Original commit message from CVS:
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_chain),
(close_library), (open_library),
(gst_real_audio_dec_probe_modules), (gst_real_audio_dec_getcaps),
(gst_real_audio_dec_setcaps), (gst_real_audio_dec_init),
(gst_real_audio_dec_change_state), (gst_real_audio_dec_finalize):
Add raversions we can support on the caps.
Refactor the loading of the real codecs like realvideo so that we can
implement probing.
Probe all supported formats by trying to load the .so files, only report
the versions on the caps that we can actually load.
* gst/real/gstrealvideodec.c: (gst_real_video_dec_chain),
(gst_real_video_dec_getcaps), (gst_real_video_dec_setcaps),
(open_library), (close_library),
(gst_real_video_dec_probe_modules),
(gst_real_video_dec_change_state), (gst_real_video_dec_init),
(gst_real_video_dec_finalize), (gst_real_video_dec_class_init):
* gst/real/gstrealvideodec.h:
Change the loading of the library like the audio decoder.
Probe the supported formats by trying to load the .so files and only
report the versions on the caps that we can actually load.
Original commit message from CVS:
patch by: Sebastian Pölsterl
* gst/mpegtsparse/mpegtspacketizer.c:
Handle character sets in strings coming from DVB SI according
to the DVB SI spec.
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
Fix deadlock when shutting down, use a new lock instead to properly
shutdown.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_add_new_pads),
(gst_deinterleave_src_query):
* gst/interleave/interleave.c: (gst_interleave_src_query_duration),
(gst_interleave_src_query):
Properly implement duration and position queries in bytes format. We
have to take the upstream reply and divide/multiply it by the number
of channels to get the correct result.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_pad_get_property), (gst_interleave_pad_class_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad):
* gst/interleave/interleave.h:
Use an always increasing integer for the number in the name of the
requested sink pads to guarantuee a unique name. Add a "channel"
property to GstInterleavePad to make it possible for applications
to retrieve the channel number in the output for every pad.
Use g_type_register_static_simple() instead of
g_type_register_static() to save some relocations.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_change_state):
Stop GstCollectPads before calling the parent's state change function
when going from PAUSED to READY as we otherwise deadlock.
Fixes bug #536258.
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_nal_bs_init),
(gst_h264_parse_sink_setcaps), (gst_h264_parse_chain_forward),
(gst_h264_parse_queue_buffer), (gst_h264_parse_chain_reverse),
(gst_h264_parse_chain):
* gst/h264parse/gsth264parse.h:
Parse codec_data and use the nalu_size_length field to get the NALU
length in packetized h264.
When queueing a packetized buffer in reverse mode, don't unref the
buffer twice.
Avoid accessing the buffer TIMESTAMP field after we pushed it on
the adaptor.
Original commit message from CVS:
* gst/interleave/interleave.c:
(gst_interleave_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init):
Use new gst_audio_check_channel_positions() function and register
the GstInterleavePad type from a threadsafe context.
Original commit message from CVS:
* gst/interleave/interleave.c: (gst_interleave_pad_get_type),
(gst_interleave_finalize), (gst_audio_check_channel_positions),
(gst_interleave_set_channel_positions),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_set_property), (gst_interleave_get_property),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_sink_setcaps), (gst_interleave_src_query_duration),
(gst_interleave_src_query_latency), (gst_interleave_collected):
* gst/interleave/interleave.h:
Allow setting channel positions via a property and allow using the
channel positions on the input as the channel positions of the output.
Fix some broken logic and memory leaks.
* tests/check/Makefile.am:
* tests/check/elements/interleave.c: (src_handoff_float32),
(sink_handoff_float32), (GST_START_TEST), (interleave_suite):
Add unit tests for checking correct handling of channel positions.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_change_state), (new_payload_found),
(new_ssrc_pad_found):
Break out of callbacks when we are shutting down.
Make sure no state changes can happen when we reconfigure.
Original commit message from CVS:
* gst/interleave/deinterleave.c:
Add another example launch line.
* gst/interleave/interleave.c: (interleave_24),
(gst_interleave_finalize), (gst_interleave_base_init),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_change_state), (__remove_channels),
(__set_channels), (gst_interleave_sink_getcaps),
(gst_interleave_set_process_function),
(gst_interleave_sink_setcaps), (gst_interleave_sink_event),
(gst_interleave_src_query_duration), (gst_interleave_src_query),
(forward_event_func), (forward_event), (gst_interleave_src_event),
(gst_interleave_collected):
* gst/interleave/interleave.h:
Major rewrite of interleave using GstCollectpads. This new version
also supports almost all raw audio formats and has better caps
negotiation. Fixes bug #506594.
Also update docs and add some more examples.
* tests/check/elements/interleave.c: (interleave_chain_func),
(GST_START_TEST), (src_handoff_float32), (sink_handoff_float32),
(interleave_suite):
Add some more extensive unit tests for interleave.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
When checking the seqnum, reset the jitterbuffer if the gap is too big,
we need to do this so that we can better handle a restarted source.
Fix some comments.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
Tweak the skew resync diff.
Use our working seqnum compare function in -base.
Rework the jitterbuffer insert code to make it clearer and more
performant by only retrieving the seqnum of the input buffer once and by
adding some G_LIKELY compiler hints.
Improve debugging for duplicate packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Fix a comment, we don't do skew correction here..
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_set_property):
Propagate the do-lost and latency properties to the jitterbuffers when
they are changed on rtpbin.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps):
* gst/interleave/deinterleave.h:
Don't set a getcaps() function on the src pads as it's not required
and the default getcaps() function returns the correct results for
our src pads.
Complete documentation and add myself to the authors of the element.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_push),
(gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain),
(gst_mpeg4vparse_change_state):
Move some code around to integrate the startcode searching with the
other bits of parsing, avoid a whole bunch of peeks.
Get rid of invalid data that should not happen according to the specs.
