| Commit message (Collapse) | Author | Age | Files | Lines |
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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.
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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.
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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.
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"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.
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->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.
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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.
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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.
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Original commit message from CVS:
* gst/avi/gstavimux.c:
* tests/check/elements/avimux.c:
Add some debug and sync tests with the fix.
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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.
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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.
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Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
(gst_iir_equalizer_band_set_property):
Add a missing break.
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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.
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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
Use allowed name for the GstStructure.
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Original commit message from CVS:
* ext/gconf/gstswitchsink.c:
* gst/autodetect/gstautoaudiosink.c:
Use new gst_bus_pop_filtered().
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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.
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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.
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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.
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Original commit message from CVS:
* gst/rtp/gstrtpmp4adepay.c:
Fix caps as to not confuse autopluggers.
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(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).
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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.
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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.
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Original commit message from CVS:
* gst/rtp/gstrtpgsmdepay.c:
* gst/rtp/gstrtpgsmpay.c:
Add log category.
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Original commit message from CVS:
* gst/avi/gstavimux.c:
* gst/avi/gstavimux.h:
Also save codec data for audio streams. Fixes #482495.
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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).
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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.
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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.
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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.
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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).
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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.
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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.
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Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Update documentation.
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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.
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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).
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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.
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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.
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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.
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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.
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#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.
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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.
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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)
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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.
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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.
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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.
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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).
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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.
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#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.
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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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Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst-libs/gst/app/gstappsink.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvparse.c:
* gst/interleave/deinterleave.c:
* gst/switch/gstswitch.c:
Printf format fixes (#476128).
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they were received.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
Make udpsrc timestamp outgoing buffers based on when they were received.
Also make it output a segment in time.
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Original commit message from CVS:
* gst/avi/gstavidemux.c:
Plug a little leak. Little code cleanups.
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