Original commit message from CVS:
* gst/mxf/mxfdemux.c: (gst_mxf_pad_init),
(gst_mxf_demux_reset_metadata), (gst_mxf_demux_combine_flows),
(gst_mxf_demux_handle_generic_container_essence_element):
Properly combine flow returns of the source pads.
Improve debugging and reset pad fields together with the
metadata to prevent invalid memory references.
* gst/mxf/mxfparse.c: (mxf_metadata_preface_parse):
Fix typo.
Original commit message from CVS:
* gst/mxf/mxfaes-bwf.c:
(mxf_metadata_wave_audio_essence_descriptor_reset):
* gst/mxf/mxfmpeg.c: (mxf_metadata_mpeg_video_descriptor_reset):
* gst/mxf/mxfparse.c: (mxf_metadata_file_descriptor_handle_tag),
(mxf_metadata_file_descriptor_reset),
(mxf_metadata_generic_sound_essence_descriptor_reset),
(mxf_metadata_generic_picture_essence_descriptor_reset),
(mxf_metadata_cdci_picture_essence_descriptor_reset),
(mxf_metadata_rgba_picture_essence_descriptor_reset),
(mxf_metadata_multiple_descriptor_reset):
* gst/mxf/mxfparse.h:
When memset'ing the descriptors to zeroes only touch the part
of this descriptor, not it's parent type.
Set correct default values for all metadata fields.
Original commit message from CVS:
* gst/mpegdemux/gstmpegtsdemux.c:
* gst/mpegdemux/gstmpegtsdemux.h:
Enable latency code, since we're definitely after 0.10.12
Original commit message from CVS:
Patch by: Jonathan Matthew <notverysmart at gmail dot com>
* ext/spc/gstspc.c: (gst_spc_dec_sink_event), (spc_setup):
Post an error and push EOS when we can't start playback for some reason.
also avoid a crash when fed an empty file. Fixes#480543.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (test_pipeline):
Make unit test again faster to prevent timeouts with valgrind.
Original commit message from CVS:
* gst/mxf/mxfmpeg.c: (mxf_mpeg_video_create_caps):
* gst/mxf/mxfparse.c:
(mxf_metadata_generic_picture_essence_descriptor_set_caps):
* gst/mxf/mxfparse.h:
Set video dimensions and pixel aspect ratio on the caps.
Original commit message from CVS:
* gst/mxf/mxfaes-bwf.c:
(mxf_metadata_wave_audio_essence_descriptor_handle_tag):
* gst/mxf/mxfaes-bwf.h:
* gst/mxf/mxfdemux.c:
(gst_mxf_demux_handle_metadata_generic_descriptor),
(gst_mxf_demux_handle_metadata_file_descriptor),
(gst_mxf_demux_handle_metadata_multiple_descriptor),
(gst_mxf_demux_handle_metadata_generic_picture_essence_descriptor),
(gst_mxf_demux_handle_metadata_cdci_picture_essence_descriptor),
(gst_mxf_demux_handle_metadata_mpeg_video_descriptor),
(gst_mxf_demux_handle_metadata_generic_sound_essence_descriptor),
(gst_mxf_demux_handle_metadata_wave_audio_essence_descriptor),
(gst_mxf_demux_handle_header_metadata_resolve_references),
(gst_mxf_demux_handle_metadata):
* gst/mxf/mxfmpeg.c:
(mxf_metadata_mpeg_video_descriptor_handle_tag):
* gst/mxf/mxfmpeg.h:
* gst/mxf/mxfparse.c: (mxf_partition_pack_parse),
(mxf_primer_pack_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_descriptor_parse),
(mxf_metadata_generic_descriptor_handle_tag),
(mxf_metadata_generic_descriptor_reset),
(mxf_metadata_file_descriptor_handle_tag),
(mxf_metadata_generic_sound_essence_descriptor_handle_tag),
(mxf_metadata_generic_picture_essence_descriptor_handle_tag),
(mxf_metadata_cdci_picture_essence_descriptor_handle_tag),
(mxf_metadata_multiple_descriptor_handle_tag),
(mxf_metadata_locator_parse):
* gst/mxf/mxfparse.h:
Rewrite parsing of descriptors to go top-down instead of bottom-up
which makes it possible to have the buffer data non-writable.