Fixes#533559.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_class_init),
(gst_deinterleave_init), (gst_deinterleave_add_new_pads),
(gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property),
(gst_deinterleave_get_property):
* gst/interleave/deinterleave.h:
Add a property to select whether channel positions should be kept on
the mono output buffers or should be dropped.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_finalize),
(gst_deinterleave_init), (gst_deinterleave_sink_event),
(gst_deinterleave_process), (gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Queue events until src pads were added and they can be sent. Otherwise
downstream will never get the first newsegment event.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps),
(gst_deinterleave_getcaps):
Always set the channel positions when gst_audio_get_channel_positions()
returns something, even if they're not set in the caps. This makes
sure that the output channels can be interleaved again correctly
in the mono/stereo cases too.
Don't ask for the peercaps of the current pad in getcaps() as this
might call getcaps() again and deadlock.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.c: (deinterleave_24),
(gst_deinterleave_finalize), (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps),
(gst_deinterleave_set_process_function),
(gst_deinterleave_sink_setcaps), (__remove_channels),
(__set_channels), (gst_deinterleave_getcaps),
(gst_deinterleave_process), (gst_deinterleave_chain),
(gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Add support for all raw audio formats and provide better negotiation
if the caps are changing.
Don't allow changes of the channel positions and set the position of
the corresponding channel on the src pad caps.
General cleanup and smaller bugfixes.
* tests/check/elements/deinterleave.c: (float_buffer_check_probe):
Check the channel positions on the output buffer caps.
Original commit message from CVS:
* docs/Makefile.am:
Don't attempt to build plugin docs when they're disabled.
* gst/bayer/Makefile.am:
Add libgstvideo to the link.
* gst/rtpmanager/Makefile.am:
Fix link order, and move LIBS things to _LIBS
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Simply drop bad RTP packets with a warning instead of just posting an
error and stopping. This is a perfectly recoverable event and we don't
force people to use an rtpbin to filter out bad packets first.
Original commit message from CVS:
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_init):
Set fixed caps on the srcpad after we created the pad...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.h:
* gst/interleave/interleave.h:
* gst/interleave/plugin.h:
Split definitions into separate header files for better documentation
generation.
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps),
(gst_deinterleave_process):
Don't use alloca, allow caps changes as long as the number of channels
does not change, don't use g_warning, return NOT_NEGOTIATED as early
as possible and some other cleanup.
* gst/interleave/interleave.c: (gst_interleave_base_init),
(gst_interleave_class_init):
Do some random cleanup.
* tests/check/Makefile.am:
* tests/check/elements/deinterleave.c: (GST_START_TEST),
(deinterleave_chain_func), (deinterleave_pad_added),
(deinterleave_suite):
Add unit tests for the deinterleave element.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c:
(gst_mpeg4vparse_set_new_caps), (gst_mpeg4vparse_align),
(get_bits), (next_start_code), (gst_mpeg4vparse_handle_vos),
(gst_mpeg4vparse_push), (gst_mpeg4vparse_drain),
(gst_mpeg4vparse_chain), (gst_mpeg4vparse_sink_setcaps),
(gst_mpeg4vparse_sink_event), (gst_mpeg4vparse_src_query),
(gst_mpeg4vparse_set_property), (gst_mpeg4vparse_get_property),
(gst_mpeg4vparse_class_init), (gst_mpeg4vparse_init):
* gst/mpeg4videoparse/mpeg4videoparse.h:
Parse the config data (either outbound or in the stream) to set
width/height, apect ration, framerate in the caps if applicable.
Mark frames as GST_BUFFER_FLAG_DELTA_UNIT when they are not
intra frames
Set the timestamps of outgoing buffers to the buffer in
which the VOP header was found.
Drop incoming data untill configuration is found (by default,
configurable using a property).
Report a 1 frame latency. Fixes#532723.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Avoid waiting for a negative (huge) duration when the last packet has a
lower timestamp than the current packet.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
Make sure to unref the rtpsession returned by gst_pad_get_parent() to
prevent a memory leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
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/inspect/plugin-gstinterlace.xml:
* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
Random doc of the day: the deinterlace element.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Make sure all schedule EIT and non-actual transport stream
EITs are parsed. Also add present-following flag and
actual-transport-stream flag to eit bus message.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
Make sure to unref the caps used by RTPSource to prevent a memory leak.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/rtpsession.c: (source_clock_rate),
(rtp_session_process_bye), (rtp_session_send_bye_locked):
Unlock the session lock when calling one of our callbacks.
Fixes#532011.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/subenc/gstsrtenc.c: (gst_srt_enc_timestamp_to_string):
Declare variables at the beginning of blocks. Fixes compilation with
gcc 2.x and other compilers. Fixes bug #530611.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
Detect SI pids (NIT, SDT, EIT etc.) based on table id and not
by pid number. This allows for example the EPG data from UK's
freesat to be picked up.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose new jitterbuffer property in rtpbin too.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Disable sending out rtp packet lost events by default and make a
property to enabe it. We will likely enable it by default when the base
depayloaders have a default handler for them so that we don't send these
events all through the pipeline for now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Remove private version of a function that is in -base now.
Add src event handler.
Rework the jitterbuffer pushing loop so that it can quickly react to
lost packets and instruct the depayloader of them. This can then be used
to implement error concealment data.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
(create_send_rtcp_src):
Set up some internal links functions for the RTCP and sync pads because
the defaults are really not correct.
Implement a query handler for the RTCP src pad, mostly to correctly
report about the latency.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain):
* gst/rtpmanager/rtpsession.c: (update_arrival_stats),
(rtp_session_process_sr), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Also keep track of the first buffer timestamp together with the first
RTP timestamp as they both are needed to construct the timing of
outgoing packets in the jitterbuffer and are therefore also needed to
manage lip-sync. This fixes lip-sync if the first RTP packets arrive
with a wildly different gap.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script):
Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes
crash caused by a strlen on a NULL string (#527622).
Original commit message from CVS:
* configure.ac:
Bump core/base requirements to released versions to avoid
confusion.
