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2007-10-01Added SDP demuxer element. Fixes #426657.Wim Taymans1-0/+23
Original commit message from CVS: * configure.ac: * gst/sdp/gstsdpdemux.c: (_do_init), (gst_sdp_demux_base_init), (gst_sdp_demux_class_init), (gst_sdp_demux_init), (gst_sdp_demux_finalize), (gst_sdp_demux_set_property), (gst_sdp_demux_get_property), (find_stream_by_id), (find_stream_by_pt), (find_stream_by_udpsrc), (find_stream), (gst_sdp_demux_stream_free), (gst_sdp_demux_create_stream), (gst_sdp_demux_cleanup), (get_default_rate_for_pt), (gst_sdp_demux_parse_rtpmap), (gst_sdp_demux_media_to_caps), (new_session_pad), (request_pt_map), (gst_sdp_demux_do_stream_eos), (on_bye_ssrc), (on_timeout), (gst_sdp_demux_configure_manager), (gst_sdp_demux_stream_configure_udp), (gst_sdp_demux_stream_configure_udp_sink), (gst_sdp_demux_combine_flows), (gst_sdp_demux_stream_push_event), (gst_sdp_demux_handle_message), (gst_sdp_demux_start), (gst_sdp_demux_sink_event), (gst_sdp_demux_sink_chain), (gst_sdp_demux_change_state): * gst/sdp/gstsdpdemux.h: * gst/sdp/gstsdpelem.c: (plugin_init): Added SDP demuxer element. Fixes #426657.
2007-10-01gst/mpegtsparse/: Remove useless src pad that only results in not linked ↵mutex at runbox dot com1-0/+14
errors, fix a broken pointer dereference and... Original commit message from CVS: Patch by: mutex at runbox dot com * gst/mpegtsparse/mpegtspacketizer.c: (mpegts_packetizer_parse_adaptation_field_control): * gst/mpegtsparse/mpegtsparse.c: (mpegts_parse_base_init), (mpegts_parse_init), (mpegts_parse_push): * gst/mpegtsparse/mpegtsparse.h: Remove useless src pad that only results in not linked errors, fix a broken pointer dereference and make MAX_CONTINUITY constant conform to the standard to stop outputting corrupted data. Fixes #481276, #481279.
2007-09-29ext/mythtv/gstmythtvsrc.c: Re-apply docs patch from #468039; fix tab.Tim-Philipp Müller1-0/+8
Original commit message from CVS: * ext/mythtv/gstmythtvsrc.c: Re-apply docs patch from #468039; fix tab. * gst/mpegtsparse/.cvsignore: Ignore marshaller files generated at build time.
2007-09-29ext/gio/: Some minor cleanup and allow setting the location only when the ↵Sebastian Dröge1-0/+9
element is not playing or paused. Original commit message from CVS: * ext/gio/gstgiosink.c: (gst_gio_sink_base_init), (gst_gio_sink_set_property), (gst_gio_sink_render): * ext/gio/gstgiosrc.c: (gst_gio_src_base_init), (gst_gio_src_set_property): Some minor cleanup and allow setting the location only when the element is not playing or paused.
2007-09-29configure.ac: Update gio's pkg-config file name as currently in SVN.Sebastian Dröge1-0/+9
Original commit message from CVS: * configure.ac: Update gio's pkg-config file name as currently in SVN. * ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_set_location): Remove special casing for a NULL query string. g_strjoin won't add the separator if there's only one string.
2007-09-28gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose.Wim Taymans1-0/+10
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (free_client): Fix crasher in dispose. * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): Handle cases where input buffers have no timestamps so that no clock skew can be calculated, in this case interpollate timestamps based on rtp timestamp and assume a 0 clock skew.
