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.
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:
* 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:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbenc.h:
Pass the discont flag from the input buffer on to the output buffer in
the AMR encoder.
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:
* ext/amrwb/gstamrwbparse.c:
* ext/amrwb/gstamrwbparse.h:
Add flush seek handler. Taken from recent armnbparse changes.
Sync the code more and use #defines for HEADER.
Original commit message from CVS:
* ext/amrwb/gstamrwbparse.c:
* ext/amrwb/gstamrwbparse.h:
Fix the duration query. Also set caps on the pads and buffers more
correctly. Taken from recent armnbparse changes.
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:
* ext/resindvd/resindvdsrc.c:
Fix next/prev chapter seeking at the beginning or end.
Use 64-bit scaling utility functions for converting MPEG
timestamps.
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:
* configure.ac
* ext/metadata/README:
* ext/metadata/metadataexif.c:
* ext/metadata/metadatatags.c:
* ext/metadata/metadatatags.h:
Start using core geo tags (bump req). Fix handling of location
references.
* tests/check/Makefile.am:
Sort blacklisted elements and remove moved ones. Add new test.
* tests/check/pipelines/metadata.c:
Add first tests for metadata element.
* tests/icles/metadata_editor.c:
Move free to correct place.
Original commit message from CVS:
* tests/check/generic/states.c:
Stop test on state-change error. Should be applied on other modules if
we agree that it makes sense.
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:
* ext/metadata/metadata_mapping.htm:
* ext/metadata/metadataxmp.c:
* ext/metadata/Makefile.am:
Add mapping of format and mime type to xmp.
Original commit message from CVS:
* ext/metadata/README:
* ext/metadata/metadataexif.c:
* ext/metadata/metadatatags.c:
* ext/metadata/metadatatags.h:
Reverting. Will need to wait for core 0.10.21 release.
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:
* sys/Makefile.am:
* sys/wasapi/Makefile.am:
* sys/wasapi/gstwasapi.c:
* sys/wasapi/gstwasapisink.c:
* sys/wasapi/gstwasapisink.h:
* sys/wasapi/gstwasapisrc.c:
* sys/wasapi/gstwasapisrc.h:
* sys/wasapi/gstwasapiutil.c:
* sys/wasapi/gstwasapiutil.h:
New plugin for audio capture and playback using Windows Audio Session
API (WASAPI) available with Vista and newer (#520901).
Comes with hardcoded caps and obviously needs lots of love. Haven't
had time to work on this code since it was written, was initially just
a quick experiment to play around with this new API.
Original commit message from CVS:
* sys/dshowdecwrapper/gstdshowaudiodec.cpp
(AudioFakeSink.DoRenderSample):
Fix a couple of signed/unsigned comparison warnings.
Original commit message from CVS:
* sys/dshowdecwrapper/gstdshowaudiodec.h (AudioFakeSink.AudioFakeSink):
* sys/dshowdecwrapper/gstdshowvideodec.h (VideoFakeSink.VideoFakeSink):
Use the _T() macro to support both Unicode and MBCS.
Original commit message from CVS:
* ext/resindvd/gstmpegdemux.c:
* ext/resindvd/gstmpegdemux.h:
* ext/resindvd/resindvdbin.c:
* ext/resindvd/resindvdsrc.c:
* ext/resindvd/rsnstreamselector.c:
Add in Title/Chapter seeking, and simple but buggy audio
and subtitle stream selection.
Original commit message from CVS:
* sys/dshowdecwrapper/gstdshowaudiodec.cpp:
* sys/dshowdecwrapper/gstdshowaudiodec.h:
* sys/dshowdecwrapper/gstdshowfakesrc.cpp:
* sys/dshowdecwrapper/gstdshowutil.cpp:
* sys/dshowdecwrapper/gstdshowutil.h:
* sys/dshowdecwrapper/gstdshowvideodec.cpp:
* sys/dshowdecwrapper/gstdshowvideodec.h:
Prefer known-good filters, create directly by GUID if possible,
fall back to creating highest-merit filter otherwise.
Fixes playback with random dshow filters installed in some
cases.
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:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_start),
(gst_neonhttp_src_send_request_and_redirect):
Clean up the debug logging code and #ifdef mess a bit: whether or not
gstreamer debug messages should be output should not depend on an
element property; also, GST_ELEMENT_ERROR will leave a line in the log
already, so merge the more useful debug log messages with the less useful
error debug strings.
Original commit message from CVS:
* ext/neon/gstneonhttpsrc.c: (gst_neonhttp_src_start):
Don't post LIBRARY_INIT errors where we should be posting
RESOURCE OPEN_READ errors. Fixes#552506.