* gst/deinterlace/gstdeinterlace.c: (deinterlace_debug),
(GST_CAT_DEFAULT), (gst_deinterlace_base_init),
(gst_deinterlace_set_caps), (plugin_init):
Add debug category, use _set_element_details_simple and
remove special code path for Y42B to calculate offsets and
strides; libgstvideo knows how to handle this format now.
Original commit message from CVS:
* gst/cdxaparse/Makefile.am:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxastrip.c:
* gst/cdxaparse/gstcdxastrip.h:
* gst/cdxaparse/gstvcdparse.c:
* gst/cdxaparse/gstvcdparse.h:
Port VCD parser (formerly cdxastrip) from 0.8 to 0.10. Doesn't do
anything the 0.8 version didn't do though.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_session):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
Avoid leaking pads in the RTP manager.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Cable delivery subsystem descriptors' frequency's bcd
is measured in 100Hz units so adjust multiplier accordingly.
Original commit message from CVS:
* gst/nsf/Makefile.am:
* gst/nsf/fds_snd.c:
* gst/nsf/mmc5_snd.c:
* gst/nsf/nsf.c:
* gst/nsf/types.h:
* gst/nsf/vrc7_snd.c:
* gst/nsf/vrcvisnd.c:
* gst/nsf/memguard.c:
* gst/nsf/memguard.h:
Remove memguard again and apply hopefully all previously dropped
local patches. Should be really better than the old version now.
Original commit message from CVS:
* gst/nsf/memguard.c: (_my_free):
* gst/nsf/types.h:
Unbreak compilation by disabling memguard and doing some dirty hack
fixes to make it compile on 64bits.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
(gst_input_selector_set_active_pad), (gst_input_selector_switch):
Do g_object_notify() only when not holding the lock to get the property
because otherwise we run into a deadlock with the deep-notify handlers
that are possibly installed.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_event), (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_input_selector_set_active_pad):
Release the selector lock when pad alloc happens on a non selected pad.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_init), (gst_selector_pad_set_property),
(gst_selector_pad_get_property), (gst_selector_pad_event),
(gst_selector_pad_bufferalloc), (gst_selector_pad_chain),
(gst_input_selector_set_active_pad):
Add pad property to configure behaviour of the unselected pad, it can
return OK or NOT_LINKED, based on the use case.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
(gst_selector_pad_get_running_time), (gst_selector_pad_reset),
(gst_selector_pad_event), (gst_selector_pad_bufferalloc),
(gst_input_selector_wait), (gst_selector_pad_chain),
(gst_input_selector_class_init), (gst_input_selector_init),
(gst_input_selector_dispose), (gst_segment_set_start),
(gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_get_property),
(gst_input_selector_get_linked_pad),
(gst_input_selector_is_active_sinkpad),
(gst_input_selector_activate_sinkpad),
(gst_input_selector_request_new_pad),
(gst_input_selector_release_pad),
(gst_input_selector_change_state), (gst_input_selector_block),
(gst_input_selector_switch):
* gst/selector/gstinputselector.h:
Figure out the locking a bit more.
Mark buffers with discont after switching.
Fix initial segment forwarding, make sure to only forward one segment
regardless of what the sequence of buffers/segments is. See #522203.
Improve flushing when blocked.
Return NOT_LINKED when a stream is not selected.
Not API change for the switch signal in the docs.
Fix start/time/accum values of the new segment.
Correctly unlock and flush a blocking selector when going to READY.
Original commit message from CVS:
* gst/freeze/FAQ:
* gst/freeze/Makefile.am:
* gst/freeze/gstfreeze.c:
Add example to source code documentation blob and remove the 3 line
FAQ.
* gst/interleave/interleave.c:
Add a source code documentation blob.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_event),
(gst_selector_pad_bufferalloc), (gst_selector_pad_chain),
(gst_input_selector_class_init),
(gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_push_pending_stop):
Add lots of debugging.
Fix time member in the newsegment event.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_class_init),
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
(gst_selector_pad_event), (gst_input_selector_class_init),
(gst_input_selector_init), (gst_input_selector_set_active_pad),
(gst_input_selector_set_property),
(gst_input_selector_get_property),
(gst_input_selector_request_new_pad),
(gst_input_selector_release_pad),
(gst_input_selector_push_pending_stop),
(gst_input_selector_switch):
* gst/selector/gstinputselector.h:
Various cleanups.
Added tags to the pads.
Select active pad based on the pad object instead of its name.
Fix refcount in set_active_pad.
Add property to get the number of pads.
* gst/selector/gstoutputselector.c:
(gst_output_selector_class_init),
(gst_output_selector_set_property),
(gst_output_selector_get_property):
Various cleanups.
Select the active pad based on the pad object instead of its name.
Fix locking when setting the active pad.
* gst/selector/gstselector-marshal.list:
* tests/check/elements/selector.c: (cleanup_pad),
(selector_set_active_pad), (run_input_selector_buffer_count):
Fixes for pad instead of padname for pad selection.
Original commit message from CVS:
Based on patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
(join_rtcp_thread), (gst_rtp_session_change_state):
Avoid a deadlock when joining the RTCP thread in PAUSED because it might
be blocked downstream. Also avoid spawning multiple rtcp threads.
Fixes#520894.
Original commit message from CVS:
Patch by: Stefan Kost <ensonic@users.sf.net>
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Don't try to reset the clock skew when we have no timestamps.
Fixes#519005.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Add parsing of cable delivery system descriptor.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/mve/gstmvedemux.c: (gst_mve_audio_data),
(gst_mve_demux_get_type):
Fix audio discontinuity that happens when silent chunks are
followed by real data again. Fixes bug #519905.
Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Only send PMTs to program pads that the PMT is for even if
on same pid.
As a by-product, we now no longer hardcode any psi pid numbers.
Also remove pcr stream from old pmt when we apply a new pmt.
Original commit message from CVS:
* gst/selector/gstinputselector.c:
* gst/selector/gstinputselector.h:
Added "select-all" property to make it work like aggregator in 0.8.
