Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered. Fix warnings that gtk-doc points out.
Introduce a new max-delay property that can only
be set before going to PLAYING or PAUSED. This
is used to limit the maximum delay and is set
to the current delay by default.
Using this will make sure that we have enough data
in our internal ringbuffer for the echo. With dynamic
reallocation of the ringbuffer as used before silence
could've been used as the echo directly after setting
a new delay.
Link to properties. Correct titles for examples. Document a few trivial cases. Keep lists in section file and docs/plugins/Makefile.am alphabetically ordered.
Save some allocations if the echo delay is increased often
during playback by always allocating enough memory to hold
data up to the next complete second, i.e. in the worst case
allocate memory for one additional second.
Add a note to the docs that audioecho's reverb will
sound metallic. This happens because for a real
reverb filter additional filtering is necessary.
Also note which values should be used for the delay
property to get an echo effect.
The element can add an echo and a simple reverb effect to
an audio stream but for a real reverb filter it would need
some additional filtering to prevent a metallic-sounding
result.
Original commit message from CVS:
Patch by: Luotao Fu <l dot fu at pengutronix dot de>
* gst/videocrop/gstvideocrop.c:
(gst_video_crop_get_image_details_from_caps):
Add 8bit grayscale support to videocrop plugin. Fixes#567952.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Implement a simple compensation algorithm for rounding errors.
This makes sure that a spectrum message is posted on the bus
every interval nanoseconds. Fixes bug #567955.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_segments):
Catch invalid and commonly wrong playback rates in the elst atoms.
Fixes#567800.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_reset_state):
Don't call gst_fft_f32_free() with NULL to prevent a
crash. Fixes bug #567642.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Use correct types for frame/fft counters and some minor
cleanup.
Original commit message from CVS:
* gst/spectrum/Makefile.am:
* gst/spectrum/README:
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_reset_state), (gst_spectrum_finalize),
(gst_spectrum_set_property), (gst_spectrum_start),
(gst_spectrum_stop), (gst_spectrum_setup),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Post a spectrum message on the bus for every interval, even
if the interval is small than the length of the FFT.
Fixes bug #567642.
Major cleanup of the spectrum element.
Original commit message from CVS:
Patch by: Thiago Sousa Santos <thiagoss@embedded.ufcg.edu.br>
* gst/qtdemux/qtdemux.c:
Fix format string for guint64.
Original commit message from CVS:
* gst/audiofx/audiochebband.c: (gst_audio_cheb_band_class_init),
(gst_audio_cheb_band_init), (gst_audio_cheb_band_finalize),
(gst_audio_cheb_band_set_property):
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_class_init),
(gst_audio_cheb_limit_init), (gst_audio_cheb_limit_finalize),
(gst_audio_cheb_limit_set_property):
* gst/audiofx/audiocheblimit.h:
* gst/audiofx/audiowsincband.c: (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_finalize),
(gst_audio_wsincband_set_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c: (gst_audio_wsinclimit_class_init),
(gst_audio_wsinclimit_init), (gst_audio_wsinclimit_finalize),
(gst_audio_wsinclimit_set_property):
* gst/audiofx/audiowsinclimit.h:
Use a custom mutex for protecting the instance fields instead of
the GstObject lock. Using the latter can lead to deadlocks, especially
with the FIR filters when updating the latency.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbasefirfilter.c:
(gst_audio_fx_base_fir_filter_dispose),
(gst_audio_fx_base_fir_filter_base_init),
(gst_audio_fx_base_fir_filter_class_init),
(gst_audio_fx_base_fir_filter_init),
(gst_audio_fx_base_fir_filter_push_residue),
(gst_audio_fx_base_fir_filter_setup),
(gst_audio_fx_base_fir_filter_transform),
(gst_audio_fx_base_fir_filter_start),
(gst_audio_fx_base_fir_filter_stop),
(gst_audio_fx_base_fir_filter_query),
(gst_audio_fx_base_fir_filter_query_type),
(gst_audio_fx_base_fir_filter_event),
(gst_audio_fx_base_fir_filter_set_kernel):
* gst/audiofx/audiofxbasefirfilter.h:
* gst/audiofx/audiofxbaseiirfilter.c:
Implement a base class for generic audio FIR filters.
