Original commit message from CVS:
* gst/mxf/mxfaes-bwf.c:
(mxf_metadata_wave_audio_essence_descriptor_parse):
* gst/mxf/mxfaes-bwf.h:
* gst/mxf/mxfdemux.c: (gst_mxf_demux_pull_range),
(gst_mxf_demux_pull_klv_packet),
(gst_mxf_demux_parse_footer_metadata),
(gst_mxf_demux_handle_klv_packet),
(gst_mxf_demux_pull_and_handle_klv_packet), (gst_mxf_demux_chain):
* gst/mxf/mxfmpeg.c: (mxf_metadata_mpeg_video_descriptor_parse):
* gst/mxf/mxfmpeg.h:
* gst/mxf/mxfparse.c: (mxf_timestamp_parse), (mxf_fraction_parse),
(mxf_utf16_to_utf8), (mxf_product_version_parse),
(mxf_partition_pack_parse), (mxf_primer_pack_parse),
(mxf_local_tag_parse), (mxf_metadata_preface_parse),
(mxf_metadata_identification_parse),
(mxf_metadata_content_storage_parse),
(mxf_metadata_essence_container_data_parse),
(mxf_metadata_generic_package_parse), (mxf_metadata_track_parse),
(mxf_metadata_sequence_parse),
(mxf_metadata_structural_component_parse),
(mxf_metadata_generic_descriptor_parse),
(mxf_metadata_file_descriptor_parse),
(mxf_metadata_generic_sound_essence_descriptor_parse),
(mxf_metadata_generic_picture_essence_descriptor_parse),
(mxf_metadata_cdci_picture_essence_descriptor_parse),
(mxf_metadata_multiple_descriptor_parse),
(mxf_metadata_locator_parse):
* gst/mxf/mxfparse.h:
Use guint instead of guint64 or gsize for all buffer sizes and
use correct format strings for them. Only local tag set sizes
are still guint16 as they can't be larger.
Only allow KLV packets of sizes below 1<<32 as GStreamer only uses
guint for buffer sizes. The MXF standard allows packet sizes up
to 1<<64.
Original commit message from CVS:
* gst/dccp/gstdccp.c: (gst_dccp_socket_write):
Use G_GSIZE_FORMAT instead of "%u" for a size_t variable in
the format string to prevent a compiler warning.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Release the right pads on rtpbin. Fixes#561752.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
(gst_speex_resample_get_property):
Add a "filter-length" property that maps to the quality values
for compatibilty with audioresample.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain_rtcp):
Initialize return value to fix compiler warning about uninitialized
variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream), (free_stream),
(new_ssrc_pad_found):
Remove internal sync pad, use signals instead to get lip-sync
notifications.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
(remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
(gst_rtp_jitter_buffer_release_pad),
(gst_rtp_jitter_buffer_sink_rtcp_event),
(gst_rtp_jitter_buffer_chain_rtcp),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Make it possible to send SR packets to the jitterbuffer.
Check if the SR timestamps are valid by comparing them to the RTP
timestamps.
Signal the SR packet and the timing information to listeners.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
Remove some unused code.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last seen RTP timestamp so that we can filter out
invalid SR packets.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Fix GST_DEBUG call to only have as many arguments as required
by the format string. Fixes a compiler warning.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
Do not try to keep track of the clock-rate ourselves but simply get the
value from the jitterbuffer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add some debug info.
Pass the clock-rate to the jitterbuffer.
Also pass the clock-rate along with the rtp timestamp when getting the
sync parameters.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix some debug.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of clock-rate changes and return the clock-rate together with
the rtp timestamps used for sync.
Don't try to construct timestamps when we have no base_time.
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Request a new clock-rate when the payload type changes.
Reset the jitter calculation when the clock-rate changes.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Guard against a NULL dereference I somehow encountered -
with a FLUSH_STOP arriving either before basetransform _start(),
or after _stop().
* gst/typefind/gsttypefindfunctions.c:
Make sure we never jump backwards when typefinding corrupt mov files.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
Small cleanups and some more debug info.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
Also configure the next expected output seqnum when we get a seqnum-base
on the caps.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_class_init), (gst_deinterlace2_init),
(gst_deinterlace2_set_property), (gst_deinterlace2_get_property):
Bring properties into this century.
Original commit message from CVS:
* gst/mpegdemux/gstmpegtsdemux.c:
Make private section pads have a caps set so they are not tried
to be linked in parse_launch for example.
Original commit message from CVS:
patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
No need to reclaculate flush in this case.
Fixes some bad decode errors introduced.
Original commit message from CVS:
patch by: Sebastian Pölsterl
* gst/mpegdemux/gstmpegdesc.c:
Length should be a guint8 not a gint.
* gst/mpegdemux/mpegtspacketizer.c:
Convert text to utf8 for each descriptor separately and not
concatenate them first and convert after.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
Add TODO at the top of the file for enabling SSE/ARM specific
optimizations and choosing the fastest implementation at runtime.
Add g_assert_not_reached() at two places that should really never
be reached.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_check_discont):
Fix format string and arguments.
