
      behavior. As an added bonus we usually get PresentationTime set for
      each frame, so we can expand on this later for smarter latency
      reporting (by looking at the diff between the timestamp from the
      driver and the time according to the GStreamer clock).
      Use an internal worker thread for opening the device, setting caps,
      changing its state and closing it. This way we're a lot more
      compatible with drivers that rely on hacks to do video-effects
      between the low-level NT API and the application. Ick.
      Start the ks clock and set the pin to KSSTATE_RUN on the first
      create() so that we'll hopefully get hold of the GStreamer clock
      from the very beginning. This way there's no chance that the
      timestamps will make a sudden jump in the beginning of the stream
      when we're running with a clock.
      * sys/winks/kshelpers.c (CHECK_OPTIONS_FLAG,
      ks_options_flags_to_string):
      Reorder the flags to match the headerfile order, and make the string
      a bit more compact.
      * sys/winks/ksvideohelpers.c (ks_video_probe_filter_for_caps):
      Avoid leaking KSPROPERTY_PIN_DATARANGES.
      0ff4dc30
    • Mark Nauwelaerts's avatar
      Add jp2k plugin. Fixes #550657. · e262a725
      Mark Nauwelaerts authored
      Original commit message from CVS:
      * configure.ac:
      * ext/Makefile.am:
      * ext/jp2k/Makefile.am:
      * ext/jp2k/gstjasperdec.c: (gst_jasper_dec_base_init),
      (gst_jasper_dec_class_init), (gst_jasper_dec_init),
      (gst_jasper_dec_reset), (gst_jasper_dec_sink_setcaps),
      (gst_jasper_dec_negotiate), (gst_jasper_dec_get_picture),
      (gst_jasper_dec_chain), (gst_jasper_dec_set_property),
      (gst_jasper_dec_get_property), (gst_jasper_dec_change_state),
      (plugin_init):
      * ext/jp2k/gstjasperdec.h:
      Add jp2k plugin.  Fixes #550657.
      e262a725
    • Edward Hervey's avatar
      gst/mpegdemux/: Fix conflicting public names in new mpeg demuxers. · 7359989b
      Edward Hervey authored
      Original commit message from CVS:
      * gst/mpegdemux/flumpegdemux.c: (plugin_init):
      * gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_sync_get_type),
      (gst_flups_demux_get_type), (gst_flups_demux_plugin_init):
      * gst/mpegdemux/gstmpegtsdemux.c: (gst_fluts_demux_get_type),
      (gst_fluts_demux_plugin_init):
      Fix conflicting public names in new mpeg demuxers.
      Fixes #550468
      7359989b
    • Michael Smith's avatar
      gst/aiffparse/aiffparse.c: Support chunks in AIFF in any order in pull mode,... · 8618e452
      Michael Smith authored
      gst/aiffparse/aiffparse.c: Support chunks in AIFF in any order in pull mode, and any order so long as we get COMM bef...
      
      Original commit message from CVS:
      * gst/aiffparse/aiffparse.c:
      Support chunks in AIFF in any order in pull mode, and any order so
      long as we get COMM before the actual data (SSND) in push mode.
      Fixes playback of AIFC files.
      8618e452
  5. 08 Sep, 2008 1 commit
  6. 05 Sep, 2008 2 commits
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver. · a35d1dde
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
      (create_session), (gst_rtp_bin_associate),
      (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
      (gst_rtp_bin_request_new_pad):
      * gst/rtpmanager/gstrtpbin.h:
      Add signal to notify listeners when a sender becomes a receiver.
      Tweak lip-sync code, don't store our own copy of the ts-offset of the
      jitterbuffer, don't adjust sync if the change is less than 4msec.
      Get the RTP timestamp <-> GStreamer timestamp relation directly from
      the jitterbuffer instead of our inaccurate version from the source.
      * gst/rtpmanager/gstrtpjitterbuffer.c:
      (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
      (gst_rtp_jitter_buffer_get_sync):
      * gst/rtpmanager/gstrtpjitterbuffer.h:
      Add G_LIKELY macros, use global defines for max packet reorder and
      dropouts.
      Reset the jitterbuffer clock skew detection when packets seqnums are
      changed unexpectedly.
      * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
      (gst_rtp_session_class_init), (gst_rtp_session_init):
      * gst/rtpmanager/gstrtpsession.h:
      Add sender timeout signal.
      * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
      (calculate_skew), (rtp_jitter_buffer_insert),
      (rtp_jitter_buffer_get_sync):
      * gst/rtpmanager/rtpjitterbuffer.h:
      Add some G_LIKELY macros.
      Keep track of the extended RTP timestamp so that we can report the RTP
      timestamp <-> GStreamer timestamp relation for lip-sync.
      Remove server timestamp gap detection code, the server can sometimes
      make a huge gap in timestamps (talk spurts,...) see #549774.
      Detect timetamp weirdness instead by observing the sender/receiver
      timestamp relation and resync if it changes more than 1 second.
      Add method to report about the current rtp <-> gst timestamp relation
      which is needed for lip-sync.
      * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
      (on_sender_timeout), (check_collision), (rtp_session_process_sr),
      (session_cleanup):
      * gst/rtpmanager/rtpsession.h:
      Add sender timeout signal.
      Remove inaccurate rtp <-> gst timestamp relation code, the
      jitterbuffer can now do an accurate reporting about this.
      * gst/rtpmanager/rtpsource.c: (rtp_source_init),
      (rtp_source_update_caps), (calculate_jitter),
      (rtp_source_process_rtp):
      * gst/rtpmanager/rtpsource.h:
      Remove inaccurate rtp <-> gst timestamp relation code.
      * gst/rtpmanager/rtpstats.h:
      Define global max-reorder and max-dropout constants for use in various
      subsystems.
      a35d1dde
    • Sebastian Pölsterl's avatar
      sys/dvb/gstdvbsrc.c: Add DVB Adapter name to structure sent over bus. · 64cd01e7
      Sebastian Pölsterl authored and Zaheer Abbas Merali's avatar Zaheer Abbas Merali committed
      Original commit message from CVS:
      patch by: Sebastian Pölsterl
      * sys/dvb/gstdvbsrc.c:
      Add DVB Adapter name to structure sent over bus.
      64cd01e7
  7. 03 Sep, 2008 1 commit
  8. 02 Sep, 2008 5 commits
    • Edward Hervey's avatar
      gst/mpegdemux/: Fix build on macosx. · 104ca25c
      Edward Hervey authored
      Original commit message from CVS:
      * gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_parse_pack_start):
      * gst/mpegdemux/gstmpegtsdemux.c: (gst_fluts_demux_data_cb):
      Fix build on macosx.
      104ca25c
    • Zaheer Abbas Merali's avatar
      Add Fluendo MPEG PS and TS demuxers to gst-plugins-bad. This is now dual licensed MPL and LGPL. · 555486f8
      Zaheer Abbas Merali authored
      Original commit message from CVS:
      * configure.ac:
      * gst/mpegdemux/Makefile.am:
      * gst/mpegdemux/flumpegdemux.c:
      * gst/mpegdemux/flutspatinfo.c:
      * gst/mpegdemux/flutspatinfo.h:
      * gst/mpegdemux/flutspmtinfo.c:
      * gst/mpegdemux/flutspmtinfo.h:
      * gst/mpegdemux/flutspmtstreaminfo.c:
      * gst/mpegdemux/flutspmtstreaminfo.h:
      * gst/mpegdemux/gstmpegdefs.h:
      * gst/mpegdemux/gstmpegdemux.c:
      * gst/mpegdemux/gstmpegdemux.h:
      * gst/mpegdemux/gstmpegdesc.c:
      * gst/mpegdemux/gstmpegdesc.h:
      * gst/mpegdemux/gstmpegtsdemux.c:
      * gst/mpegdemux/gstmpegtsdemux.h:
      * gst/mpegdemux/gstpesfilter.c:
      * gst/mpegdemux/gstpesfilter.h:
      * gst/mpegdemux/gstsectionfilter.c:
      * gst/mpegdemux/gstsectionfilter.h:
      Add Fluendo MPEG PS and TS demuxers to gst-plugins-bad. This
      is now dual licensed MPL and LGPL.
      555486f8
    • Wim Taymans's avatar
      gst/mpegtsmux/mpegtsmux.c: Set caps on outgoing buffers. · 0ce15bab
      Wim Taymans authored
      Original commit message from CVS:
      * gst/mpegtsmux/mpegtsmux.c: (new_packet_cb):
      Set caps on outgoing buffers.