2007-09-28gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now ↵Wim Taymans1-0/+18
in the lower level object. Original commit message from CVS: * gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query): Remove jitter correction code, it's now in the lower level object. Use new -core method for doing a peer query. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (calculate_skew), (rtp_jitter_buffer_insert): * gst/rtpmanager/rtpjitterbuffer.h: Move jitter correction to the lowlevel jitterbuffer. Increase the max window size. When filling the window, already start estimating the skew using a parabolic weighting factor so that we have a much better startup behaviour that gets more accurate with the more samples we have. Increase the default weighting factor for the steady state to get smoother timestamps.
2007-09-28ext/neon/gstneonhttpsrc.c: Now that we require libneon >= 0.26 remove the ↵Sebastian Dröge1-0/+7
neon 0.25 backward compatibility stuff. Als... Original commit message from CVS: * ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_set_proxy), (gst_neonhttp_src_send_request_and_redirect): Now that we require libneon >= 0.26 remove the neon 0.25 backward compatibility stuff. Also fix the default location.
2007-09-28configure.ac: We require libneon >= 26 now for the query field in ne_uri.Wim Taymans1-0/+5
Original commit message from CVS: * configure.ac: We require libneon >= 26 now for the query field in ne_uri.
2007-09-28ext/neon/gstneonhttpsrc.*: Don't discard GET parameters from URL if existing.Wouter Cloetens1-0/+11
Original commit message from CVS: Patch by: Wouter Cloetens <wouter@mind.be> * ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_dispose), (gst_neonhttp_src_set_location), (gst_neonhttp_src_send_request_and_redirect): * ext/neon/gstneonhttpsrc.h: Don't discard GET parameters from URL if existing. Fixes #481200.
2007-09-27gst/librfb/gstrfbsrc.*: Added a property for incremental screen updatesThijs Vermeir1-0/+6
Original commit message from CVS: * gst/librfb/gstrfbsrc.c: * gst/librfb/gstrfbsrc.h: Added a property for incremental screen updates
2007-09-27ext/xvid/gstxvidenc.*: Remove superfluous 'frame-encoded' signal (people can ↵Tim-Philipp Müller1-0/+8
use an upstream identity's 'handoff' sig... Original commit message from CVS: * ext/xvid/gstxvidenc.c: * ext/xvid/gstxvidenc.h: Remove superfluous 'frame-encoded' signal (people can use an upstream identity's 'handoff' signal or a pad probe for this if they must know).
2007-09-27gst/flv/gstflvparse.c: I got it wrong again, audio rate was not detected ↵Julien Moutte1-0/+6
correctly in all cases. Original commit message from CVS: 2007-09-27 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): I got it wrong again, audio rate was not detected correctly in all cases.
2007-09-26gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher.Wim Taymans1-0/+12
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose), (gst_rtp_bin_finalize): Fix cleanup crasher. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (calculate_skew): * gst/rtpmanager/rtpjitterbuffer.h: Dynamically adjust the skew calculation window so that we calculate it over a period of around 2 seconds.
2007-09-26gst/librfb/gstrfbsrc.c: fix bug from generic/states.gdbThijs Vermeir1-0/+5
Original commit message from CVS: * gst/librfb/gstrfbsrc.c: fix bug from generic/states.gdb
2007-09-26gst/flv/gstflvparse.c: codec_data is needed for every tag not just the first ↵Julien Moutte1-0/+7
one. (Fix a stupid bug i introduced with... Original commit message from CVS: 2007-09-26 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): codec_data is needed for every tag not just the first one. (Fix a stupid bug i introduced without testing)
2007-09-26gst/flv/gstflvparse.c: Fix bit masks operations to be sure we detect the ↵Julien Moutte1-0/+7
codec_tags and sample rates correctly. Original commit message from CVS: 2007-09-26 Julien MOUTTE <julien@moutte.net> * gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio), (gst_flv_parse_tag_video): Fix bit masks operations to be sure we detect the codec_tags and sample rates correctly. Fix raw audio caps generation.