* gst/speexresample/resample_sse.h:
Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
(_benchmark_int_int), (_benchmark_integer_resampling),
(plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add missing headers to Makefile.am.
Update copyright, years and my mail address.
Benchmark the integer resampling implementation against the
float implementation and use the faster one for 8/16 bit integer
input. On most recent systems the floating point version is faster.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
The length for the buffer conversion function is the number of
audio frames, i.e. we need to multiply it by the number of channels
to get the number of values. Also spotted by the unit test after
running in valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_funcs),
(gst_speex_resample_transform_size),
(gst_speex_resample_convert_buffer),
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add support for int8, int24 and int32 input by converting internally
to/from int16 or double.
Original commit message from CVS:
* gst/audiobuffer/Makefile.am:
* gst/audiobuffer/gstaudioringbuffer.c:
(gst_int_ring_buffer_acquire), (gst_int_ring_buffer_release),
(gst_int_ring_buffer_start), (gst_int_ring_buffer_base_init),
(gst_int_ring_buffer_class_init), (gst_int_ring_buffer_init),
(gst_int_ring_buffer_new), (gst_audio_ringbuffer_get_type),
(gst_audio_ringbuffer_class_init), (gst_audio_ringbuffer_init),
(gst_audio_ringbuffer_finalize), (gst_audio_ringbuffer_getcaps),
(gst_audio_ringbuffer_setcaps), (gst_audio_ringbuffer_bufferalloc),
(gst_audio_ringbuffer_handle_sink_event),
(gst_audio_ringbuffer_render), (gst_audio_ringbuffer_chain),
(gst_audio_ringbuffer_handle_src_event),
(gst_audio_ringbuffer_handle_src_query),
(gst_audio_ringbuffer_get_range),
(gst_audio_ringbuffer_src_checkgetrange_function),
(gst_audio_ringbuffer_sink_activate_push),
(gst_audio_ringbuffer_src_activate_push),
(gst_audio_ringbuffer_src_activate_pull),
(gst_audio_ringbuffer_change_state),
(gst_audio_ringbuffer_set_property),
(gst_audio_ringbuffer_get_property), (plugin_init):
Add first version of an audioringbuffer element that can be inserted in
the pipeline to convert push-based upstream into a pull-based
downstream.
Original commit message from CVS:
Patch by: Robin Stocker <robin at nibor dot org>
* gst/real/gstrealvideodec.c: (gst_real_video_dec_setcaps):
A RealVideo video inside a container (for example MKV) should use the
PAR which is specified on the sinkpad caps. Fixes#558416.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_loop), (gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull), (gst_flv_demux_sink_event),
(gst_flv_demux_dispose), (gst_flv_demux_init):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video), (gst_flv_parse_tag_timestamp):
Put the GstSegment directly into the instance struct instead of
allocating and free'ing it again.
Push tags already if only one pad was added, no need to wait for
the second one.
When generating our index set has_video and has_audio if we find
video or audio in case the FLV header has incorrect data.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
(gst_speex_resample_process):
Fixate to the nearest supported rate instead of the first one.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (audioresample_fixate_caps):
Fixate the rate to the nearest supported rate instead of
the first one. Fixes bug #549510.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header),
(gst_flv_demux_create_index):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type),
(gst_flv_parse_header):
* gst/flv/gstflvparse.h:
Don't memcpy() all data we want to push downstream, instead just
create subbuffers and push them downstream.
Fix some minor memory leaks.
Original commit message from CVS:
* gst/flv/Makefile.am:
Fix (non-critical) syntax error and add all required CFLAGS and LIBS.
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_metadata_item), (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_timestamp), (gst_flv_parse_tag_type):
Rewrite the script tag parsing to make sure we don't try to read
more data than we have. Also use GST_READ_UINT24_BE directly and
fix some minor memory leaks.
This should make all crashes on fuzzed FLV files disappear.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (FLV_GET_STRING),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_type), (gst_flv_parse_header):
Properly check everywhere that we have enough data to parse and
don't read outside the allocated memory region.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
If the caps change during playback and negotiation fails error out
instead of trying to continue.
Original commit message from CVS:
* gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_request_new_pad), (gst_flv_mux_write_buffer),
(gst_flv_mux_collected):
* gst/flv/gstflvmux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate):
Add support for Speex audio and allow buffers without valid
timestamp in the muxer.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_loop),
(gst_flv_demux_find_offset), (gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull):
Don't post an error message on the bus if sending EOS downstream
didn't work. Fixes bug #550454.
Fix seek event handling to look at the flags of the seek event
instead of assuming some random flags, don't send segment-start
messages when operating in push mode and push seek events upstream
if we couldn't handle them.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_create_index),
(gst_flv_demux_loop):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_script),
(gst_flv_parse_tag_audio), (gst_flv_parse_tag_video),
(gst_flv_parse_tag_timestamp):
* gst/flv/gstflvparse.h:
In pull mode we create our own index before doing anything else
and don't use the index provided by some files (which are more than
often incorrect and cause failed seeks).