      0ce15bab
    • Tim-Philipp Müller's avatar
      Enable/fix up translations for these plugins. · b7276b6f
      Tim-Philipp Müller authored
      Original commit message from CVS:
      * ext/resindvd/plugin.c: (plugin_init):
      * ext/resindvd/resindvdsrc.c:
      * ext/twolame/gsttwolame.c: (plugin_init):
      * gst/aiffparse/aiffparse.c: (plugin_init):
      Enable/fix up translations for these plugins.
      * po/LINGUAS:
      Add 'ca' to LINGUAS.
      * po/POTFILES.in:
      * po/POTFILES.skip:
      Add more files for translation and more files which tools
      should skip.
      b7276b6f
    • Edward Hervey's avatar
      gst/mpegtsmux/tsmux/tsmux.c: Fix build on macosx. · d5e8bc14
      Edward Hervey authored
      Original commit message from CVS:
      * gst/mpegtsmux/tsmux/tsmux.c: (tsmux_write_ts_header):
      Fix build on macosx.
      d5e8bc14
  9. 01 Sep, 2008 7 commits
    • Christian Schaller's avatar
      update spec file and add missing subdirs in Makefile.am · c35f3328
      Christian Schaller authored
      Original commit message from CVS:
      update spec file and add missing subdirs in Makefile.am
      c35f3328
    • Sebastian Dröge's avatar
      gst/mpegtsmux/mpegtsmux_aac.c: Allocate a fixed size buffer on the stack instead of using malloc(). · 7ac07782
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/mpegtsmux/mpegtsmux_aac.c: (mpegtsmux_prepare_aac):
      Allocate a fixed size buffer on the stack instead of using malloc().
      * gst/mpegtsmux/tsmux/tsmux.c: (tsmux_new), (tsmux_free),
      (tsmux_program_new), (tsmux_program_free):
      * gst/mpegtsmux/tsmux/tsmuxstream.c: (tsmux_stream_new),
      (tsmux_stream_free), (tsmux_stream_consume),
      (tsmux_stream_add_data):
      Use GSlice.
      7ac07782
    • Sebastian Dröge's avatar
      gst/mpegtsmux/mpegtsmux.c: Add support for muxing MPEG4 video. · b24d3831
      Sebastian Dröge authored
      Original commit message from CVS:
      * gst/mpegtsmux/mpegtsmux.c: (mpegtsmux_create_stream):
      Add support for muxing MPEG4 video.
      b24d3831
    • Edward Hervey's avatar
      gst/mpegtsmux/tsmux/: Fix build of mpegtsmux. · b76d9f5b
      Edward Hervey authored
      Original commit message from CVS:
      * gst/mpegtsmux/tsmux/tsmux.h:
      * gst/mpegtsmux/tsmux/tsmuxstream.h:
      Fix build of mpegtsmux.
      b76d9f5b
    • Sebastian Dröge's avatar
      Add Fluendo MPEG-TS muxer and libtsmux to gst-plugins-bad. This is renamed to... · 845094c3
      Sebastian Dröge authored
      Add Fluendo MPEG-TS muxer and libtsmux to gst-plugins-bad. This is renamed to mpegtsmux to prevent conflicts. Also al...