2007-09-24Massive leak fixing, plus code cleanups.Stefan Kost1-0/+18
Original commit message from CVS: * ext/audioresample/gstaudioresample.c: * ext/x264/gstx264enc.c: * gst/dvdspu/gstdvdspu.c: * gst/dvdspu/gstdvdspu.h: * gst/festival/gstfestival.c: * gst/h264parse/gsth264parse.c: * gst/mpegtsparse/mpegtspacketizer.c: * gst/mpegtsparse/mpegtsparse.c: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * gst/nuvdemux/gstnuvdemux.c: * sys/dshowsrcwrapper/gstdshowaudiosrc.c: * sys/dshowsrcwrapper/gstdshowvideosrc.c: * sys/vcd/vcdsrc.c: Massive leak fixing, plus code cleanups.
2007-09-22po/LINGUAS: Added translations.Thomas Vander Stichele1-0/+5
Original commit message from CVS: * po/LINGUAS: Added translations.
2007-09-22po/pl.po: Added Polish translation.Thomas Vander Stichele1-0/+7
Original commit message from CVS: translated by: Jakub Bogusz <qboosh@pld-linux.org> * po/pl.po: Added Polish translation.
2007-09-22po/fi.po: Added Finnish translation.Thomas Vander Stichele1-0/+7
Original commit message from CVS: translated by: Ilkka Tuohela <hile@iki.fi> * po/fi.po: Added Finnish translation.
2007-09-22po/es.po: Added Spanish translation.Thomas Vander Stichele1-0/+7
Original commit message from CVS: translated by: Jorge González González <aloriel@gmail.com> * po/es.po: Added Spanish translation.
2007-09-22po/bg.po: Added Bulgarian translation.Thomas Vander Stichele1-0/+7
Original commit message from CVS: translated by: Alexander Shopov <ash@contact.bg> * po/bg.po: Added Bulgarian translation.
2007-09-21docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.Sebastian Dröge1-0/+8
Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy. * ext/gio/gstgiosink.h: * ext/gio/gstgiosrc.h: Mark private fields of the instance structs private.
2007-09-21docs/plugins/: Add the GIO plugin to the docs and do a make update while ↵Sebastian Dröge1-0/+40
doing that. Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.signals: * docs/plugins/inspect/plugin-bz2.xml: * docs/plugins/inspect/plugin-cdxaparse.xml: * docs/plugins/inspect/plugin-dfbvideosink.xml: * docs/plugins/inspect/plugin-dtsdec.xml: * docs/plugins/inspect/plugin-equalizer.xml: * docs/plugins/inspect/plugin-faac.xml: * docs/plugins/inspect/plugin-faad.xml: * docs/plugins/inspect/plugin-filter.xml: * docs/plugins/inspect/plugin-freeze.xml: * docs/plugins/inspect/plugin-gio.xml: * docs/plugins/inspect/plugin-gsm.xml: * docs/plugins/inspect/plugin-gstrtpmanager.xml: * docs/plugins/inspect/plugin-h264parse.xml: * docs/plugins/inspect/plugin-modplug.xml: * docs/plugins/inspect/plugin-mpeg2enc.xml: * docs/plugins/inspect/plugin-musepack.xml: * docs/plugins/inspect/plugin-musicbrainz.xml: * docs/plugins/inspect/plugin-nsfdec.xml: * docs/plugins/inspect/plugin-replaygain.xml: * docs/plugins/inspect/plugin-soundtouch.xml: * docs/plugins/inspect/plugin-spcdec.xml: * docs/plugins/inspect/plugin-spectrum.xml: * docs/plugins/inspect/plugin-speed.xml: * docs/plugins/inspect/plugin-tta.xml: * docs/plugins/inspect/plugin-videosignal.xml: * docs/plugins/inspect/plugin-xingheader.xml: * docs/plugins/inspect/plugin-xvid.xml: Add the GIO plugin to the docs and do a make update while doing that. * ext/gio/gstgiosrc.c: (gst_gio_src_start): Fix a small memleak.