For push mode we still use the index provided by the file and extend it
while doing the playback.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_push_src_event),
(gst_flv_demux_loop), (gst_flv_demux_handle_seek_pull),
(gst_flv_demux_sink_event):
Instead of using gst_pad_event_default() use a small
gst_pad_push_event() wrapper that only does what we want and is much
more simple.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_change_state),
(gst_flv_demux_set_index), (gst_flv_demux_init):
* gst/flv/gstflvdemux.h:
If our index was created by the element and not provided from the
outside we should destroy it when starting a new stream to get
all old entries removed.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range):
Improve debugging a bit when pulling a buffer from upstream fails.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_cleanup),
(gst_flv_demux_handle_seek_pull), (gst_flv_demux_dispose):
* gst/flv/gstflvdemux.h:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Close the currently playing segment from the streaming thread
instead of the thread where the seek event is handled.
Original commit message from CVS:
Patch by: David Härdeman <david at hardeman dot nu>
* gst/mpegdemux/mpegtspacketizer.c: (mpegts_packetizer_parse_nit):
Add support for the frequency list descriptor, which provides
additional frequencies that should be scanned by a DVB application.
Fixes bug #557814.
Original commit message from CVS:
Patch by: vanista <vanista at gmail dot com>
* gst/mpegtsmux/mpegtsmux.c: (mpegtsmux_choose_best_stream):
Fix EOS logic by correctly popping the collect pad buffers only
when we've chosen to use them instead of popping them always and
storing them in a private queue.
Before the pipeline would deadlock if all pads go EOS at the same
time. Fixes bug #557763.
Original commit message from CVS:
Patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
Properly handle some resync cases in the optimised
buffering strategy.
Original commit message from CVS:
* gst/flv/gstflvmux.c: (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_write_buffer):
Don't set video_codec to the value that actually should go
into audio codec, otherwise we create invalid files.
Fixes bug #556564.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix problem with using the output seqnum counter to check for input
seqnum discontinuities.
Improve gap detection and recovery, reset and flush the jitterbuffer on
seqnum restart. Fixes#556520.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert):
Fix wrong G_LIKELY.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
Install event handler on the rtcp_src pad, make LATENCY event return
TRUE.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_send_data):
Make sure the mpegpsdemux element creates valid newsegment events.
Fixes#556428
Original commit message from CVS:
patch by: Sebastian Pölsterl
* gst/mpegdemux/mpegtspacketizer.c:
Fixes segfault in get_encoding_and_convert.
Fixes#556482
Original commit message from CVS:
patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
Fixes a segfault in the adaptation buffer size strategy.
Fixes#556440
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_event),
(gst_input_selector_query):
Gracefully handle the cases when we dont' have otherpad.
Fixes#556430
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Use gst_pad_alloc_buffer_and_set_caps() to make sure we get
a buffer with caps that we can work with (i.e. the pad's caps).
Add non-keyframe video frames to the index too but without the
keyframe flag.
Add audio frames to the index only if we have no video stream.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_audio),
(gst_flv_parse_tag_video):
Create pads from the pad templates, use fixed caps on them
and only activate them after the caps are set.
Original commit message from CVS:
2008-10-10 Jan Schmidt <jan.schmidt@sun.com>
* gst/flacparse/gstbaseparse.c (gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations):
Fix compiler warning on OS/X about parameters not matching
the debug format string.
Original commit message from CVS:
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_class_init):
Fix unused variable compiler warning when not building
X86 assembly.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_loop):
* gst/flv/gstflvparse.c: (gst_flv_parse_tag_timestamp):
* gst/flv/gstflvparse.h:
Get an approximate duration of the file by looking at the timestamp
of the last tag in pull mode. If we get (maybe better) duration from
metadata later we'll use that instead.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_pull_range),
(gst_flv_demux_pull_tag), (gst_flv_demux_pull_header):
Refactor _pull_range() logic with checks into a seperate function
to make things a bit more readable.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain),
(gst_flv_demux_base_init):
Use gst_element_class_set_details_simple().
If we get GST_FLOW_NOT_LINKED in the parse loop but at least
one of the pads is linked continue the loop.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate),
(gst_flv_parse_tag_audio), (gst_flv_parse_video_negotiate):
Correct caps for video codec id 5: It's On2 VP6 with alpha channel
which needs a different decoder and has different caps.
Add support for audio codec id 14, which is MP3 with 8kHz sampling
rate.
Fix endianness and signedness for raw audio codec ids.
Add support for alaw and mulaw audio.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_chain):
Go out of the parse loop as soon as we get an error instead
of parsing until the GstAdapter is empty.
Add some explanations about the header and tag size.
Don't print synchronizing message if everything is fine.
Original commit message from CVS:
* gst/flv/Makefile.am:
* gst/flv/gstflvdemux.c: (plugin_init):
* gst/flv/gstflvmux.c: (gst_flv_mux_base_init),
(gst_flv_mux_class_init), (gst_flv_mux_init),
(gst_flv_mux_finalize), (gst_flv_mux_reset),
(gst_flv_mux_handle_src_event), (gst_flv_mux_handle_sink_event),
(gst_flv_mux_video_pad_setcaps), (gst_flv_mux_audio_pad_setcaps),
(gst_flv_mux_request_new_pad), (gst_flv_mux_release_pad),
(gst_flv_mux_write_header), (gst_flv_mux_write_buffer),
(gst_flv_mux_collected), (gst_flv_mux_change_state):
* gst/flv/gstflvmux.h:
Add first version of a FLV muxer. The only missing feature is writing
of stream metadata.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_send_data),
(gst_flups_demux_parse_pack_start):
Prevent a division by zero if last mux rate was zero.