Improve debugging a bit.
Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
Add new directory for new acmmp3dec plugin
* sys/acmenc/Makefile.am:
* sys/acmenc/acmmp3dec.c:
Add new acmmp3dec plugin, which uses ACM for decoding mp3 on windows.
Original commit message from CVS:
* gst/mxf/mxfdemux.c:
(gst_mxf_demux_handle_generic_container_essence_element):
Make sure to only output generic container essence elements
for a track if the body SID of the surrounding partition is
the same as the body SID of the track's source package.
Original commit message from CVS:
2008-11-24 Julien Moutte <julien@fluendo.com>
* gst/flv/gstflvdemux.c: (gst_flv_demux_find_offset),
(gst_flv_demux_handle_seek_push),
(gst_flv_demux_handle_seek_pull):
Fix non key unit seeking by always going to the previous
keyframe. Mark
the discont flag when we've moved in the file.
* gst/flv/gstflvparse.c: (gst_flv_parse_audio_negotiate): MP3
streams
are parsed already, makes autoplugged pipelines shorter.
Original commit message from CVS:
* gst/mxf/mxfdemux.c: (gst_mxf_demux_pull_klv_packet),
(gst_mxf_demux_handle_klv_packet), (gst_mxf_demux_chain):
* gst/mxf/mxfparse.c: (mxf_product_version_parse),
(mxf_metadata_identification_parse),
(mxf_metadata_content_storage_parse):
Allow non-MXF KLV packets and just drop them instead of throwing
an error and handle 9 byte product versions as written by Avid.
This doesn't add support for the non-standard Avid MXF files
but at least makes it possible to parse their header metadata.
Fix a copy&paste error in debug output.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
Query port latencies for sink/src delays.
* ext/jack/gstjackbin.c:
No printf please.
Original commit message from CVS:
* gst/mxf/mxfdemux.c: (gst_mxf_demux_pull_klv_packet),
(gst_mxf_demux_chain):
Actually we support a length stored inside 8 bytes but it must
be smaller than G_MAXUINT for GstBuffer.
Original commit message from CVS:
Patch by: Tal Shalif <tshalif at nargila dot org>
* ext/resindvd/resindvdsrc.c:
(rsn_dvdsrc_prepare_streamsinfo_event):
Fix format string. Fixes bug #561992.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_set_property),
(rtp_session_get_property):
Add property to configure the RTCP MTU.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(copy_source), (rtp_session_create_sources),
(rtp_session_get_property):
Add G_PARAM_STATIC_STRINGS.
Add property to return a GValueArray of all known RTPSources in the
session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_create_sdes), (rtp_source_set_property),
(rtp_source_get_property):
Remove properties to set the various SDES items, an application is never
supposed to change the RTPSource data.
Change the SDES getter properties to one SDES property that returns all
SDES items in a GstStructure.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (GST_START_TEST):
Make the unit test a bit faster to prevent timeouts, especially
with valgrind.
Original commit message from CVS:
* gst/mxf/mxfdemux.c: (gst_mxf_demux_push_src_event),
(gst_mxf_demux_handle_header_metadata_update_streams):
* gst/mxf/mxfparse.c: (gst_mxf_ul_hash),
(mxf_partition_pack_parse), (mxf_primer_pack_parse),
(mxf_metadata_preface_parse), (mxf_metadata_content_storage_parse),
(mxf_metadata_generic_package_parse),
(mxf_metadata_sequence_parse),
(mxf_metadata_generic_descriptor_parse),
(mxf_metadata_multiple_descriptor_parse):
Some more format string fixes and usage of guint instead of gint
where negative values don't make sense.
