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* gst/equalizer/: Allow setting 0 as bandwidth and handle this correctly.Sebastian Dröge2007-11-032-7/+22
| | | | | | | | | | | | | | Original commit message from CVS: * gst/equalizer/demo.c: (main): * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_class_init), (setup_filter): Allow setting 0 as bandwidth and handle this correctly. Also handle a bandwidth of rate/2 properly. * gst/equalizer/gstiirequalizernbands.c: (gst_iir_equalizer_nbands_class_init): Make it possible to generate a N-band equalizer with 1 bands. The previous limit of 2 was caused by a nowadays replaced calculation doing a division by zero if number of bands was 1.
* Fix includes for MSVC and GLib-2.14.0 (#492388).Ole André Vadla Ravnås2007-11-027-24/+36
| | | | | | | | | | | | | | | Original commit message from CVS: Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> * configure.ac: * gst/udp/gstdynudpsink.c: * gst/udp/gstdynudpsink.h: * gst/udp/gstmultiudpsink.c: * gst/udp/gstmultiudpsink.h: * gst/udp/gstudpsink.c: * gst/udp/gstudpsink.h: Fix includes for MSVC and GLib-2.14.0 (#492388). * gst/udp/gstudpsrc.c: (gst_udpsrc_start): No more pipe define since GLib-2.14.0, need to use _pipe() directly.
* gst/law/mulaw-decode.*: Calculate outgoing buffer duration if incoming ↵Edward Hervey2007-11-022-1/+11
| | | | | | | | | | | buffer didn't have a valid duration. Original commit message from CVS: * gst/law/mulaw-decode.c: (mulawdec_sink_setcaps), (gst_mulawdec_chain): * gst/law/mulaw-decode.h: Calculate outgoing buffer duration if incoming buffer didn't have a valid duration.
* gst/equalizer/: Add small demo application based on the spectrum demo ↵Sebastian Dröge2007-10-302-12/+36
| | | | | | | | | | | | | | | | | applications that gets white noise as input, pu... Original commit message from CVS: * gst/equalizer/Makefile.am: * gst/equalizer/demo.c: (on_window_destroy), (on_configure_event), (on_gain_changed), (on_bandwidth_changed), (on_freq_changed), (draw_spectrum), (message_handler), (main): Add small demo application based on the spectrum demo applications that gets white noise as input, pushes it through an equalizer and paints the spectrum. For every equalizer band it's possible to set gain, bandwidth and frequency. * gst/equalizer/gstiirequalizer.c: (setup_filter): Add some guarding against too large or too small frequencies and bandwidths. Also improve debugging a bit.
* gst/equalizer/gstiirequalizer.c: Replace filters with a bit better filters ↵Sebastian Dröge2007-10-301-48/+70
| | | | | | | | | | | | | | | | | | for which we can actually find documentati... Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_band_get_property), (gst_iir_equalizer_band_class_init), (arg_to_scale), (setup_filter), (gst_iir_equalizer_compute_frequencies): Replace filters with a bit better filters for which we can actually find documentation, which don't change anything on zero gain, etc. Make the frequency property of the bands writable, rename the band-width property to bandwidth and change the meaning to the frequency difference between bandedges, change the meaning of the gain property to dB instead of a weird scale between -1 and 1 that has no real meaning.
* gst/qtdemux/qtdemux.c: Smarter combine_flow code that also deals with ↵Wim Taymans2007-10-301-11/+30
| | | | | | | | | | | downstream elements returning UNEXPECTED when t... Original commit message from CVS: * gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment), (gst_qtdemux_combine_flows), (gst_qtdemux_loop_state_movie): Smarter combine_flow code that also deals with downstream elements returning UNEXPECTED when they receive data out of the segment boundaries. Fixes #491305.
* gst/interleave/interleave.c: Let's not call every request pad we create ↵Tim-Philipp Müller2007-10-271-12/+15
| | | | | | | | | | | | | "sink%d", that'll create problems if there's ... Original commit message from CVS: * gst/interleave/interleave.c: (gst_interleave_request_new_pad): Let's not call every request pad we create "sink%d", that'll create problems if there's to be more than one pad. Fixes #490682. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/interleave.c: Add unit test for the above.