If we're going to send a NEWSEGMENT event but the segment start
and the current buffer timestamp differ by more than a second we
will start the NEWSEGMENT at the buffer timestamp.
This fixes playback of the tv2-1_25.mpg file, which has 0 as first SCR
but the first PTS are around 1 hour and 40 minutes.
Fixes bug #553755.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin-marshal.list:
Add marshaller for new action signal.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add action signal to retrieve the internal RTPSession object.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_get_property), (gst_rtp_session_release_pad):
Add property to access the internal RTPSession object.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(check_collision):
* gst/rtpmanager/rtpsession.h:
Add action signal to retrieve an RTPSource object by SSRC.
See #555396.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps):
Only update the seqnum-base when it was not already configured for the
streams.
Original commit message from CVS:
* gst/mpegtsparse/mpegtsparse.c:
Actually copy the structure passed in when assigning it because
it gets freed straight after the function call.
Re: pat_info and pmt_info GstStructures.
Original commit message from CVS:
Patch by: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
Fix wrong firing of critical introduced by previous optimisation.
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_finalize),
(gst_base_parse_class_init), (gst_base_parse_push_buffer),
(gst_base_parse_change_state), (gst_base_parse_set_index),
(gst_base_parse_get_index):
Add support for GstIndex.
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_class_init),
(gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations),
(gst_base_parse_convert), (gst_base_parse_frame_in_segment):
* gst/flacparse/gstbaseparse.h:
Provide a vfunc for the subclass to decide whether a frame is inside
the segment or not and add a default implementation.
Fix approximate bitrate calculations.
Original commit message from CVS:
* gst/flacparse/gstbaseparse.c: (gst_base_parse_class_init),
(gst_base_parse_init), (gst_base_parse_push_buffer),
(gst_base_parse_update_upstream_durations), (gst_base_parse_chain),
(gst_base_parse_loop), (gst_base_parse_activate),
(gst_base_parse_convert), (gst_base_parse_query):
Approximate the average bitrate, duration and size if possible
and add a default conversion function which uses this for
time<->byte conversions.
* gst/flacparse/gstflacparse.c: (gst_flac_parse_get_frame_size):
Fix parsing if upstream gives -1 as duration.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
(on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
Ref the rtpsource object before we release the session lock when we emit
the signals.
Original commit message from CVS:
Patch from: Josep Torra
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
Use a preallocated buffer per stream for PES packets sent on src pads.
Adaptively adjust buffer size appropriately.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
Original commit message from CVS:
* gst/aiffparse/aiffparse.c:
Support chunks in AIFF in any order in pull mode, and any order so
long as we get COMM before the actual data (SSND) in push mode.
Fixes playback of AIFC files.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_reset),
(gst_input_selector_reset), (gst_input_selector_change_state):
Reset the selector state when going to READY.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
Original commit message from CVS:
* gst/mpegdemux/gstmpegdemux.c: (gst_flups_demux_parse_pack_start):
* gst/mpegdemux/gstmpegtsdemux.c: (gst_fluts_demux_data_cb):
Fix build on macosx.
Original commit message from CVS:
* ext/resindvd/plugin.c: (plugin_init):
* ext/resindvd/resindvdsrc.c:
* ext/twolame/gsttwolame.c: (plugin_init):
* gst/aiffparse/aiffparse.c: (plugin_init):
Enable/fix up translations for these plugins.
* po/LINGUAS:
Add 'ca' to LINGUAS.
* po/POTFILES.in:
* po/POTFILES.skip:
Add more files for translation and more files which tools
should skip.
Original commit message from CVS:
* gst/mpegtsmux/mpegtsmux_aac.c: (mpegtsmux_prepare_aac):
Allocate a fixed size buffer on the stack instead of using malloc().
* gst/mpegtsmux/tsmux/tsmux.c: (tsmux_new), (tsmux_free),
(tsmux_program_new), (tsmux_program_free):
* gst/mpegtsmux/tsmux/tsmuxstream.c: (tsmux_stream_new),
(tsmux_stream_free), (tsmux_stream_consume),
(tsmux_stream_add_data):
Use GSlice.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_init),
(gst_input_selector_event), (gst_input_selector_query):
Reuse the get_linked_pads for both source and sinkpads because they are
the same.
Implement a custum event handler and get the internally linked pad
directly instead of relying on the default (slower) implementation.
Original commit message from CVS:
* gst/dccp/gstdccp.c:
* gst/dccp/gstdccpclientsrc.c:
Fix compilation on Solaris by including filio.h as needed.
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
Fix compilation with Forte - apparently it hates concatenating a
macro argument that starts with an underscore??
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
(gst_rtp_session_event_send_rtp_sink):
Send EOS when the session object instructs us to.
* gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible for the session manager to instruct us to send EOS. We
currently will EOS when the session is a sender and when the sender part
goes EOS. This is not entirely correct behaviour because the session
could still participate as a receiver.
Fixes#549409.
Original commit message from CVS:
* gst/aiffparse/aiffparse.c:
Read size of chunks preceeding the audio data with the
correct endianness. Fixes playback of some files.
Fixes#538500
Original commit message from CVS:
* configure.ac:
* gst/aiffparse/Makefile.am:
* gst/aiffparse/aiffparse.c:
* gst/aiffparse/aiffparse.h:
Add an AIFF parsing element, heavily based on wavparse.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_input_selector_init),
(gst_input_selector_query):
Implement the LATENCY query in a better way by taking the latency of all
sinkpads and taking the min/max instead of just taking a random pad.
Original commit message from CVS:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
* gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
Unroll the loop to handle two bytes at once. This should give
a small speedup and makes it possible to handle chroma and luma
different which is needed later.
Original commit message from CVS:
* gst/dccp/gstdccpserversink.c:
* gst/dccp/gstdccpserversink.h:
Don't put globals only used by one '.c' file in a header !
Declare it as static, fixes build on macosx.
Original commit message from CVS:
* gst/dccp/gstdccp.c: (gst_dccp_read_buffer),
(gst_dccp_send_buffer), (gst_dccp_set_sock_windowsize):
size_t's size varies by platform/architecture. Use glib convenience
macro instead. Fixes build on macosx.
Remove ending '\n' in debug statements.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_method_class_init):
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_class_init):
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
* gst/deinterlace2/tvtime/tomsmocomp/StrangeBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
* gst/deinterlace2/tvtime/tomsmocomp/WierdBob.inc:
* gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
First part of the C implementation of the tomsmocomp deinterlacing
algorithm. This only supports search-effort=0 currently, is painfully
slow and needs some cleanup later when all search-effort settings
are implemented in C.
Original commit message from CVS:
* gst/pcapparse/gstpcapparse.c:
* sys/winscreencap/gstdx9screencapsrc.c:
* sys/winscreencap/gstgdiscreencapsrc.c:
Added documentation blobs. Thanks to Stefan for noticing!
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Print the pad-name in debug log.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
Use "-" instead of "_" in property names. Can we call them just
"device" like everywhere else?
Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Make the buffer metadata writable before inserting it in the
jitterbuffer because the jitterbuffer will modify the timestamps.
* gst/rtpmanager/rtpjitterbuffer.c:
Update method comment about requiring writable metadata on buffers.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_rtcp):
Make the RTCP buffer metadata writable because we want to modify the
metadata.
Fixes#546312.
Original commit message from CVS:
* gst/selector/gstinputselector.c: (gst_selector_pad_bufferalloc),
(gst_selector_pad_chain), (gst_input_selector_getcaps),
(gst_input_selector_activate_sinkpad):
Move the select-all logic into the activation of the currently selected
pad. We want to remember the last pad with activity in select-all mode.
Fix the getcaps function, we can produce the union of the upstream caps
in select-all mode, not the intersection like proxy_getcaps() does.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Fix debug by logging the right seqnum.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (get_pt_map):
Release lock before emitting the request-pt-map signal.
Fixes#543480.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_simple_method_interpolate_scanline),
(gst_deinterlace_simple_method_copy_scanline),
(gst_deinterlace_simple_method_deinterlace_frame):
* gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy):
* gst/deinterlace2/tvtime/greedyh.c:
(deinterlace_frame_di_greedyh):
* gst/deinterlace2/tvtime/scalerbob.c:
(deinterlace_scanline_scaler_bob):
* gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy):
* gst/deinterlace2/tvtime/weave.c: (deinterlace_scanline_weave),
(copy_scanline):
* gst/deinterlace2/tvtime/weavebff.c: (deinterlace_scanline_weave),
(copy_scanline):
* gst/deinterlace2/tvtime/weavetff.c: (deinterlace_scanline_weave),
(copy_scanline):
Use oil_memcpy() instead of memcpy() as it's faster for the sizes that
are usually used here.
Original commit message from CVS:
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
(deinterlace_line_mmx), (gst_deinterlace_method_vfir_class_init):
Implement the VFIR deinterlacing method as simple method.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_simple_method_interpolate_scanline),
(gst_deinterlace_simple_method_copy_scanline),
(gst_deinterlace_simple_method_deinterlace_frame),
(gst_deinterlace_simple_method_class_init),
(gst_deinterlace_simple_method_init):
* gst/deinterlace2/gstdeinterlace2.h:
Add a GstDeinterlaceSimpleMethod subclass of GstDeinterlaceMethod that
can be used by simple deinterlacing methods. They only have to provide
a function for interpolating a scanline or copying a scanline.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_chain):
Respect the latency of the deinterlacing algorithm for the timestamps
of every buffer.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.asm:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
Add the MMX registers to the clobbered registers only if __MMX__ is
defined.
Original commit message from CVS:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method),
(gst_deinterlace2_class_init):
Enable tomsmocomp again as the C port will be ready for the next
release.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init),
(gst_greatest_common_divisor), (gst_fraction_double),
(gst_deinterlace2_getcaps), (gst_deinterlace2_setcaps):
Don't use proxy_getcaps() but implement our own getcaps() function
that doubles/halfs the framerate if all fields should be sent out.