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:
* ext/x264/gstx264enc.c: (gst_x264_enc_set_src_caps):
Construct source caps in more conventional (and correct) manner.
Original commit message from CVS:
* gst-libs/gst/play/.cvsignore:
* gst-libs/gst/play/play.h:
* gst-libs/gst/play/play.vcproj:
Remove cruft. This is not entered by make and its not even compilable.
Original commit message from CVS:
* ext/dirac/gstdiracenc.cc:
Set pixel-aspect-ratio correctly in the encoder API, as well
as some default gstreamerish colorspace properties. Also,
apparently, change a bunch of indentation.
Original commit message from CVS:
* ext/jp2k/gstjasperdec.c: (gst_jasper_dec_init),
(gst_jasper_dec_reset), (gst_jasper_dec_negotiate),
(gst_jasper_dec_get_picture):
* ext/jp2k/gstjasperdec.h:
Make pad template caps reflect the supported formats.
Add or modify some debug statements, and slightly simplify image
passing to encoding library.
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:
* tests/check/Makefile.am:
* tests/check/elements/aacparse.c:
* tests/check/elements/amrparse.c:
Add unit tests for new parsers.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/qtmux.c: (setup_src_pad),
(teardown_src_pad), (setup_qtmux), (cleanup_qtmux),
(check_qtmux_pad), (GST_START_TEST), (qtmux_suite), (main):
Add unit test for qtmux.
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:
* configure.ac:
Use AC_TRY_COMPILE instead of AC_TRY_RUN as the result of the linking
is what is interesting, not that it actually runs.
Fixes cross-compilation and fixes bug #558639.
Original commit message from CVS:
* sys/qtwrapper/audiodecoders.c:
Add ALAC support.
Fix decode of mono AAC files created by itunes.
Set output format correctly (don't ask quicktime to
resample for us).
Use a larger decode buffer to avoid problems with large
ALAC packets.
Fix decode to loop until we have all output data.
* sys/qtwrapper/qtutils.c:
Fix includes so we compile on more OSes.
Original commit message from CVS:
* configure.ac:
Require at least Gtk 2.8.0 for the demos (that's the oldest I can
test with; I'm fairly certain Gtk 2.0.0 is not good enough any
longer); clean up some unused Gtk-related configure cruft.
* examples/scaletempo/demo-gui.c:
Define Gtk 2.12 function to noop when compiling against older Gtk.
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:
* ext/resindvd/resindvdsrc.c:
* ext/resindvd/resindvdsrc.h:
Better fix for #546319 and similar cases by explicitly
registering when we're in playing state or not.
Original commit message from CVS:
* ext/ladspa/gstladspa.c:
Whitespace.
* ext/ladspa/gstsignalprocessor.c:
Add a FIXME:. not sure if this code does the forwarding correctly.
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:
* ext/resindvd/resindvdsrc.c:
Make sure to start the NAV packet processing when changing
state to PLAYING by passing a flag that indicates the state
change is in progress.
Fixes: #546319
Original commit message from CVS:
* ext/resindvd/resin-play:
Remove $@ to fix parse_launch warning
* ext/resindvd/resin-play2:
Add a version that uses deinterlace and xvimagesink.
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/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:
* ext/apexsink/gstapexplugin.c: (plugin_init):
Set apexsink's rank to NONE so it doesn't get used by
autoaudiosink (there's no point really). (#556588)
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:
2008-10-16 Michael Smith <msmith@songbirdnest.com>
* sys/acmenc/Makefile.am:
Remove incorrect use of DIRECTSOUND_LDFLAGS
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:
* ext/apexsink/gstapexraop.c: (gst_apexraop_connect),
(gst_apexraop_set_volume):
Fix format string compiler warnings.
Original commit message from CVS:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Add some spaces in translateable strings.
Fixes: #555969#555968#555965
Original commit message from CVS:
* tests/check/pipelines/metadata.c:
Make the metadata test not fail when jpegenc isn't available....
as it isn't here, because it's not in this module, and
therefore not in the plugin path when the check runs.
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.