* gst/selector/gstoutputselector.c:
Fix resend-latest behavoiur.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/selector.c:
Add unit tests for selector.
Original commit message from CVS:
* configure.ac:
* ext/mpeg2enc/Makefile.am:
* ext/soundtouch/Makefile.am:
* gst/modplug/Makefile.am:
Check for and define ERROR_CXXFLAGS and GST_CXXFLAGS and use them
when building C++ code.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
Ignore streams that did not receive an SR packet when doing
synchronisation. Fixes#516160.
Original commit message from CVS:
* gst/dvdspu/gstdvdspu.c: (gst_dvd_spu_handle_new_spu_buf):
Set n_line_ctrl_i to 0 whenever we free line_ctrl_i. Patch based
on an idea by Jan Schmidt, fixes bug #516436.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.c:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtsparse.c:
Make sure the gstmpegdesc debug lines do not critical
when GST_DEBUG is enabled and also actually output.
Thanks to Alessandro Decina for spotting.
Fixes#516448
Original commit message from CVS:
* ext/xvid/gstxvidenc.c:
* gst/vmnc/vmncdec.c:
* sys/glsink/glimagesink.c:
* sys/glsink/gstgldisplay.c:
Fix some finalize leaks by chaining up to the parent method.
Original commit message from CVS:
* gst/selector/Makefile.am:
Listing the marshal.h in the nodist_HEADERS breaks distcheck, so
let's not do that
* tests/check/Makefile.am:
Disable the crashing cdaudio plugin from the states test so I can make
pre-releases.
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
* gst/multifile/gstmultifilesrc.c:
Use g_file_[sg]et_contents() instead of using stdio functions.
Should be less error prone.
* tests/check/elements/multifile.c:
Create a temporary directory using standard functions instead of
creating a directory in the current dir.
Original commit message from CVS:
* configure.ac:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-xingheader.xml:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/xingmux.c:
* tests/check/elements/xingmux_testdata.h:
Remove the xingmux plugin, as the element has moved into
mpegaudioparse in -ugly.
Original commit message from CVS:
* ext\neon\gstneonhttpsrc.c:
Include unistd.h only if _HAVE_UNISTD_H is defined
* gst\mpegvideoparse\mpegvideoparse.c:
Use G_GUINT64_CONSTANT GLIB macro for constant
* sys\dshowsrcwrapper\gstdshowaudiosrc.c:
* sys\dshowsrcwrapper\gstdshowvideosrc.c:
* sys\dshowdecwrapper\gstdshowaudiodec.c:
* sys\dshowdecwrapper\gstdshowaudiodec.h:
* sys\dshowdecwrapper\gstdshowdecwrapper.c:
* sys\dshowdecwrapper\gstdshowdecwrapper.h:
* sys\dshowdecwrapper\gstdshowvideodec.c
* sys\dshowdecwrapper\gstdshowvideodec.h:
Add a DirectShow decoder wrapper.
* win32\MANIFEST:
Add new win32 files to MANIFEST
* win32\vs6\gst_plugins_bad.dsw:
* win32\vs6\libgstdshow.dsp:
* win32\vs6\libgstdshowdecwrapper.dsp:
* win32\vs6\libgstflv.dsp:
Add new projects to bad workspace
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Parse component descriptor.
* gst/mpegtsparse/mpegtsparse.c:
Add SI pids to every program (but hardcoded currently).
Original commit message from CVS:
* gst/multifile/gstmultifilesink.c:
Add a fixme comment.
* gst/selector/gstoutputselector.c:
Fix same leak as in input-selector.
* tests/icles/output-selector-test.c:
Improve the test.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Add flag to both sdt and nit structures to say
whether the table is for the actual network/ts
or not.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_event):
Don't leak event on pads that are not linked. Fixes#512826.
Original commit message from CVS:
* configure.ac:
Bump core/base requirements to released versions, to avoid confusion.
* gst/deinterlace/Makefile.am:
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_set_caps):
Use the new GstVideoFormat API to get strides, plane offsets etc..
For Y42B we still need to calculate these ourselves, since the lib
in -base doesn't know about this format yet and we can't bump the
requirement to CVS right now. Fix the Y42B stride, offset and size
calculations for odd widths and heights while we're at it though
(to match those in videotestsrc).
Original commit message from CVS:
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Try to get the new clock-rate from the buffer caps when we receive a new
payload type instead of always firing the signal. Fixes#512774.
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_h264_parse_chain_forward):
Try to avoid 'unused variable' compiler warning if debugging is
disabled (not bullet proof, but seems to do for now). (#512654)
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
Don't implement get_unit_size() ourselves, the GstAudioFilter base
class already does this for us.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(create_stream), (payload_type_change), (new_ssrc_pad_found):
Also handle lip-sync when the clock-rate is not provided with caps but
with a signal.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove the fixed clock-rate from the jitterbuffer and extend it so that
a clock-rate can be provided with each buffer instead. Fixes#511686.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Remove old unused variable.
Track pt on input buffers and get the clock-rate when it changes.
Ignore packets with unknown clock-rate. See #511686.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
wrong function. Fixes#511920
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Fix network name descriptor, the length is actually the
descriptor length not stored in the byte after.
Fix bounds checking to be more correct.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Parse and add to relevant bus messages the terrestrial delivery
system descriptor and the logical channel descriptor.
Do bounds checking on data stored in descriptor before use.
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:
* gst/xingheader/gstxingmux.c:
* gst/xingheader/gstxingmux.h:
Add documentation for the xingheader plugin.
* tests/check/elements/xingmux.c: (GST_START_TEST):
Set element state to PLAYING before doing something else.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/xingmux.c: (setup_xingmux),
(cleanup_xingmux), (GST_START_TEST), (xingmux_suite), (main):
* tests/check/elements/xingmux_testdata.h:
Add simple unit test for the xingmux element.
* gst/xingheader/gstxingmux.c: (generate_xing_header),
(gst_xing_mux_finalize), (xing_reset):
Fix a memleak and invalid seek tables with less than 100 MP3 frames.