2007-09-21Add a GIO/GVFS plugin with source and sink elements. This will only be ↵René Stadler1-0/+36
enabled when --enable-experimental is given to... Original commit message from CVS: Patch by: René Stadler <mail at renestadler dot de> * configure.ac: * ext/Makefile.am: * ext/gio/Makefile.am: * ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek), (gst_gio_get_supported_protocols), (gst_gio_uri_handler_get_type_sink), (gst_gio_uri_handler_get_type_src), (gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri), (gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init), (gst_gio_uri_handler_do_init), (plugin_init): * ext/gio/gstgio.h: * ext/gio/gstgiosink.c: (gst_gio_sink_base_init), (gst_gio_sink_class_init), (gst_gio_sink_init), (gst_gio_sink_finalize), (gst_gio_sink_set_property), (gst_gio_sink_get_property), (gst_gio_sink_start), (gst_gio_sink_stop), (gst_gio_sink_unlock), (gst_gio_sink_unlock_stop), (gst_gio_sink_event), (gst_gio_sink_render), (gst_gio_sink_query): * ext/gio/gstgiosink.h: * ext/gio/gstgiosrc.c: (gst_gio_src_base_init), (gst_gio_src_class_init), (gst_gio_src_init), (gst_gio_src_finalize), (gst_gio_src_set_property), (gst_gio_src_get_property), (gst_gio_src_start), (gst_gio_src_stop), (gst_gio_src_get_size), (gst_gio_src_is_seekable), (gst_gio_src_unlock), (gst_gio_src_unlock_stop), (gst_gio_src_check_get_range), (gst_gio_src_create): * ext/gio/gstgiosrc.h: Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to configure for now as the GIO API is not stable yet. Fixes #476916.
2007-09-21gst/librfb/: Added offset-x, offset-y, width and height property for ↵Thijs Vermeir1-0/+8
selecting a region from the screen Original commit message from CVS: * gst/librfb/gstrfbsrc.c: * gst/librfb/rfbdecoder.c: * gst/librfb/rfbdecoder.h: Added offset-x, offset-y, width and height property for selecting a region from the screen
2007-09-21gst/librfb/gstrfbsrc.c: Minimum raw encoding is working nowThijs Vermeir1-0/+7
Original commit message from CVS: * gst/librfb/gstrfbsrc.c: Minimum raw encoding is working now * gst/librfb/rfbdecoder.c: fix address while reading from stream
2007-09-20gst/librfb/gstrfbsrc.c: raw encoding is working, but it looks like the ↵Thijs Vermeir1-0/+6
ffmpegcolorspace plugin can't handle high reso... Original commit message from CVS: * gst/librfb/gstrfbsrc.c: raw encoding is working, but it looks like the ffmpegcolorspace plugin can't handle high resolutions
2007-09-20gst/librfb/gstrfbsrc.c: bpp, depth and endianness are now set from the stream.Thijs Vermeir1-0/+6
Original commit message from CVS: * gst/librfb/gstrfbsrc.c: bpp, depth and endianness are now set from the stream.
2007-09-20examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.Wim Taymans1-0/+5
Original commit message from CVS: * examples/app/appsrc_ex.c: (main): Fix compilation after changing the name of a method.
2007-09-20Fix memory leaks. More to come.Stefan Kost1-0/+12
Original commit message from CVS: * ext/alsaspdif/alsaspdifsink.c: * ext/timidity/gsttimidity.c: * ext/timidity/gstwildmidi.c: * gst/mpegvideoparse/mpegvideoparse.c: Fix memory leaks. More to come. * tests/check/Makefile.am: * tests/check/generic/states.c: Improved state change unit test.
2007-09-20gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. ↵Wim Taymans1-0/+14
Fixes #478566. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session), (gst_rtp_bin_class_init): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpsession.c: (on_ssrc_active), (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_event_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_ssrc_active), (rtp_session_process_rb): * gst/rtpmanager/rtpsession.h: Add notification of active SSRCs to various RTP elements. Fixes #478566.