Original commit message from CVS:
* gst/festival/gstfestival.c:
Guard unistd.h with HAVE_UNISTD_H
* gst/modplug/libmodplug/load_it.cpp:
* gst/modplug/libmodplug/sndfile.cpp:
* gst/modplug/libmodplug/sndfile.h:
* gst/modplug/libmodplug/stdafx.h:
Support for compiling with MSVC: use _MSC_VER for detecting MSVC instead
of MSC_VER. Make CanPackSamples take the type it's passed. Change scope
of a variable in load_it.cpp to MSVC's scoping rules.
* gst/sdp/gstsdpdemux.c:
Guard unistd.h with HAVE_UNISTD_H
Fixes#544457.
Original commit message from CVS:
* gst/modplug/libmodplug/Makefile.am:
Use GST_CXXFLAGS instead of GST_CFLAGS for CXXFLAGS. GST_CFLAGS
can contain compiler parameters that are invalid for C++.
Fixes bug #543860.
Original commit message from CVS:
* configure.ac:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_methods_get_type), (gst_deinterlace2_set_method),
(gst_deinterlace2_class_init), (gst_deinterlace2_init):
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/greedy.c:
(gst_deinterlace_method_greedy_l_class_init):
* gst/deinterlace2/tvtime/greedyh.c:
(gst_deinterlace_method_greedy_h_class_init):
* gst/deinterlace2/tvtime/vfir.c:
(gst_deinterlace_method_vfir_class_init):
Disable the tomsmocomp algorithm for this release as it's buggy
and has no C implementation yet.
Build the deinterlace2 plugin on all architectures but still mark it
as experimental.
Build the x86 inline assembly only if GCC inline assembly is supported
and only on x86 or amd64. Fixes bug #543286.
Original commit message from CVS:
* ext/resindvd/gstmpegdemux.c:
Remove whitespace line.
* ext/resindvd/resindvdbin.c:
* ext/resindvd/resindvdbin.h:
Make it so that audio/video pads aren't added to the bin
until after data flow starts.
* ext/resindvd/resin-play:
Move video buffer queue outside resindvdbin
* ext/resindvd/resindvdsrc.c:
Check that the nav_clock_id didn't already get unscheduled.
* gst/dvdspu/gstdvdspu.c:
Remove assert that sometimes triggers erroneously.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.asm:
Always use the C implementation if width is not a multiple of 4. The
assembly optimized version only handle this and calling the C
implementation for the remaining part doesn't work because it needs
previous calculations.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.asm:
* gst/deinterlace2/tvtime/greedyh.c:
* gst/deinterlace2/tvtime/greedyhmacros.h:
Some cleanup, use 3DNOW instead of TDNOW in macros.
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_class_init):
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/tomsmocomp/tomsmocompmacros.h:
The SSE method in fact only needs MMXEXT, declare it as such.
Original commit message from CVS:
* gst/festival/gstfestival.c:
Add a note to the docs, that festival need to be running. Also log an
error it it is not. Fixes#541327
Original commit message from CVS:
* ext/spc/gstspc.c: (spc_setup):
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopTop.inc:
Don't use declarations after statements in the remaining code.
Original commit message from CVS:
patch by: Alessandro Decina
* gst/mpegtsparse/mpegtsparse.c:
Fix memory leak by unreffing structures when not needed.
Fixes#539292.
Original commit message from CVS:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll2.inc:
Mark internal processing functions as static inline for quite some
speedup as they're used only once and need to get many local variables
passed as parameter.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_method_get_latency),
(gst_deinterlace2_set_method), (gst_deinterlace2_class_init),
(gst_deinterlace2_push_history), (gst_deinterlace2_chain),
(gst_deinterlace2_setcaps), (gst_deinterlace2_src_query):
* gst/deinterlace2/gstdeinterlace2.h:
Include latency of the method in the returned latency.
Fix outputting of all fields, i.e. doubling of the framerate.