Original commit message from CVS:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
Parsed the satellite delivery system descriptor and
added into nit's transport structure for delivery
over the bus.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Remove leaks introduced by not freeing g_strndup'd strings.
Fix start_time and duration parsing in EIT.
Original commit message from CVS:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/gstmpegdesc.c:
* gst/mpegtsparse/gstmpegdesc.h:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Added descriptor searching infrastructure from Fluendo TS demuxer.
Add channel name and provider to the sdt structure sent in the
bus message.
Original commit message from CVS:
2008-01-22 Julien Moutte <julien@fluendo.com>
* gst/h264parse/gsth264parse.c: (gst_h264_parse_chain_forward):
Parse NAL units in forward mode to mark delta units flags.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/nuvdemux/gstnuvdemux.c:
One less to do. Its 'nuv' not 'nvu'. As an extra bonus I mention what
it actually is.
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:
Update lists again. Those whole can build ivorbisdec, mythtvsrc,
nvudemux and theoradecexp, please commit the inspect/plugin-xxx.xml.
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-rawparse.xml
* docs/plugins/inspect/plugin-videoparse.xml:
Replace videoparse with rawparse.
* gst/dvdspu/gstdvdspu.h:
Help gtk-doc to recognize the object struct.
Original commit message from CVS:
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
Don't use gtk-doc comment style for non gtk-doc comments.
Make one static function static.
Original commit message from CVS:
Patch by: Gabriel Bouvigne <bouvigne at mp3-tech dot org>
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init),
(gst_deinterlace_init), (gst_deinterlace_set_caps),
(gst_deinterlace_transform_ip), (gst_deinterlace_set_property),
(gst_deinterlace_get_property):
* gst/deinterlace/gstdeinterlace.h:
Provide 4:2:2 support
Also deinterlace chroma planes
Allow to turn on/off deinterlacing
Change of default thresholds, in order to provide acceptable results
with default params. Fixes#511001.
Original commit message from CVS:
* gst/dvdspu/gstdvdspu-render.c: (gst_dvd_spu_render_spu):
* gst/dvdspu/gstdvdspu.c: (dvdspu_debug), (GST_CAT_DEFAULT),
(subpic_sink_factory), (gst_dvd_spu_base_init),
(gst_dvd_spu_class_init), (gst_dvd_spu_init), (gst_dvd_spu_clear),
(gst_dvd_spu_dispose), (gst_dvd_spu_finalize),
(gst_dvd_spu_flush_spu_info), (gst_dvd_spu_buffer_alloc),
(gst_dvd_spu_src_event), (gst_dvd_spu_video_set_caps),
(gst_dvd_spu_video_proxy_getcaps), (gst_dvd_spu_video_event),
(gst_dvd_spu_video_chain), (dvspu_handle_vid_buffer),
(gst_dvd_spu_redraw_still), (gst_dvd_spu_parse_chg_colcon),
(gst_dvd_spu_exec_cmd_blk), (gst_dvd_spu_finish_spu_buf),
(gst_dvd_spu_setup_cmd_blk), (gst_dvd_spu_handle_new_spu_buf),
(gst_dvd_spu_handle_dvd_event), (gst_dvd_spu_advance_spu),
(gst_dvd_spu_check_still_updates), (gst_dvd_spu_subpic_chain),
(gst_dvd_spu_subpic_event), (gst_dvd_spu_change_state),
(gst_dvd_spu_plugin_init):
* gst/dvdspu/gstdvdspu.h: (GST_TYPE_DVD_SPU):
Fix up dvdspu element again after previous namespace mangling:
rename debug category variable to old name, matching that in
dvdspu-render.c, to avoid undefined symbol error when loading
the module; same for the _render function in dvdspu-render.c:
we must use the same name in both .c files; change functions
now called gstgst_* back to gst_* again; and while we're at it,
we may as well canonicalise the namespace properly, namely to
gst_dvd_spu_*.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Add symbols from -unused.txt to the right place.
* gst/dvdspu/gstdvdspu.c:
* gst/dvdspu/gstdvdspu.h:
Coherent namespace usage.
* gst/spectrum/gstspectrum.c:
Fix broken XML fragment in doc snippet even more.
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_push_buffer),
(gst_raw_parse_loop):
Handle framesizes > 4096 with multiple frames per buffer correctly
in pull mode and handle short reads better.
Also put offset and offset_end on outgoing buffers.
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_loop):
Improve handling of unknown or too small upstream sizes in
pull mode.
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_loop),
(gst_raw_parse_handle_seek_push):
Improve debugging a bit and for handling multiple frames per buffer
in pull mode choose the next smallest multiply of framesize below
4096 instead of always handling 1024 frames.
Original commit message from CVS:
* gst/h264parse/gsth264parse.c: (gst_h264_parse_flush_decode),
(gst_h264_parse_queue_buffer), (gst_h264_parse_chain_reverse):
Set timestamps more correctly.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_finalize):
Unparent all bands from the equalizer when finalizing to stop
leaking them.
Original commit message from CVS:
* gst/xingheader/gstxingmux.c: (generate_xing_header):
Bitrate is 4 bits, not 8 so check for 0xe as maximum value instead
of 0xfe.
Original commit message from CVS:
* gst/xingheader/gstxingmux.c: (has_xing_header),
(generate_xing_header), (gst_xing_mux_chain),
(gst_xing_mux_sink_event):
Choose smallest possible frame size for the Xing header, properly
set the timestamp, duration and offset on the outgoing buffers,
only send NEWSEGMENT events in BYTE format downstream and also
drop VBRI headers if already existing.
Original commit message from CVS:
* gst/xingheader/Makefile.am:
* gst/xingheader/gstxingmux.c: (parse_header), (get_xing_offset),
(has_xing_header), (generate_xing_header),
(gst_xing_mux_base_init), (gst_xing_mux_finalize), (xing_reset),
(gst_xing_mux_init), (gst_xing_mux_chain),
(gst_xing_mux_sink_event), (gst_xing_mux_change_state):
* gst/xingheader/gstxingmux.h:
Major cleanup and rewrite of xingmux with less bugs and new features:
- Handles other layers as 3
- Write TOC
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
If we find the caps in the cache, use it to parse the clock-rate instead
of returning an error. Fixes a TODO as found by Youness Alaoui.