2007-09-20gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access ↵Wim Taymans1-0/+16
the element even without linking to the ... Original commit message from CVS: * gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID), (gst_app_sink_class_init), (gst_app_sink_init), (gst_app_sink_dispose), (gst_app_sink_finalize), (gst_app_sink_set_property), (gst_app_sink_get_property), (gst_app_sink_flush_unlocked), (gst_app_sink_start), (gst_app_sink_event), (gst_app_sink_getcaps), (gst_app_sink_set_caps), (gst_app_sink_get_caps), (gst_app_sink_is_eos), (gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer): * gst-libs/gst/app/gstappsink.h: Add properties, signals and actions to access the element even without linking to the library. Fix some method names and signatures.
2007-09-19ext/faad/gstfaad.c: Don't set channel positions on regular mono and stereo ↵Jonathan Matthew1-0/+9
cases. Original commit message from CVS: Patch by: Jonathan Matthew <jonathan at kaolin wh9 net> * ext/faad/gstfaad.c: (gst_faad_chanpos_to_gst), (gst_faad_srcgetcaps), (gst_faad_update_caps): Don't set channel positions on regular mono and stereo cases. Fixes #476370.
2007-09-19gst/librfb/: It is now possible to connect to a vncserver. there are still ↵Thijs Vermeir1-0/+9
some issues with the ouput of the screen. ... Original commit message from CVS: * gst/librfb/gstrfbsrc.c: * gst/librfb/rfbdecoder.c: * gst/librfb/rfbdecoder.h: It is now possible to connect to a vncserver. there are still some issues with the ouput of the screen. Looks like some lines are confused
2007-09-19Ignore registries in any format.Stefan Kost1-0/+6
Original commit message from CVS: * docs/plugins/.cvsignore: * tests/check/.cvsignore: Ignore registries in any format.
2007-09-19gst/real/gstrealvideodec.*: Don't generate an error for occasional decoding ↵Wim Taymans1-0/+11
errors. Original commit message from CVS: * gst/real/gstrealvideodec.c: (gst_real_video_dec_chain), (open_library), (gst_real_video_dec_init), (gst_real_video_dec_set_property), (gst_real_video_dec_get_property), (gst_real_video_dec_class_init): * gst/real/gstrealvideodec.h: Don't generate an error for occasional decoding errors. Add max-errors property. Error out when we receive max-errors in a row. Fixes #478159.
2007-09-19gst/librfb/gstrfbsrc.c: Add password property (write only)Thijs Vermeir1-0/+10
Original commit message from CVS: * gst/librfb/gstrfbsrc.c: Add password property (write only) * gst/librfb/rfbdecoder.c: Read the reason on failure Use the password property for authentication * gst/librfb/rfbdecoder.h: Add defines for version checking
2007-09-19ext/directfb/dfbvideosink.c: When finalizing GstDfbSurface, a subclass of ↵Sebastian Dröge1-0/+7
GstBuffer, correctly chain up to the parent... Original commit message from CVS: * ext/directfb/dfbvideosink.c: (gst_dfbvideosink_surface_destroy), (gst_dfbsurface_class_init): When finalizing GstDfbSurface, a subclass of GstBuffer, correctly chain up to the parent class to free everything, including caps.
2007-09-19gst/librfb/: VNC Authentication should be working now temperaly with fake ↵Thijs Vermeir1-0/+11
password 'testtest' Original commit message from CVS: * gst/librfb/Makefile.am: * gst/librfb/d3des.c: * gst/librfb/d3des.h: * gst/librfb/rfbdecoder.c: * gst/librfb/vncauth.c: * gst/librfb/vncauth.h: VNC Authentication should be working now temperaly with fake password 'testtest'
2007-09-18gst/librfb/rfbdecoder.*: Added some documentation about security handling ↵Thijs Vermeir1-0/+7
start implementing security handling for rf... Original commit message from CVS: * gst/librfb/rfbdecoder.c: * gst/librfb/rfbdecoder.h: Added some documentation about security handling start implementing security handling for rfb 3.3
2007-09-18gst/spectrum/: Handling window resize.Stefan Kost1-0/+6
Original commit message from CVS: * gst/spectrum/demo-audiotest.c: * gst/spectrum/demo-osssrc.c: Handling window resize.