Original commit message from CVS:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace_method_class_init), (gst_deinterlace_method_init),
(gst_deinterlace_method_deinterlace_frame),
(gst_deinterlace_method_get_fields_required),
(gst_deinterlace2_methods_get_type), (_do_init),
(gst_deinterlace2_set_method), (gst_deinterlace2_class_init),
(gst_deinterlace2_child_proxy_get_child_by_index),
(gst_deinterlace2_child_proxy_get_children_count),
(gst_deinterlace2_child_proxy_interface_init),
(gst_deinterlace2_init), (gst_deinterlace2_finalize),
(gst_deinterlace2_chain), (gst_deinterlace2_src_query):
* gst/deinterlace2/gstdeinterlace2.h:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_c),
(deinterlace_greedy_packed422_scanline_mmx),
(deinterlace_greedy_packed422_scanline_mmxext),
(deinterlace_frame_di_greedy),
(gst_deinterlace_method_greedy_l_set_property),
(gst_deinterlace_method_greedy_l_get_property),
(gst_deinterlace_method_greedy_l_class_init),
(gst_deinterlace_method_greedy_l_init):
* gst/deinterlace2/tvtime/greedyh.asm:
* gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C),
(deinterlace_frame_di_greedyh),
(gst_deinterlace_method_greedy_h_set_property),
(gst_deinterlace_method_greedy_h_get_property),
(gst_deinterlace_method_greedy_h_class_init),
(gst_deinterlace_method_greedy_h_init):
* gst/deinterlace2/tvtime/greedyh.h:
* gst/deinterlace2/tvtime/plugins.h:
* gst/deinterlace2/tvtime/tomsmocomp.c:
(gst_deinterlace_method_tomsmocomp_set_property),
(gst_deinterlace_method_tomsmocomp_get_property),
(gst_deinterlace_method_tomsmocomp_class_init),
(gst_deinterlace_method_tomsmocomp_init):
* gst/deinterlace2/tvtime/tomsmocomp.h:
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir),
(gst_deinterlace_method_vfir_class_init),
(gst_deinterlace_method_vfir_init):
Use a GstObject subtype for the deinterlacing methods and export
the different settings for each deinterlacing method via GObject
properties.
Implement GstChildProxy interface to allow access to the used
deinterlacing method and to allow adjusting the different settings.
Move global variables of the tomsmocomp deinterlacing method into
function local variables to make it possible to use this deinterlacing
method from different instances.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.asm:
Support widths that are not a multiply of 4 when using the assembly
optimized greedyh implementations.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.c:
(deinterlace_frame_di_greedyh):
Only build the assembly optimized implementations on x86.
Original commit message from CVS:
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
* configure.ac:
* gst/festival/Makefile.am:
* gst/festival/gstfestival.c:
Fix built of the festival plugin with mingw32 by linking to ws2_32.dll
and including winsock headers. Fixes bug #541522.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_send_rtp):
* gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp):
Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
pipeline is running normally.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_send_bye_locked), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_report_blocks), (session_cleanup),
(is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Do not mix the use of g_get_current_time() with gst_clock_get_time().
Original commit message from CVS:
* gst/mpegvideoparse/mpegvideoparse.c:
* gst/mpegvideoparse/mpegvideoparse.h:
Queue new-segment events and send them after caps have been
determined and set, so that the decoder will have been auto-plugged.
Original commit message from CVS:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/tvtime/tomsmocomp.c: (tomsmocomp_init),
(tomsmocomp_filter_mmx), (tomsmocomp_filter_3dnow),
(tomsmocomp_filter_sse), (deinterlace_frame_di_tomsmocomp):
* gst/deinterlace2/tvtime/tomsmocomp.h:
Remove useless file and mark everything possible as static.
* gst/deinterlace2/tvtime/greedy.c:
* gst/deinterlace2/tvtime/greedyh.c:
Use "_stdint.h" instead of <stdint.h>.
Original commit message from CVS:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_init):
* gst/deinterlace2/tvtime/greedy.c: (deinterlace_frame_di_greedy):
* gst/deinterlace2/tvtime/greedyh.c:
(deinterlace_frame_di_greedyh):
* gst/deinterlace2/tvtime/speedtools.h:
* gst/deinterlace2/tvtime/speedy.c:
* gst/deinterlace2/tvtime/speedy.h:
* gst/deinterlace2/tvtime/tomsmocomp.c: (Fieldcopy):
* gst/deinterlace2/tvtime/tomsmocomp/TomsMoCompAll.inc:
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir):
Get rid of speedy.[ch] as we don't use most of it's code anyway
and it doesn't seem to be relicensed to LGPL. Use memcpy() instead
of the speedy memcpy everywhere instead.
* gst/deinterlace2/gstdeinterlace2.h:
Remove many unused declarations.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c: (gst_deinterlace2_src_query):
Divide latency be 2 to convert from fields to frames.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_c),
(deinterlace_greedy_packed422_scanline_mmx),
(deinterlace_greedy_packed422_scanline_mmxext),
(deinterlace_frame_di_greedy):
Don't use scanlines function from gstdeinterlace2 as it's
not appropiate for this method. Instead implement deinterlace_frame
function by taking the one from greedyh.
* gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C):
Small fix for the C implementation.
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_frame_vfir):
Don't use the scanlines function from gstdeinterlace2 as it's only
used for this method and will be removed. Instead implement
deinterlace_frame function and make it a bit more efficient.
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_class_init), (gst_deinterlace2_set_method),
(gst_deinterlace2_push_history), (gst_deinterlace2_chain),
(gst_deinterlace2_setcaps), (gst_deinterlace2_sink_event),
(gst_deinterlace2_change_state), (gst_deinterlace2_src_event),
(gst_deinterlace2_src_query):
Fix coding style and remove scanlines function as it's unused now.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedyh.asm:
* gst/deinterlace2/tvtime/greedyh.c: (greedyDScaler_C),
(deinterlace_frame_di_greedyh), (dscaler_greedyh_get_method):
* gst/deinterlace2/tvtime/greedyhmacros.h:
Add a C implementation for the greedyh deinterlacing method, clean
up the code a bit and mark the SSE version as MMXEXT as it doesn't
require any SSE instructions.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_set_property), (gst_deinterlace2_chain),
(gst_deinterlace2_setcaps):
If we're outputting all fields the framerate has to be doubled.