Original commit message from CVS:
Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(rtp_session_set_process_rtp_callback),
(rtp_session_set_send_rtp_callback),
(rtp_session_set_send_rtcp_callback),
(rtp_session_set_sync_rtcp_callback),
(rtp_session_set_clock_rate_callback),
(rtp_session_set_reconsider_callback), (source_push_rtp),
(source_clock_rate), (rtp_session_process_bye),
(rtp_session_process_rtcp), (rtp_session_send_bye),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible to use different user_data for each of the callbacks.
Fixes#508587.
Original commit message from CVS:
* gst/mpegvideoparse/mpegpacketiser.c: (mpeg_util_find_start_code):
Small meaningless cleanup.
* gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_flush),
(mpegvideoparse_drain_avail), (gst_mpegvideoparse_chain_forward),
(scan_keyframe), (gst_mpegvideoparse_flush_decode),
(gst_mpegvideoparse_chain_reverse), (gst_mpegvideoparse_chain),
(mpv_parse_sink_event), (gst_mpegvideoparse_change_state):
* gst/mpegvideoparse/mpegvideoparse.h:
Track segment events.
Do the first part of reverse playback by sending data between two
I-frames to the decoder.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink):
Don't set fixed caps, we can basically do everything the upsteam peer
pad can renegotiate to. Fixes#507940.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Don't unref the popped buffer when we don't have ownership.
Fixes#507020.
Original commit message from CVS:
* ext/musicbrainz/gsttrm.c:
Don't emit signiture when going to READY, because it might
not be ready.
* ext/nas/nassink.c:
Remove useless call that sleeps for 5 seconds. Yup, it calls
sleep(1) 5 times. Go NAS.
* gst/librfb/gstrfbsrc.c:
* gst/librfb/rfbdecoder.c:
Initialize our debug categories properly.
* gst/rawparse/gstrawparse.c:
Don't register element details for a non-element. Be much more
rude when subclass doesn't set a pad template (assert!). Don't
unref the pad template; we don't own it.
* gst/videosignal/gstvideoanalyse.c:
Initialize debug category.
* tests/check/Makefile.am:
Ignore nassink element in tests because it has unavoidable
long timeouts.
Original commit message from CVS:
* ext/ladspa/gstladspa.c: (gst_ladspa_get_property):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_show_frame):
* gst/mve/gstmvemux.c: (gst_mve_mux_request_new_pad):
* sys/dvb/dvbbasebin.c: (dvb_base_bin_class_init):
Fix 'xyz may be used uninitialized' compiler warnings caused
by broken g_assert_not_reached() macro in GLib-2.15.x and don't
abort() in any case but properly report the error.
Original commit message from CVS:
* gst/videoparse/Makefile.am:
* gst/videoparse/README:
* gst/videoparse/gstvideoparse.c:
Remove videoparse element, because it was moved to gst/rawparse/
Original commit message from CVS:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_src_event):
Always seek on frame boundaries, will produce nothing useful
otherwise.
Original commit message from CVS:
* configure.ac:
* gst/rawparse/Makefile.am:
* gst/rawparse/README:
* gst/rawparse/gstaudioparse.c: (gst_audio_parse_format_get_type),
(gst_audio_parse_endianness_get_type), (gst_audio_parse_base_init),
(gst_audio_parse_class_init), (gst_audio_parse_init),
(gst_audio_parse_set_property), (gst_audio_parse_get_property),
(gst_audio_parse_update_frame_size), (gst_audio_parse_get_caps):
* gst/rawparse/gstaudioparse.h:
* gst/rawparse/gstrawparse.c: (gst_raw_parse_base_init),
(gst_raw_parse_class_init), (gst_raw_parse_init),
(gst_raw_parse_dispose),
(gst_raw_parse_class_set_src_pad_template),
(gst_raw_parse_class_set_multiple_frames_per_buffer),
(gst_raw_parse_reset), (gst_raw_parse_chain),
(gst_raw_parse_convert), (gst_raw_parse_sink_event),
(gst_raw_parse_src_event), (gst_raw_parse_src_query_type),
(gst_raw_parse_src_query), (gst_raw_parse_set_framesize),
(gst_raw_parse_set_fps), (gst_raw_parse_get_fps),
(gst_raw_parse_is_negotiated):
* gst/rawparse/gstrawparse.h:
* gst/rawparse/gstvideoparse.c: (gst_video_parse_format_get_type),
(gst_video_parse_endianness_get_type), (gst_video_parse_base_init),
(gst_video_parse_class_init), (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property),
(gst_video_parse_format_to_fourcc),
(gst_video_parse_update_frame_size), (gst_video_parse_get_caps):
* gst/rawparse/gstvideoparse.h:
* gst/rawparse/plugin.c: (plugin_init):
Add new plugin rawparse that contains a base class for raw data
parsers and the two elements audioparse and videoparse that can
be used to parse raw audio and video. These are inspired by the
old videoparse element which the new rawparse plugin deprecates.
Original commit message from CVS:
2007-12-18 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch-marshal.list:
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (enum, gst_selector_pad_class_init)
(gst_selector_pad_get_property)
(gst_selector_pad_get_running_time)
(gst_stream_selector_class_init, gst_segment_get_timestamp)
(gst_segment_set_stop, gst_segment_set_start)
(gst_stream_selector_set_active_pad, gst_stream_selector_block)
(gst_stream_selector_push_pending_stop)
(gst_stream_selector_switch): Change so that the signals and
properties deal in running time, not buffer time. Document the
signals more. Change uint64 in API to int64, to reflect what's in
GstSegment.