      
      Original commit message from CVS:
      * configure.ac:
      * gst/mpegtsmux/Makefile.am:
      * gst/mpegtsmux/mpegtsmux.c: (mpegtsmux_base_init),
      (mpegtsmux_class_init), (mpegtsmux_init), (mpegtsmux_dispose),
      (gst_mpegtsmux_set_property), (gst_mpegtsmux_get_property),
      (release_buffer_cb), (mpegtsmux_create_stream),
      (mpegtsmux_create_streams), (mpegtsmux_choose_best_stream),
      (mpegtsmux_collected), (mpegtsmux_request_new_pad),
      (mpegtsmux_release_pad), (new_packet_cb),
      (mpegtsdemux_prepare_srcpad), (mpegtsmux_change_state),
      (plugin_init):
      * gst/mpegtsmux/mpegtsmux.h:
      * gst/mpegtsmux/mpegtsmux_aac.c: (mpegtsmux_prepare_aac):
      * gst/mpegtsmux/mpegtsmux_aac.h:
      * gst/mpegtsmux/mpegtsmux_h264.c: (mpegtsmux_prepare_h264):
      * gst/mpegtsmux/mpegtsmux_h264.h:
      * gst/mpegtsmux/tsmux/Makefile.am:
      * gst/mpegtsmux/tsmux/crc.h:
      * gst/mpegtsmux/tsmux/tsmux.c: (tsmux_new), (tsmux_set_write_func),
      (tsmux_set_pat_frequency), (tsmux_get_pat_frequency), (tsmux_free),
      (tsmux_program_new), (tsmux_set_pmt_frequency),
      (tsmux_get_pmt_frequency), (tsmux_program_add_stream),
      (tsmux_program_set_pcr_stream), (tsmux_get_new_pid),
      (tsmux_create_stream), (tsmux_find_stream), (tsmux_packet_out),
      (tsmux_write_adaptation_field), (tsmux_write_ts_header),
      (tsmux_write_stream_packet), (tsmux_program_free),
      (tsmux_write_section), (tsmux_write_section_hdr),
      (tsmux_write_pat), (tsmux_write_pmt):
      * gst/mpegtsmux/tsmux/tsmux.h:
      * gst/mpegtsmux/tsmux/tsmuxcommon.h:
      * gst/mpegtsmux/tsmux/tsmuxstream.c: (tsmux_stream_new),
      (tsmux_stream_get_pid), (tsmux_stream_free),
      (tsmux_stream_set_buffer_release_func), (tsmux_stream_consume),
      (tsmux_stream_at_pes_start), (tsmux_stream_bytes_avail),
      (tsmux_stream_bytes_in_buffer), (tsmux_stream_get_data),
      (tsmux_stream_pes_header_length),
      (tsmux_stream_find_pts_dts_within),
      (tsmux_stream_write_pes_header), (tsmux_stream_add_data),
      (tsmux_stream_get_es_descrs), (tsmux_stream_pcr_ref),
      (tsmux_stream_pcr_unref), (tsmux_stream_is_pcr),
      (tsmux_stream_get_pts):
      * gst/mpegtsmux/tsmux/tsmuxstream.h:
      Add Fluendo MPEG-TS muxer and libtsmux to gst-plugins-bad. This
      is renamed to mpegtsmux to prevent conflicts. Also all relevant
      informations about copyright and license are added to the top of
      every file but apart from that no changes compared to the latest
      SVN versions happened.
      845094c3
    • Edward Hervey's avatar
      tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ...... · 122498e1
      Edward Hervey authored
      tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
      
      Original commit message from CVS:
      * tests/check/elements/audioresample.c: (setup_audioresample),
      (fail_unless_perfect_stream), (test_perfect_stream_instance),
      (test_discont_stream_instance):
      Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
      Add debugging for coherence.
      122498e1
    • Wim Taymans's avatar
      gst/selector/gstinputselector.c: Reuse the get_linked_pads for both source and... · 7d9e7b68
      Wim Taymans authored
      gst/selector/gstinputselector.c: Reuse the get_linked_pads for both source and sinkpads because they are the same.
      
      Original commit message from CVS:
      * gst/selector/gstinputselector.c: (gst_input_selector_init),
      (gst_input_selector_event), (gst_input_selector_query):
      Reuse the get_linked_pads for both source and sinkpads because they are
      the same.
      Implement a custum event handler and get the internally linked pad
      directly instead of relying on the default (slower) implementation.
      7d9e7b68
  10. 31 Aug, 2008 2 commits
    • Sebastian Dröge's avatar
      ext/celt/gstceltdec.c: Correctly take the granulepos from upstream if possible... · 16e70c80
      Sebastian Dröge authored
      ext/celt/gstceltdec.c: Correctly take the granulepos from upstream if possible and correctly handle the granulepos in...
      
      Original commit message from CVS:
      * ext/celt/gstceltdec.c: (celt_dec_chain_parse_data):
      Correctly take the granulepos from upstream if possible and
      correctly handle the granulepos in various calculations: the
      granulepos is the sample number of the _last_ sample in a frame, not
      the first.
      * ext/celt/gstceltenc.c: (gst_celt_enc_sinkevent),
      (gst_celt_enc_encode), (gst_celt_enc_chain),
      (gst_celt_enc_change_state):
      * ext/celt/gstceltenc.h:
      Handle non-zero start timestamps in the encoder and detect/handle
      stream discontinuities. Fixes bug #547075.
      16e70c80
    • Rov Juvano's avatar
      Add scaletempo plugin, which allows to scale the speed of audio without... · 315cb1ab
      Rov Juvano authored and Sebastian Dröge's avatar Sebastian Dröge committed
      Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r...