* gst/audiofx/audiowsincband.c:
(gst_gst_audio_wsincband_mode_get_type),
(gst_gst_audio_wsincband_window_get_type),
(gst_audio_wsincband_base_init), (gst_audio_wsincband_class_init),
(gst_audio_wsincband_init), (gst_audio_wsincband_build_kernel),
(gst_audio_wsincband_setup), (gst_audio_wsincband_set_property),
(gst_audio_wsincband_get_property):
* gst/audiofx/audiowsincband.h:
* gst/audiofx/audiowsinclimit.c:
(gst_audio_wsinclimit_mode_get_type),
(gst_audio_wsinclimit_window_get_type),
(gst_audio_wsinclimit_base_init),
(gst_audio_wsinclimit_class_init), (gst_audio_wsinclimit_init),
(gst_audio_wsinclimit_build_kernel), (gst_audio_wsinclimit_setup),
(gst_audio_wsinclimit_set_property),
(gst_audio_wsinclimit_get_property):
* gst/audiofx/audiowsinclimit.h:
* tests/check/elements/audiowsincband.c: (GST_START_TEST):
* tests/check/elements/audiowsinclimit.c: (GST_START_TEST):
Use this new base class for audiowsincband and audiowsinclimit.
Also cleanup both elements.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
In push mode, error out if we get EOS before we've created any srcpads.
Handle (in pull mode) some files that have a truncated moov atom where
the final sub-atom is a 'free' atom and the contents of that are not
present in the file.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_video_pad_setcaps):
Some cleanups, refactoring and minor enhancements in caps handling.
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
(gst_matroska_mux_init), (gst_matroska_pad_reset),
(gst_matroska_pad_free), (gst_matroska_mux_reset),
(gst_matroska_mux_video_pad_setcaps),
(gst_matroska_mux_request_new_pad):
* tests/check/elements/matroskamux.c: (teardown_src_pad):
Only remove, release or reset what is appropriate upon state change.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_reset),
(gst_matroska_mux_handle_sink_event), (gst_matroska_mux_finish):
* gst/matroska/matroska-mux.h:
Remove internal taglist and fully use tagsetter interface.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_reset),
(gst_avi_mux_riff_get_avi_header):
* gst/avi/gstavimux.h:
Ensure header size invariance during subsequent rewrite by using
tags snapshot.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofxbaseiirfilter.c:
(gst_audio_fx_base_iir_filter_base_init),
(gst_audio_fx_base_iir_filter_dispose),
(gst_audio_fx_base_iir_filter_class_init),
(gst_audio_fx_base_iir_filter_init),
(gst_audio_fx_base_iir_filter_calculate_gain),
(gst_audio_fx_base_iir_filter_set_coefficients),
(gst_audio_fx_base_iir_filter_setup), (process),
(gst_audio_fx_base_iir_filter_transform_ip),
(gst_audio_fx_base_iir_filter_stop):
* gst/audiofx/audiofxbaseiirfilter.h:
Implement a base class for IIR filters.
* gst/audiofx/audiochebband.c: (gst_audio_cheb_band_base_init),
(gst_audio_cheb_band_class_init), (gst_audio_cheb_band_init),
(generate_coefficients), (gst_audio_cheb_band_set_property),
(gst_audio_cheb_band_setup):
* gst/audiofx/audiochebband.h:
* gst/audiofx/audiocheblimit.c: (gst_audio_cheb_limit_base_init),
(gst_audio_cheb_limit_class_init), (gst_audio_cheb_limit_init),
(generate_coefficients), (gst_audio_cheb_limit_set_property),
(gst_audio_cheb_limit_setup):
* gst/audiofx/audiocheblimit.h:
Use the IIR filter base class for the chebyshev filters.