Original commit message from CVS:
* gst/multifile/gstmultifilesrc.c:
* gst/multifile/gstmultifilesrc.h:
When subsequent files are read, if the file doesn't exist, send
an EOS instead of causing an error.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_selector_pad_chain): Return OK when
a buffer is ignored, not NOT_LINKED. No sense in making a source
element error out; at least fdsrc considers NOT_LINKED to be a
fatal error. Patch 11/12. There is no patch 12/12. Foo.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch-marshal.list:
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (gst_stream_selector_class_init)
(gst_stream_selector_block): Make the block() signal return the
last stop time of the active pad. Patch 10/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_selector_pad_get_property)
(gst_selector_pad_class_init, gst_stream_selector_class_init)
(gst_stream_selector_get_property): Expose 'last-stop-time' as a
pad property, not an element property.
(gst_selector_pad_chain): Mark the last_stop time as timestamp +
duration, not timestamp. Patch 9/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_change_state)
(gst_stream_selector_block, gst_stream_selector_switch): Use the
cond mechanism instead of blocked pads. Patch 8/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelector):
* gst/switch/gstswitch.c (gst_stream_selector_wait)
(gst_selector_pad_chain, gst_stream_selector_init)
(gst_stream_selector_dispose): Add infrastructure for new blocking
mechanism that does not use gst_pad_set_blocked, which does not
work on sink pads. Patch 7/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelector): Add some
state variables.
* gst/switch/gstswitch.c (gst_stream_selector_push_pending_stop)
(gst_selector_pad_chain): Push any pending stop event.
(gst_stream_selector_set_active_pad)
(gst_stream_selector_set_property): Factor out setting the active
pad to a function. Close the segment of the previous active pad if
told to do so via a stop_time != GST_CLOCK_TIME_NONE.
(gst_stream_selector_switch): Implement switch vmethod. Patch 5/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_stream_selector_block): Implement
the block() signal. This implementation will be replaced in future
patches, however. Patch 4/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h (struct _GstStreamSelectorClass):
* gst/switch/gstswitch.c (gst_stream_selector_class_init): Add
`block' and `switch' signals.
* gst/switch/Makefile.am:
* gst/switch/gstswitch-marshal.list: Add foo to generate a
marshaller for the `switch' signal. Patch 2/12.
Original commit message from CVS:
2007-12-17 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.h:
* gst/switch/gstswitch.c: Replace with files from
gststreamselector.[ch], registered as the "switch" plugin, with
"GstSwitch" types. Patch 1/12.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_dispose),
(gst_video_parse_sink_event):
Free the adapter on dispose and correctly reset on newsegment events.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_sink_event),
(gst_video_parse_src_event), (gst_video_parse_src_query):
Improve duration query by first asking upstream and if it can't handle
the query try to get the duration in bytes from upstream and convert.
For seeks, try if upstream handles this already first and do our
conversion to byte format only if it doesn't and if we get a
newsegment event in time format keep it and only do our conversions
if the event has another format.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c:
(gst_video_parse_format_get_type),
(gst_video_parse_endianness_get_type),
(gst_video_parse_class_init), (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property),
(gst_video_parse_format_to_fourcc),
(gst_video_parse_update_block_size), (gst_video_parse_chain),
(gst_video_parse_sink_event):
Add support for video/x-raw-rgb and video/x-raw-gray. Also send
downstream elements downstream, not upstream.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
Hash streams by pid again. Add a linked list inside each
stream with a list of sub_tables. Fix multiple sections
as it was borked with my last commit.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_init),
(gst_video_parse_src_event), (gst_video_parse_src_query_type):
Implement a query type function for the src pad, implement seeking
and use ANY caps for the sink pad as the element doesn't care what
caps the input has and everything is handled via properties.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_convert),
(gst_video_parse_sink_event):
Handle -1 values for the CONVERT query too.
Original commit message from CVS:
* gst/videoparse/gstvideoparse.c: (gst_video_parse_sink_event):
Add YV12 to the pad templates as it is supported too and allow
-1 as stop position for NEWSEGMENT events.
Original commit message from CVS:
* gst/videoparse/Makefile.am:
Add $(GST_PLUGINS_BASE_CFLAGS) to CFLAGS to fix the build.
* gst/videoparse/gstvideoparse.c: (gst_video_parse_init),
(gst_video_parse_set_property), (gst_video_parse_get_property):
Use g_value_[sg]et_enum() for enum properties, g_value_[sg]et_int()
gives a g_critical().
Original commit message from CVS:
* gst/videoparse/README:
* gst/videoparse/gstvideoparse.c:
Add a bunch of features: handle format specification, handle
queries and conversion. Works much like a normal parser now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
(gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
(gst_rtp_ssrc_demux_change_state):
Clean up the dynamic pads when going to READY.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Don't strdup (and thus leak) codec name strings when passing
them to gst_tag_list_add().
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(on_ssrc_sdes):
Post a message when the SDES infor changes for a source.
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
Update some comments.
Original commit message from CVS:
Based on patch by: <mutex at runbox dot com>
* gst/videoparse/gstvideoparse.c: (gst_video_parse_src_query):
Forward the query upstream, the default element event handler does
something different. Fixes#502879.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
* gst/rtpmanager/rtpjitterbuffer.c:
Update comment.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_set_property), (gst_rtp_session_get_property):
Define some GObject properties to set SDES and other configuration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_ssrc_sdes), (rtp_session_set_bandwidth),
(rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
(rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
(rtp_session_get_sdes_string), (obtain_source),
(rtp_session_get_internal_source), (rtp_session_process_sdes),
(rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
(is_rtcp_time):
* gst/rtpmanager/rtpsession.h:
Add signal when new SDES infor has been found for a source.
Create properties for SDES and other info.
Simplify the SDES API.