* Improve documentation, write some tests for multifilesrc/sink for upcoming ↵David Schleef2007-10-253-60/+79
| | | | | | | | | | | | | ->good review. Original commit message from CVS: * gst/multifile/Makefile.am: * gst/multifile/gstmultifilesink.c: * gst/multifile/gstmultifilesrc.c: * tests/check/Makefile.am: * tests/check/elements/multifile.c: Improve documentation, write some tests for multifilesrc/sink for upcoming ->good review.
* gst/rtsp/gstrtspsrc.c: Fix race when pausing a RTSP stream in interleaved.Tommi Myöhänen2007-10-221-0/+17
| | | | | | | | Original commit message from CVS: Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com> * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved): Fix race when pausing a RTSP stream in interleaved. Fixes #475784.
* gst/rtp/gstrtpmp4vpay.c: Use correct unref function for buffers. #488844.Peter Kjellerstedt2007-10-221-1/+1
| | | | | | | Original commit message from CVS: Patch by: Peter Kjellerstedt <pkj at axis com> * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_finalize): Use correct unref function for buffers. #488844.
* Add some debug and sync tests with the fix.Stefan Kost2007-10-191-0/+2
| | | | | | | Original commit message from CVS: * gst/avi/gstavimux.c: * tests/check/elements/avimux.c: Add some debug and sync tests with the fix.
* gst/udp/gstudpsrc.c: When the socket is used by the app for other purposes, ↵Laurent Glayal2007-10-181-8/+6
| | | | | | | | | | | don't generate an error if there is activ... Original commit message from CVS: Based on patch by: Laurent Glayal <spglegle yahoo fr> * gst/udp/gstudpsrc.c: (gst_udpsrc_create): When the socket is used by the app for other purposes, don't generate an error if there is activaty on the socket that is not data related. Fixes #487488.
* gst/rtp/gstrtph264pay.c: Set marker bit correctly.Anders Skargren2007-10-181-2/+1
| | | | | | | Original commit message from CVS: Patch by: Anders Skargren <anders dot skargren at axis dot com> * gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer): Set marker bit correctly.
* gst/equalizer/gstiirequalizer.c: Add a missing break.Sebastian Dröge2007-10-181-0/+1
| | | | | | | Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property): Add a missing break.
* gst/equalizer/gstiirequalizer.*: Move bandwidth property to the separate ↵Sebastian Dröge2007-10-182-79/+66
| | | | | | | | | | | | | | bands and add float64 support. Original commit message from CVS: * gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_band_set_property), (gst_iir_equalizer_band_get_property), (gst_iir_equalizer_band_class_init), (gst_iir_equalizer_band_init), (gst_iir_equalizer_band_get_type), (gst_iir_equalizer_class_init), (setup_filter), (gst_iir_equalizer_setup): * gst/equalizer/gstiirequalizer.h: Move bandwidth property to the separate bands and add float64 support.
* gst/rtsp/gstrtspsrc.c: Use allowed name for the GstStructure.Wim Taymans2007-10-171-1/+1
| | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open): Use allowed name for the GstStructure.
* Use new gst_bus_pop_filtered().Tim-Philipp Müller2007-10-171-7/+3
| | | | | | | Original commit message from CVS: * ext/gconf/gstswitchsink.c: * gst/autodetect/gstautoaudiosink.c: Use new gst_bus_pop_filtered().
* gst/id3demux/id3v2frames.c: Extract license/copyright URIs from ID3v2 WCOP ↵Jason Kivlighn2007-10-111-0/+57
| | | | | | | | | | | | | | frames (Fixes #447000). Original commit message from CVS: Based on patch by: Jason Kivlighn <jkivlighn gmail com> * gst/id3demux/id3v2frames.c: Extract license/copyright URIs from ID3v2 WCOP frames (Fixes #447000). * tests/check/elements/id3demux.c: * tests/files/Makefile.am: * tests/files/id3-447000-wcop.tag: Add simple unit test.