2007-09-18ChangeLog: Add missing newline.Stefan Kost1-0/+20
Original commit message from CVS: * ChangeLog: Add missing newline. * gst/librfb/rfbdecoder.c: Fix the build (missing stdlib.h). * gst/spectrum/gstspectrum.c: * gst/spectrum/gstspectrum.h: Use basetransform segment so that it is correctly managed on flushes and start/stop. Report message timestamp as stream time, which is what an application can understand. (Yes these are adapted from wim recent level element changes)
2007-09-17gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one ↵Wim Taymans1-0/+22
was created first in the ssrc demuxer. Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Link to the right pads regardless of which one was created first in the ssrc demuxer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsource.c: (calculate_jitter): Improve debugging. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links): * gst/rtpmanager/gstrtpssrcdemux.h: Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
2007-09-16gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.Wim Taymans1-0/+27
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): Use lock to protect variable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop): Reconstruct GST timestamp from RTP timestamps based on measured clock skew and sync offset. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_set_tail_changed), (rtp_jitter_buffer_set_clock_rate), (rtp_jitter_buffer_get_clock_rate), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek): * gst/rtpmanager/rtpjitterbuffer.h: Measure clock skew. Add callback to be notfied when a new packet was inserted at the tail. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Remove clock skew detection, it's move to the jitterbuffer now.
2007-09-16ext/amrwb/gstamrwbenc.*: Add property to control bandmode. Fixes #477306.Daniel Charles1-0/+10
Original commit message from CVS: Patch by: Daniel Charles <dcharles at ti dot com> * ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_bandmode_get_type), (gst_amrwbenc_set_property), (gst_amrwbenc_get_property), (gst_amrwbenc_class_init), (gst_amrwbenc_chain): * ext/amrwb/gstamrwbenc.h: Add property to control bandmode. Fixes #477306.
2007-09-15gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.Wim Taymans1-0/+18
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session): Also set NTP base time on new sessions. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query), (gst_rtp_jitter_buffer_set_property), (gst_rtp_jitter_buffer_get_property): Use the right lock to protect our variables. Fix some comment. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_getcaps_send_rtp), (gst_rtp_session_chain_send_rtp), (create_send_rtp_sink): Implement getcaps on the sender sinkpad so that payloaders can negotiate the right SSRC.
2007-09-12gst/rtpmanager/: Various leak fixes.Wim Taymans1-0/+19
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_session), (free_session), (get_client), (free_client), (gst_rtp_bin_associate), (free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose), (gst_rtp_bin_finalize): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_finalize): * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize), (gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose): * gst/rtpmanager/rtpsession.c: (rtp_session_finalize): * gst/rtpmanager/rtpsession.h: Various leak fixes.
2007-09-12gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so ↵Wim Taymans1-0/+48
that we can generate better Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base), (gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp): Calculate and configure the NTP base time so that we can generate better NTP times in SR packets. Set caps on new ghostpad. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop): Clean debug statement. * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_init), (gst_rtp_session_set_property), (gst_rtp_session_get_property), (get_current_ntp_ns_time), (rtcp_thread), (gst_rtp_session_event_recv_rtp_sink), (gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_event_send_rtp_sink), (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink), (create_send_rtp_sink): * gst/rtpmanager/gstrtpsession.h: Add ntp-ns-base property to convert running_time to NTP time. Handle NEWSEGMENT events on send and recv RTP pads so that we can calculate the running time and thus NTP time of the packets. Simplify getting the current NTP time using the pipeline clock. Implement internal links functions. Use the buffer timestamp to calculate the NTP time instead of the clock. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links), (gst_rtp_ssrc_demux_src_query): * gst/rtpmanager/gstrtpssrcdemux.h: Implement internal links function. Calculate the diff between different streams, this might be used later to get the inter stream latency. * gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp): Simple cleanup. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr): Make the clock skew window a little bigger. Apply the clock skew to all buffers, not just one with a new timestamp. Calculate and debug sender clock drift. Use extended last timestamp to interpollate for SR reports.
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