Set duration on the outgoing buffers.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_mmx),
(deinterlace_greedy_packed422_scanline_mmxext):
Optimize MMX/MMXEXT implementations a bit by requiring two less
memory accesses and fix the workaround for the missing right shift
on bytes to unset the highest bit of every byte.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_mmxext):
Remove sfence instruction as it's not needed and actually is an SSE
instruction.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_mmx),
(deinterlace_greedy_packed422_scanline):
Add plain MMX implementation for the greedyl method.
Original commit message from CVS:
* gst/deinterlace2/Makefile.am:
Move the assembly includes to noinst_HEADERS where they belong.
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
(deinterlace_line_mmx):
Fix C and MMX implementations a bit more.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_c),
(deinterlace_greedy_packed422_scanline_mmxext),
(deinterlace_greedy_packed422_scanline):
Fix the C implementation to produce correct results and optimize the
MMXEXT implementation.
Handle odd widths and don't read over array boundaries in the MMXEXT
implementation.
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_c),
(deinterlace_line_mmx), (deinterlace_scanline_vfir):
Fix a small rounding bug in the MMX implementation, the MMX
implementation doesn't actually need MMXEXT instructions so don't mark
it as such.
Handle odd widths in both implementations.
Original commit message from CVS:
* gst/deinterlace2/tvtime/greedy.c:
(deinterlace_greedy_packed422_scanline_sse),
(deinterlace_greedy_packed422_scanline_c),
(deinterlace_greedy_packed422_scanline):
Implement a C version of the greedy low motion algorithm and mark the
assembly optimized version as SSE as it uses SSE instructions
additional to MMX instructions.
Original commit message from CVS:
* gst/deinterlace2/tvtime/vfir.c: (deinterlace_line_mmxext),
(deinterlace_line_c), (deinterlace_scanline_vfir):
Make it possible to use the vfir method on X86 CPUs without MMXEXT too
but use the MMXEXT optimized code whenever possible.
Original commit message from CVS:
* gst/deinterlace2/gstdeinterlace2.c:
(gst_deinterlace2_class_init), (gst_deinterlace2_init),
(gst_deinterlace2_reset_history), (gst_deinterlace2_reset),
(gst_deinterlace2_finalize), (gst_deinterlace2_chain),
(gst_deinterlace2_sink_event), (gst_deinterlace2_change_state),
(gst_deinterlace2_src_query):
* gst/deinterlace2/gstdeinterlace2.h:
Reset element state on PAUSED->READY properly, don't leak any buffers
when finalizing, allocate buffers with gst_pad_alloc_buffer() and
properly return flow returns from gst_pad_push() instead of ignoring them.
Original commit message from CVS:
* configure.ac:
* gst/deinterlace2/Makefile.am:
* gst/deinterlace2/tvtime/greedyh.asm:
* gst/deinterlace2/tvtime/tomsmocomp/SearchLoopBottom.inc:
Fix compilation on generic x86/amd64 and include deinterlace2 in the
build system. Because of several bugs it's still enabled only
by --enable-experimental.
Original commit message from CVS:
* gst/selector/gstoutputselector.c:
* tests/icles/output-selector-test.c:
Use BOILERPLATE macro and update test to the latest api changes.
Original commit message from CVS:
* ext/resindvd/resindvdbin.c:
Parse the URI argument into the device name so dvd:///path/to/image
works.
* ext/resindvd/resindvdsrc.c:
Implement a trivial duration query reporting the current PGC length.
* gst/dvdspu/gstdvdspu.c:
Rename typo in the function name.
Original commit message from CVS:
* configure.ac:
Check for libdvdnav to build resindvd.
* ext/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/resindvd/gstmpegdefs.h:
* ext/resindvd/gstmpegdemux.c:
* ext/resindvd/gstmpegdemux.h:
* ext/resindvd/gstmpegdesc.c:
* ext/resindvd/gstmpegdesc.h:
* ext/resindvd/gstpesfilter.c:
* ext/resindvd/gstpesfilter.h:
* ext/resindvd/plugin.c:
* ext/resindvd/resin-play:
* ext/resindvd/resindvdbin.c:
* ext/resindvd/resindvdbin.h:
* ext/resindvd/resindvdsrc.c:
* ext/resindvd/resindvdsrc.h:
* ext/resindvd/rsnaudiomunge.c:
* ext/resindvd/rsnaudiomunge.h:
* ext/resindvd/rsnbasesrc.c:
* ext/resindvd/rsnbasesrc.h:
* ext/resindvd/rsnpushsrc.c:
* ext/resindvd/rsnpushsrc.h:
* ext/resindvd/rsnstreamselector.c:
* ext/resindvd/rsnstreamselector.h:
First commit of DVD-Video playback component 'rsndvdbin'
and helper elements.
Use --enable-experimental for now, but feel free to give it a
try using the resin-play script.
* gst/dvdspu/gstdvdspu.c:
Add some extra guards for malformed events.
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.