Original commit message from CVS:
Patch by: j^ <j at oil21.org>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps),
(qtdemux_audio_caps):
Add codec mapping for xvid, fmp4 and ac3 tracks.
Fixes#565850
Original commit message from CVS:
* ext/pulse/pulsemixerctrl.c:
And remove temporary comment pointing to the bug ticket.
* gst/avi/gstavimux.c:
Move reoccuring logging to LOG and log instance too.
Original commit message from CVS:
* gst/avi/gstavimux.c:
Instead of filtering wrongly just use the mergemode. Applications is
use KEEP_ALL if they want to supress tag-events. Fixes#563221 for
avi for real (I hope). Everyone chime in, before I fix the others.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
More logging.
* gst/avi/gstavimux.c:
Handle more metadata fields. Better estimate of metadata size. Don't
merge received tags, if application has specified tags using
GST_TAG_MERGE_REPLACE_ALL. Fixes#563221 for avi.
Original commit message from CVS:
* gst/rtp/gstrtpjpegdepay.c: (gst_rtp_jpeg_depay_process):
Add an EOI marker at the end of the jpeg frame when it's missing.
Fixes#563056.
Original commit message from CVS:
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush),
(gst_rtp_mp4v_pay_event):
Don't try to push packets before we could find a valid config
startcode. Fixes#563509.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
Provide the parameters that are required for the format string
to fix a compiler warning.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render):
Make gst_multiudpsink_render() ignore errors from sendto() instead of
breaking streaming. Emit a warning instead. Fixes#562572.
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
Prevent further read/write actions taken to the connect-failed socket by
erroring out quickly. See #562258.
Original commit message from CVS:
2008-11-25 Julien Moutte <julien@fluendo.com>
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps): Add MPG1 and MPG2
fourcc
to supported qtdemux video codecs as I found some video clips
using
those.
Original commit message from CVS:
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
* gst/autodetect/gstautoaudiosrc.c: (gst_auto_audio_src_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
* gst/autodetect/gstautovideosrc.c: (gst_auto_video_src_detect):
Post an error when we can't set the internal ghostpad target.
Original commit message from CVS:
* gst/videocrop/gstvideocrop.c: (gst_video_crop_init),
(gst_video_crop_transform), (gst_video_crop_transform_caps),
(gst_video_crop_set_caps), (gst_video_crop_set_property):
* gst/videocrop/gstvideocrop.h:
Fix renegotiation when changing properties using the new basetransform
features. Fixes#561502.
* tests/icles/Makefile.am:
* tests/icles/videocrop2-test.c: (make_pipeline), (main):
Add crazy interactive test unit for dynamically changing properties.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (new_session_pad),
(gst_rtspsrc_parse_range):
Add some more debugging.
Use the reanges received from the server unconditionally.
Fixes#561625.
Original commit message from CVS:
Patch by: Tal Shalif <tshalif at nargila dot org>
* gst/qtdemux/qtdemux.c: (qtdemux_audio_caps):
Use G_{BIG,LITTLE}_ENDIAN instead of the non-GLib variants as
the latter don't exist on some systems (mingw). Fixes bug #561992.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpnetutils.c:
* gst/udp/gstudpnetutils.h:
* gst/udp/gstudpsrc.c:
Fix multiudpsink on OSX by passing the specific length of the socket,
refactor that into a function shared with the same thing in udpsrc.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
(uint64_ceiling_scale), (gst_wavparse_calculate_duration),
(gst_wavparse_stream_headers):
Fix the scaling code.
Fix parsing of the INFO chunks, we were reading the wrong number of
bytes. Fixes#561580.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Make mkvdemux aware of E-AC3.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c:
* gst/rtsp/gstrtspgoogle.h:
Remove google extension again, it's not needed anymore because we never
send multiple transports anymore.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_nat_method_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free),
(gst_rtspsrc_stream_configure_udp_sinks),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_send_dummy_packets),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add property to configure NAT traversal method.