* gst/rtsp/gstrtspsrc.c: Fix compiler warning by using GST_CLOCK_TIME_NONE to ↵Jan Schmidt2007-10-081-1/+1
| | | | | | | | | initialise a GstClockTime. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush): Fix compiler warning by using GST_CLOCK_TIME_NONE to initialise a GstClockTime.
* gst/rtsp/gstrtspsrc.c: More seeking fixes, mostly passing around the new ↵Wim Taymans2007-10-081-38/+45
| | | | | | | | | | | | | | | playback segment in order to configure it pr... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek), (gst_rtspsrc_configure_caps), (gst_rtspsrc_loop_udp), (gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_play), (gst_rtspsrc_change_state): More seeking fixes, mostly passing around the new playback segment in order to configure it properly. Also reset base_time of udp sources when setting them back to PLAYING as a temporary hack until core supports seek in live sources properly.
* gst/rtp/gstrtpmp4adepay.c: Fix caps as to not confuse autopluggers.Wim Taymans2007-10-081-0/+1
| | | | | | Original commit message from CVS: * gst/rtp/gstrtpmp4adepay.c: Fix caps as to not confuse autopluggers.
* gst/id3demux/: Port ID3 tag demuxer over to the new GstTagDemux in -base ↵Tim-Philipp Müller2007-10-065-1171/+123
| | | | | | | | | | | | | | (now would be a good time to test re-importi... Original commit message from CVS: * gst/id3demux/gstid3demux.c: * gst/id3demux/gstid3demux.h: * gst/id3demux/id3tags.c: * gst/id3demux/id3tags.h: * gst/id3demux/id3v2frames.c: Port ID3 tag demuxer over to the new GstTagDemux in -base (now would be a good time to test re-importing your music collection).
* gst/apetag/: Port APE tag demuxer over to the new GstTagDemux in -base.Tim-Philipp Müller2007-10-065-1585/+14
| | | | | | | | | | Original commit message from CVS: * gst/apetag/Makefile.am: * gst/apetag/gstapedemux.c: * gst/apetag/gstapedemux.h: * gst/apetag/gsttagdemux.c: * gst/apetag/gsttagdemux.h: Port APE tag demuxer over to the new GstTagDemux in -base.
* gst/rtsp/gstrtspsrc.c: Improve flushing behaviour.Wim Taymans2007-10-051-17/+93
| | | | | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush), (gst_rtspsrc_perform_seek), (gst_rtspsrc_handle_src_event), (gst_rtspsrc_handle_internal_src_query), (gst_rtspsrc_handle_src_query), (new_session_pad), (gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_send_cmd): Improve flushing behaviour. Set state of the udp sources to PAUSE/PLAYING correctly. Handle events and queries for UDP and TCP transport now.
* gst/rtp/: Add log category.Stefan Kost2007-10-042-0/+12
| | | | | | | Original commit message from CVS: * gst/rtp/gstrtpgsmdepay.c: * gst/rtp/gstrtpgsmpay.c: Add log category.
* gst/avi/gstavimux.*: Also save codec data for audio streams. Fixes #482495.Stefan Kost2007-10-022-6/+37
| | | | | | | Original commit message from CVS: * gst/avi/gstavimux.c: * gst/avi/gstavimux.h: Also save codec data for audio streams. Fixes #482495.
* gst/avi/gstavimux.c: Fix "Index entry has invalid stream nr 1".Stefan Kost2007-10-021-10/+24
| | | | | | | Original commit message from CVS: * gst/avi/gstavimux.c: Fix "Index entry has invalid stream nr 1". Add support for muxing aac - work in progress (see #482495).