      
      Original commit message from CVS:
      Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net>
      * configure.ac:
      * docs/plugins/Makefile.am:
      * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
      * docs/plugins/gst-plugins-bad-plugins-sections.txt:
      * docs/plugins/inspect/plugin-scaletempo.xml:
      * examples/scaletempo/Makefile.am:
      * examples/scaletempo/demo-gui.c: (pop_status_bar),
      (status_bar_printf), (demo_gui_seek_bar_format), (update_position),
      (demo_gui_seek_bar_change), (demo_gui_do_change_rate),
      (demo_gui_do_set_rate), (demo_gui_do_rate_entered),
      (demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled),
      (demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause),
      (demo_gui_do_play_pause), (demo_gui_do_open_file),
      (demo_gui_do_playlist_prev), (demo_gui_do_playlist_next),
      (demo_gui_do_about_dialog), (demo_gui_do_quit),
      (demo_gui_request_set_stride), (demo_gui_request_set_overlap),
      (demo_gui_request_set_search), (demo_gui_rate_changed),
      (demo_gui_playing_started), (demo_gui_playing_paused),
      (demo_gui_playing_ended), (demo_gui_player_errored),
      (demo_gui_stride_changed), (demo_gui_overlap_changed),
      (demo_gui_search_changed), (demo_gui_set_player_func),
      (demo_gui_set_playlist_func), (build_gvalue_array),
      (create_action), (demo_gui_show_func), (demo_gui_set_player),
      (demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property),
      (demo_gui_set_property), (demo_gui_init), (demo_gui_class_init),
      (demo_gui_get_type):
      * examples/scaletempo/demo-gui.h:
      * examples/scaletempo/demo-main.c: (handle_error_message),
      (handle_quit), (main):
      * examples/scaletempo/demo-player.c: (no_pipeline),
      (demo_player_event_listener), (demo_player_state_changed_cb),
      (demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate),
      (demo_player_scale_rate_func), (demo_player_set_rate_func),
      (_set_state_and_wait), (demo_player_load_uri_func),
      (demo_player_play_func), (demo_player_pause_func), (_seek_to),
      (demo_player_seek_by_func), (demo_player_seek_to_func),
      (demo_player_get_position_func), (demo_player_get_duration_func),
      (demo_player_scale_rate), (demo_player_set_rate),
      (demo_player_load_uri), (demo_player_play), (demo_player_pause),
      (demo_player_seek_by), (demo_player_seek_to),
      (demo_player_get_position), (demo_player_get_duration),
      (demo_player_get_property), (demo_player_set_property),
      (demo_player_init), (demo_player_class_init),
      (demo_player_get_type):
      * examples/scaletempo/demo-player.h:
      * gst/scaletempo/Makefile.am:
      * gst/scaletempo/gstscaletempo.c: (best_overlap_offset_float),
      (best_overlap_offset_s16), (output_overlap_float),
      (output_overlap_s16), (fill_queue), (reinit_buffers),
      (gst_scaletempo_transform), (gst_scaletempo_transform_size),
      (gst_scaletempo_sink_event), (gst_scaletempo_set_caps),
      (gst_scaletempo_get_property), (gst_scaletempo_set_property),
      (gst_scaletempo_base_init), (gst_scaletempo_class_init),
      (gst_scaletempo_init):
      * gst/scaletempo/gstscaletempo.h:
      * gst/scaletempo/gstscaletempoplugin.c: (plugin_init):
      Add scaletempo plugin, which allows to scale the speed of audio without
      changing the pitch by handling seeks with a rate!=1.0.
      Integrate it into the docs and add the example application for it.
      Fixes bug #537700.
      315cb1ab
  11. 30 Aug, 2008 2 commits
  12. 29 Aug, 2008 5 commits
  13. 28 Aug, 2008 6 commits
    • Ole Andre Vadla Ravnaas's avatar
      sys/winks/ksvideohelpers.c (ks_video_media_type_free): Avoid leaking the... · e3fcb1d8
      Ole Andre Vadla Ravnaas authored
      sys/winks/ksvideohelpers.c (ks_video_media_type_free): Avoid leaking the KSDATARANGE member of each KsVideoMediaType.