Ignore EOS from the internal sinks.
Implement sending dummy packets as a (simple) method to open up
some firewalls.
Send PLAY request to the server after we started the udp sources.
Fixes#559545.
Original commit message from CVS:
Patch by: Yotam <sh dot yotam at gmail dot com>
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_event):
Flush the remaining frames on EOS. Fixes#560641.
Original commit message from CVS:
* gst/qtdemux/qtdemux.h (struct _GstQTDemux):
* gst/qtdemux/qtdemux.c (gst_qtdemux_do_seek): Queue up new
segment events instead of sending them from the seeking thread.
Fixes#559288.
(gst_qtdemux_push_pending_newsegment): New helper, sends out
queued newsegment events.
(gst_qtdemux_loop_state_movie): Voilà, call it here. Only need to
call it here, as we only seek when looping, and only push in the
movie state.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_tag_add_tmpo),
(qtdemux_tag_add_covr), (qtdemux_parse_udta):
* gst/qtdemux/qtdemux_fourcc.h:
* gst/qtdemux/qtdemux_types.c:
Add cover and alternative copyright tag, and enhance some existing
ones by marking them as container atoms.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_transports_string),
(gst_rtspsrc_change_state):
Only send one transport at a time for improved compatibility with some
broken servers. See #537832.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_perform_seek):
Only pause/play in the seek handler when the source was playing.
Fixes#529379.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_handle_dirac_packet):
Fix muxing of Dirac streams if the input already has the format
we need, i.e. is the output of matroskademux.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak),
(qtdemux_video_caps), (qtdemux_audio_caps):
Refactor some raw audio caps building, and handle >16-bit cases.
Fix/replace building caps from a string description.
Original commit message from CVS:
* gst/audiofx/audiowsincband.c:
* gst/audiofx/audiowsinclimit.c:
* gst/cutter/gstcutter.c:
Make author name consistent with others.
Original commit message from CVS:
Based on patch by: Eric Zhang <chao.zhang at access-company dot com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_perform_seek),
(gst_rtspsrc_stream_configure_udp_sink):
Pause the RTSP stream before doing a new play request.
Make sure that adding the udpsinks does not cause the rtspsrc to become
a sink. Fixes#559547.
Original commit message from CVS:
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_pad_free),
(gst_matroska_mux_handle_dirac_packet),
(gst_matroska_mux_write_data):
Implement Dirac muxing into Matroska comforming to the spec, i.e.
put all Dirac packages up to a picture into a Matroska block.
TODO: Implement writing of the ReferenceBlock Matroska elements,
currently the Dirac muxing is only 100% correct if Matroska version 2
is selected for muxing.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Add support for float/double as input and remove the (nowadays)
useless parsing of the depth as we require width==depth.
Original commit message from CVS:
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps):
* gst/rtp/gstrtpmpapay.c:
Narrow down the caps of the mpeg audio pay/depayloaders to only accept
mpeg version 1. Fixes#558427.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_flush),
(gst_rtp_L16_pay_getcaps):
Only put an integral amount of samples in the RTP packet.
Fixes#556641.
Original commit message from CVS:
* gst/rtp/gstrtpchannels.c: (gst_rtp_channels_get_by_index):
* gst/rtp/gstrtpchannels.h:
Add method to get possible channel positions.
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
Don't allow width=32,depth=24 as input. WAV requires that the width
is the next integer multiply of 8 from the depth.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps):
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
(gst_rtp_L16_pay_getcaps):
* gst/rtp/gstrtpchannels.c: (check_channels),
(gst_rtp_channels_get_by_pos), (gst_rtp_channels_get_by_order),
(gst_rtp_channels_create_default):
* gst/rtp/gstrtpchannels.h:
Add mappings for multichannel support. Does not completely just work
because the getcaps function does not yet return the allowed channel
mappings. See #556641.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process):
Check if clock-rate and channels are valid.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_setcaps),
(gst_rtp_ac3_depay_process):
Don't ignore the return value of set_caps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
(gst_rtp_amr_depay_process):
* gst/rtp/gstrtpamrdepay.h:
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set output caps on the buffers, the base class does that for
us.