* gst/rtsp/gstrtspsrc.*: Parse bandwidth modifiers, they are not yet ↵Wim Taymans2007-10-012-0/+56
| | | | | | | | | | | | configured in the session manager because we don't... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_bandwidth), (gst_rtspsrc_collect_bandwidth), (gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved): * gst/rtsp/gstrtspsrc.h: Parse bandwidth modifiers, they are not yet configured in the session manager because we don't have an API for that yet.
* gst/rtsp/gstrtspsrc.c: Use shiny new function in -base to get the default ↵Wim Taymans2007-10-011-50/+22
| | | | | | | | | | clock-rate. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps), (gst_rtspsrc_loop_interleaved): Use shiny new function in -base to get the default clock-rate. Update some docs.
* gst/rtsp/gstrtspsrc.*: In TCP mode, only timestamp the first buffer. TCP is ↵Wim Taymans2007-09-282-9/+22
| | | | | | | | | | | | not real time and it does not make sense ... Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_play): * gst/rtsp/gstrtspsrc.h: In TCP mode, only timestamp the first buffer. TCP is not real time and it does not make sense to try to skew compensate, also some servers send the first batch of data in a burst.
* gst/matroska/matroska-demux.c: Fix setting the discont flag on the first ↵Tim-Philipp Müller2007-09-271-0/+6
| | | | | | | | | | | buffer pushed downstream for formats with pr... Original commit message from CVS: * gst/matroska/matroska-demux.c: Fix setting the discont flag on the first buffer pushed downstream for formats with private codec data that needs to be deserialised into buffers (such as vorbis and FLAC when in a matroska container).
* gst/rtp/gstrtpmp4vpay.*: Free the config string. Fixes #480707.Antoine Tremblay2007-09-272-7/+18
| | | | | | | | | | | Original commit message from CVS: Patch by: Antoine Tremblay <hexa00 at gmail dot com> * gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init), (gst_rtp_mp4v_pay_finalize), (gst_rtp_mp4v_pay_flush), (gst_rtp_mp4v_pay_handle_buffer): * gst/rtp/gstrtpmp4vpay.h: Free the config string. Fixes #480707. Clean up the timestamp code a little.
* gst/rtsp/gstrtspsrc.*: Set timestamps on RTP buffers in interleaved mode.Wim Taymans2007-09-262-8/+20
| | | | | | | | | | | Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (gst_rtspsrc_handle_src_query), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp), (gst_rtspsrc_close): * gst/rtsp/gstrtspsrc.h: Set timestamps on RTP buffers in interleaved mode. Mark first buffers with a DISCONT. Remove flush hack now that sync for live sources has been figured out.
* gst/udp/gstudpsrc.c: Update documentation.Wim Taymans2007-09-261-3/+4
| | | | | | Original commit message from CVS: * gst/udp/gstudpsrc.c: (gst_udpsrc_create): Update documentation.
* gst/qtdemux/gstrtpxqtdepay.*: Fail if we don't know the quicktime format.Wim Taymans2007-09-262-2/+13
| | | | | | | | Original commit message from CVS: * gst/qtdemux/gstrtpxqtdepay.c: (gst_rtp_xqt_depay_process), (gst_rtp_xqt_depay_change_state): * gst/qtdemux/gstrtpxqtdepay.h: Fail if we don't know the quicktime format.
* Add support for the new GST_TAG_COMPOSER (#459809).Tim-Philipp Müller2007-09-251-0/+1
| | | | | | | | Original commit message from CVS: * ext/taglib/gstapev2mux.cc: * ext/taglib/gstid3v2mux.cc: * gst/apetag/gstapedemux.c: Add support for the new GST_TAG_COMPOSER (#459809).
* gst/law/: Compulsive clean-ups: use boilerplate macros, add debug ↵Tim-Philipp Müller2007-09-256-189/+157
| | | | | | | | | | | | | | | categories, fix up things to conform to symbol nome... Original commit message from CVS: * gst/law/alaw-decode.c: * gst/law/alaw-decode.h: * gst/law/alaw-encode.c: * gst/law/alaw-encode.h: * gst/law/alaw.c: * gst/law/mulaw-conversion.h: Compulsive clean-ups: use boilerplate macros, add debug categories, fix up things to conform to symbol nomenklatura, etc.