      
      Original commit message from CVS:
      * sys/winks/ksvideohelpers.c (ks_video_media_type_free):
      Avoid leaking the KSDATARANGE member of each KsVideoMediaType.
      e3fcb1d8
    • Jan Schmidt's avatar
      gst/dccp/: Fix compilation on Solaris by including filio.h as needed. · 9a721982
      Jan Schmidt authored
      Original commit message from CVS:
      * gst/dccp/gstdccp.c:
      * gst/dccp/gstdccpclientsrc.c:
      Fix compilation on Solaris by including filio.h as needed.
      * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
      * gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
      Fix compilation with Forte - apparently it hates concatenating a
      macro argument that starts with an underscore??
      9a721982
    • Jérémie Bernard's avatar
      Add apexsink for audio output to Apple AirPort Express Wireless devices. Fixes bug #542510. · a72dc699
      Jérémie Bernard authored and Sebastian Dröge's avatar Sebastian Dröge committed
      Original commit message from CVS:
      Patch by: Jérémie Bernard <gremimail at gmail dot com>
      * configure.ac:
      * ext/apexsink/LGPL-3.0.txt:
      * ext/apexsink/Makefile.am:
      * ext/apexsink/gstapexplugin.c: (plugin_init):
      * ext/apexsink/gstapexraop.c: (g_strdel), (gst_apexraop_send),
      (gst_apexraop_recv), (gst_apexraop_new), (gst_apexraop_free),
      (gst_apexraop_set_host), (gst_apexraop_get_host),
      (gst_apexraop_set_port), (gst_apexraop_get_port),
      (gst_apexraop_set_useragent), (gst_apexraop_get_useragent),
      (gst_apexraop_connect), (gst_apexraop_get_jacktype),
      (gst_apexraop_get_jackstatus), (gst_apexraop_close),
      (gst_apexraop_set_volume), (gst_apexraop_write_bits),
      (gst_apexraop_write), (gst_apexraop_flush):
      * ext/apexsink/gstapexraop.h:
      * ext/apexsink/gstapexsink.c: (gst_apexsink_jackstatus_get_type),
      (gst_apexsink_jacktype_get_type), (gst_apexsink_interfaces_init),
      (gst_apexsink_implements_interface_init),
      (gst_apexsink_mixer_interface_init),
      (gst_apexsink_interface_supported),
      (gst_apexsink_mixer_list_tracks), (gst_apexsink_mixer_set_volume),
      (gst_apexsink_mixer_get_volume), (gst_apexsink_base_init),
      (gst_apexsink_class_init), (gst_apexsink_init),
      (gst_apexsink_set_property), (gst_apexsink_get_property),
      (gst_apexsink_finalise), (gst_apexsink_open),
      (gst_apexsink_prepare), (gst_apexsink_write),
      (gst_apexsink_unprepare), (gst_apexsink_delay),
      (gst_apexsink_reset), (gst_apexsink_close):
      * ext/apexsink/gstapexsink.h:
      Add apexsink for audio output to Apple AirPort Express Wireless
      devices. Fixes bug #542510.
      a72dc699
    • Wim Taymans's avatar
      gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to. · 53025584
      Wim Taymans authored
      Original commit message from CVS:
      * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
      (gst_rtp_session_event_send_rtp_sink):
      Send EOS when the session object instructs us to.
      * gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
      * gst/rtpmanager/rtpsession.h:
      Make it possible for the session manager to instruct us to send EOS. We
      currently will EOS when the session is a sender and when the sender part
      goes EOS. This is not entirely correct behaviour because the session
      could still participate as a receiver.
      Fixes #549409.
      53025584
    • Michael Smith's avatar
      gst/aiffparse/aiffparse.c: Read size of chunks preceeding the audio data with... · 23dbeaab
      Michael Smith authored
      gst/aiffparse/aiffparse.c: Read size of chunks preceeding the audio data with the correct endianness. Fixes playback ...
      
      Original commit message from CVS:
      * gst/aiffparse/aiffparse.c:
      Read size of chunks preceeding the audio data with the
      correct endianness. Fixes playback of some files.
      Fixes #538500
      23dbeaab
    • Michael Smith's avatar
      Add an AIFF parsing element, heavily based on wavparse. · a8661bc4
      Michael Smith authored
      Original commit message from CVS:
      * configure.ac:
      * gst/aiffparse/Makefile.am:
      * gst/aiffparse/aiffparse.c:
      * gst/aiffparse/aiffparse.h:
      Add an AIFF parsing element, heavily based on wavparse.
      a8661bc4