The subclass will make sure we are negotiated.
* gst/rtp/gstrtpdvdepay.c: (gst_rtp_dv_depay_setcaps),
(gst_rtp_dv_depay_process), (gst_rtp_dv_depay_reset):
* gst/rtp/gstrtpdvdepay.h:
Clean up caps negotiation.
The subclass will make sure we are negotiated.
* gst/rtp/gstrtpg726depay.c: (gst_rtp_g726_depay_setcaps),
(gst_rtp_g726_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpg729depay.c: (gst_rtp_g729_depay_init),
(gst_rtp_g729_depay_setcaps), (gst_rtp_g729_depay_process):
* gst/rtp/gstrtpg729depay.h:
The subclass will make sure we are negotiated.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_setcaps),
(gst_rtp_gsm_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_setcaps):
Clean up caps negotiation.
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_setcaps),
(gst_rtp_h263_depay_process):
Clean up caps negotiation.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_setcaps),
(gst_rtp_h263_pay_flush), (gst_rtp_h263_pay_handle_buffer):
* gst/rtp/gstrtph263pay.h:
Don't ignore the return value of set_outcaps.
Do some more timestamps.
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init),
(gst_rtp_h263p_pay_setcaps), (gst_rtp_h263p_pay_flush),
(gst_rtp_h263p_pay_handle_buffer):
* gst/rtp/gstrtph263ppay.h:
Don't ignore the return value of set_outcaps.
Do some more timestamps.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps),
(gst_rtp_h264_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
Fix possible caps leak.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_setcaps):
Add some more debug info.
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps),
(gst_rtp_ilbc_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_sink_setcaps):
Clean up caps negotiation.
* gst/rtp/gstrtpmp1sdepay.c: (gst_rtp_mp1s_depay_setcaps),
(gst_rtp_mp1s_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
(gst_rtp_mp2t_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpmp4apay.c: (gst_rtp_mp4a_pay_new_caps),
(gst_rtp_mp4a_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_setcaps),
(gst_rtp_mp4g_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize),
(gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
No need to set caps on buffers, subclass does that for us.
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_new_caps),
(gst_rtp_mp4v_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_setcaps),
(gst_rtp_mpa_depay_process):
Clean up caps negotiation.
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_setcaps),
(gst_rtp_mpv_depay_process):
Clean up caps negotiation.
Actually set output caps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpmpvpay.c: (gst_rtp_mpv_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps),
(gst_rtp_pcma_depay_process):
Clean up caps negotiation.
Set output buffer duration because we can.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps),
(gst_rtp_pcmu_depay_process):
Clean up caps negotiation.
Use the marker bit to set the DISCONT flag on outgoing buffers.
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_setcaps):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_init),
(gst_rtp_speex_depay_setcaps), (gst_rtp_speex_depay_process):
Clean up caps negotiation.
Set output caps on the pad and header buffers.
Set duration on output buffers because we can.
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_parse_ident):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_setcaps),
(gst_rtp_sv3v_depay_process):
Clean up caps negotiation.
No need to validate the buffer, the base class does that for us.
No need to set caps out output buffers, subclass does that.
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_setcaps),
(gst_rtp_theora_depay_process):
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_class_init),
(gst_rtp_theora_pay_flush_packet), (encode_base64),
(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
(gst_rtp_theora_pay_handle_buffer):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process):
Don't ignore the return value of setcaps.
No need to validate the buffer, the base class does that for us.
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
Don't ignore the return value of set_outcaps.
* gst/rtp/gstrtpvrawdepay.c: (gst_rtp_vraw_depay_setcaps):
Clean up caps negotiation, don't ignore setcaps return.
* gst/rtp/gstrtpvrawpay.c: (gst_rtp_vraw_pay_setcaps):
Don't ignore the return value of set_outcaps.