* gst/law/: Use static tables for A-Law decoding and encoding; this makesLaurent Glayal2007-09-252-6/+246
| | | | | | | | | | | Original commit message from CVS: Based on patch by: Laurent Glayal <spglegle yahoo fr> * gst/law/alaw-decode.c: * gst/law/alaw-encode.c: Use static tables for A-Law decoding and encoding; this makes A-Law decoding and encoding less CPU-intensive, but increases the binary size a bit. Leaving old code around for now, selectable by a define in the code. Fixes #435435.
* gst/qtdemux/qtdemux.c: Add fourccs for MPEG2 HDV streams. Fixes #479960.Sebastian Dröge2007-09-251-0/+7
| | | | | | | Original commit message from CVS: Patch by: <j at bootlab dot org> * gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add fourccs for MPEG2 HDV streams. Fixes #479960.
* Massive leak fixing, plus code cleanups.Stefan Kost2007-09-242-4/+4
| | | | | | | | | | | | | | | | | | | 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.
* gst/rtp/gstrtpamrdepay.c: Set outgoing packet duration because we can. Fixes ↵Wim Taymans2007-09-211-0/+3
| | | | | | | | #478244 some more. Original commit message from CVS: * gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_process): Set outgoing packet duration because we can. Fixes #478244 some more.
* gst/rtp/gstrtpL16pay.c: Removed some unused code.Wim Taymans2007-09-197-8/+26
| | | | | | | | | | | | | | | Original commit message from CVS: * gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer): Removed some unused code. * gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer): * gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer): * gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer): * gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer): * gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet), (gst_rtp_theora_pay_flush_packet): * gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet): Try to preserve the incomming buffer duration on the outgoing packets. Fixes #478244.
* ChangeLog: Add missing newline.Stefan Kost2007-09-182-19/+1
| | | | | | | | | | | | | | 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)
* gst/: Fix compiler warnings shown with Forte.Jan Schmidt2007-09-172-8/+17
| | | | | | | | | | Original commit message from CVS: * gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init): * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream), (new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos), (gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_handle_message): Fix compiler warnings shown with Forte.
* gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to ↵Wim Taymans2007-09-171-4/+18
| | | | | | | | | | configure for some reason. Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams), (gst_rtspsrc_dup_printf): Give meaningfull error when all streams failed to configure for some reason.
* gst/rtp/README: Update README with the design for synchronisation rules of ↵Wim Taymans2007-09-161-22/+144
| | | | | | | | | RTP on sender and receiver. Original commit message from CVS: * gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver.
* gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the ↵Sebastian Dröge2007-09-141-42/+32
| | | | | | | | | | | | element driving the pipeline is responsible f... Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_loop), (gst_wavparse_chain): Don't push EOS from the chain function, the element driving the pipeline is responsible for this. The bug this was meant to fix seems to be queue not forwarding EOS in all cases (see #476514).
* gst/level/gstlevel.*: Use basetransform segment so that it is correctly ↵Wim Taymans2007-09-132-35/+1
| | | | | | | | | | | | | managed on flushes and start/stop. Original commit message from CVS: * gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start), (gst_level_transform_ip): * gst/level/gstlevel.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.
* gst/wavparse/gstwavparse.c: Add EOS logic for the push-based mode too. Fixes ↵Sebastian Dröge2007-09-131-32/+42
| | | | | | | | | #476514. Original commit message from CVS: * gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos), (gst_wavparse_loop), (gst_wavparse_chain): Add EOS logic for the push-based mode too. Fixes #476514.
* gst/law/: Fix law encoder timestamps.Wim Taymans2007-09-124-13/+29
| | | | | | | | | | Original commit message from CVS: * gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain): * gst/law/alaw-encode.h: * gst/law/mulaw-encode.c: (gst_mulawenc_init), (gst_mulawenc_chain): * gst/law/mulaw-encode.h: Fix law encoder timestamps.
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