Add method for getting the internal source object of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_finalize), (rtp_source_set_property),
(rtp_source_get_property), (rtp_source_set_callbacks),
(rtp_source_get_ssrc), (rtp_source_set_as_csrc),
(rtp_source_is_as_csrc), (rtp_source_is_active),
(rtp_source_is_validated), (rtp_source_is_sender),
(rtp_source_received_bye), (rtp_source_get_bye_reason),
(rtp_source_set_sdes), (rtp_source_set_sdes_string),
(rtp_source_get_sdes), (rtp_source_get_sdes_string),
(rtp_source_get_new_sr), (rtp_source_get_new_rb):
* gst/rtpmanager/rtpsource.h:
Add GObject properties for various things.
Don't leak the bye reason.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Fix list of supported and known codecs.
Emit tag with the codec name so it gets properly reported in totem and
other applications.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
The transform() methods are not called in passthrough mode so
there's no need for checking if the element is in passthrough mode.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_transform):
* gst/filter/gstlpwsinc.c: (lpwsinc_transform):
Sync the GObject properties with the controller even in passthrough
mode to get consistent property values.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
A sub table is identified by the pair table_id and
sub_table_identifier, not by pid. So hash with that.
* sys/dvb/dvbbasebin.c:
Make sure initial pids are added properly to filter,
Original commit message from CVS:
2007-12-05 Andy Wingo <wingo@pobox.com>
* gst/switch/gstswitch.c (gst_switch_set_property): Don't push
buffers from app thread when unsetting `queue-buffers', it's
dangerous and the chain function will do it for us anyway.
Original commit message from CVS:
* gst/mpegtsparse/Makefile.am:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Remove signals for pat, pmt, nit, eit, sdt. Replace with bus
messages.
* sys/dvb/dvbbasebin.c:
Instead of attaching to signals, use the bus messages.
Also fix up so the dvbsrc starts only outputting the info tables
like PAT, CAT, NIT, SDT, EIT instead of the whole ts.
Original commit message from CVS:
* gst/speexresample/README:
* gst/speexresample/arch.h:
* gst/speexresample/resample.c: (resampler_basic_direct_single),
(resampler_basic_direct_double),
(resampler_basic_interpolate_single),
(resampler_basic_interpolate_double),
(speex_resampler_process_native), (speex_resampler_process_float),
(speex_resampler_process_int),
(speex_resampler_process_interleaved_float),
(speex_resampler_process_interleaved_int),
(speex_resampler_get_input_latency),
(speex_resampler_get_output_latency):
* gst/speexresample/speex_resampler.h:
Update speex resampler to latest SVN. We're now down to only the
changes noted in README again.
* gst/speexresample/speex_resampler_wrapper.h:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_query):
Adjust to API changes.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Output segment with proper 'stop' value, makes flvdemux 100% compatible
with gnonlin.
Original commit message from CVS:
patch by: Alessandro Decina
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
* gst/mpegtsparse/mpegtsparse.h:
pat-info is now a signal not a GObject property that
gets notified.
pat-info, pmt-info now instead of passing a GObject as
a parameter, pass a GstStructure.
New signals: nit-info, sdt-info, eit-info for DVB SI information
* sys/dvb/camconditionalaccess.c:
* sys/dvb/camconditionalaccess.h:
* sys/dvb/camdevice.c:
* sys/dvb/camdevice.h:
* sys/dvb/camswclient.c:
* sys/dvb/camswclient.h:
* sys/dvb/camutils.c:
* sys/dvb/camutils.h:
Cam code now uses the pmt GstStructure passed from mpegtsparse
signals rather than the GObject.
* sys/dvb/dvbbasebin.c:
Use new signals in mpegtsparse and use GstStructures as per
mpegtsparse's modified API.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Only post the latency message if we have a resampler state already.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_update_state):
Also post GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/speexresample/resample.c: (speex_resampler_get_latency),
(speex_resampler_drain_float), (speex_resampler_drain_int),
(speex_resampler_drain_interleaved_float),
(speex_resampler_drain_interleaved_int):
* gst/speexresample/speex_resampler.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add functions to push the remaining samples and to get the latency
of the resampler. These will get added to Speex SVN in this or a
slightly changed form at some point too and should get merged then
again.
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
(gst_speex_resample_init_state),
(gst_speex_resample_transform_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_fix_output_buffer), (gst_speex_resample_process),
(gst_speex_resample_query), (gst_speex_resample_query_type):
Drop the prepending zeroes and output the remaining samples on EOS.
Also properly implement the latency query for this. speexresample
should be completely ready for production use now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
jitterbuffer can buffer an unlimited amount of time and thus has no
max_latency requirements.
Original commit message from CVS:
* gst/speexresample/README:
Add README explaining where the resampling code was taken from
and which changes were done.
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
(speex_free):
Use g_malloc() and friends instead of malloc() to achieve higher
portability and define the functions inline.
* gst/speexresample/speex_resampler.h:
Add back some useless preprocessor stuff to keep the diff between
our version and the one from the Speex SVN repository lower.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_set_property):
* gst/filter/gstlpwsinc.c: (lpwsinc_set_property):
Post a GST_MESSAGE_LATENCY if the latency changes.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Remove preset iface again. We'll re-add this after its been released
in -good.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
Some small cleanup and addition of a TODO item.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
If the resampler gives less output samples than expected
adjust the output buffer and print a warning.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
Activate preset iface and upload two presets here.
Original commit message from CVS:
Patch by: Wai-Ming Ho <webregbox at yahoo dot co dot uk>
* gst/mpegtsparse/mpegtspacketizer.c: (foreach_stream_clear),
(remove_all), (mpegts_packetizer_clear):
Ensure that the plugin does not crash when the property pat-info is
queried before a PAT is available. It also ensures that the PAT info is
cleared when the changing from PLAYING to READY.
Fixes#487892.
Original commit message from CVS:
* gst/librfb/gstrfbsrc.c:
don't forget to handle the offset's
* gst/librfb/rfbdecoder.c:
* gst/librfb/rfbdecoder.h:
precalculate some many used values
Original commit message from CVS:
patch by: Armando Taffarel Neto <taffarel@solis.coop.br>
* gst/librfb/gstrfbsrc.c:
Set the timestamp for the output buffers