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:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowdecwrapper/gstdshowaudiodec.c:
* sys/dshowdecwrapper/gstdshowaudiodec.cpp:
* sys/dshowdecwrapper/gstdshowaudiodec.h:
* sys/dshowdecwrapper/gstdshowdecwrapper.c:
* sys/dshowdecwrapper/gstdshowdecwrapper.cpp:
* sys/dshowdecwrapper/gstdshowdecwrapper.h:
* sys/dshowdecwrapper/gstdshowfakesrc.cpp:
* sys/dshowdecwrapper/gstdshowfakesrc.h:
* sys/dshowdecwrapper/gstdshowutil.cpp:
* sys/dshowdecwrapper/gstdshowutil.h:
* sys/dshowdecwrapper/gstdshowvideodec.c:
* sys/dshowdecwrapper/gstdshowvideodec.cpp:
* sys/dshowdecwrapper/gstdshowvideodec.h:
Major rewrite of dshowdecwrapper. Converts code to
C++, moves to direct use of DirectShow base classes,
make a lot of code clearer, simplify, etc.
Fix decode of MP3 on Vista by working around an apparent
bug in the decoder.
Original commit message from CVS:
* sys/winks/gstksclock.c (gst_ks_clock_worker_thread_func,
gst_ks_clock_start):
Synchronize KS clock as a single-shot operation for now, there's not
much point in doing it periodically until we're actually using the
KS timestamps for anything else than just discarding old frames.
* sys/winks/gstksvideosrc.c (gst_ks_video_src_open_device):
Provide the GstClock when opening the device if we already have one.
Original commit message from CVS:
* sys/winks/gstksvideodevice.c (GST_DEBUG_IS_ENABLED, last_timestamp,
gst_ks_video_device_prepare_buffers, gst_ks_video_device_create_pin,
gst_ks_video_device_set_state, gst_ks_video_device_request_frame,
gst_ks_video_device_read_frame):
Guard against capturing old frames by keeping track of the last
timestamp and also zero-fill the buffers before each capture.
Only assign a master clock if the pin hasn't already got one.
Actually free buffers on the way down to avoid a huge memory leak,
as this was previously done when changing state to ACQUIRE downwards
and we now skip that state on the way down.
Add some debug.
* sys/winks/gstksvideosrc.c (DEFAULT_DEVICE_PATH, DEFAULT_DEVICE_NAME,
DEFAULT_DEVICE_INDEX, KS_WORKER_LOCK, KS_WORKER_UNLOCK,
KS_WORKER_WAIT, KS_WORKER_NOTIFY, KS_WORKER_WAIT_FOR_RESULT,
KS_WORKER_NOTIFY_RESULT, KS_WORKER_STATE_STARTING,
KS_WORKER_STATE_READY, KS_WORKER_STATE_STOPPING,
KS_WORKER_STATE_ERROR, KsWorkerState, device_path, device_name,
device_index, running, worker_thread, worker_lock,
worker_notify_cond, worker_result_cond, worker_state,
worker_pending_caps, worker_setcaps_result, worker_pending_run,
worker_run_result, gst_ks_video_src_reset,
gst_ks_video_src_apply_driver_quirks, gst_ks_video_src_open_device,
gst_ks_video_src_close_device, gst_ks_video_src_worker_func,
gst_ks_video_src_start_worker, gst_ks_video_src_stop_worker,
gst_ks_video_src_change_state, gst_ks_video_src_set_clock,
gst_ks_video_src_set_caps, gst_ks_video_src_timestamp_buffer,
gst_ks_video_src_create):
Remove ENABLE_CLOCK_DEBUG define, it's GST_LEVEL_DEBUG after all.
Get rid of PROP_ENSLAVE_KSCLOCK and always slave the ks clock to the
GStreamer clock, it doesn't seem to hurt and matches DirectShow's
behavior. As an added bonus we usually get PresentationTime set for
each frame, so we can expand on this later for smarter latency
reporting (by looking at the diff between the timestamp from the
driver and the time according to the GStreamer clock).
Use an internal worker thread for opening the device, setting caps,
changing its state and closing it. This way we're a lot more
compatible with drivers that rely on hacks to do video-effects
between the low-level NT API and the application. Ick.
Start the ks clock and set the pin to KSSTATE_RUN on the first
create() so that we'll hopefully get hold of the GStreamer clock
from the very beginning. This way there's no chance that the
timestamps will make a sudden jump in the beginning of the stream
when we're running with a clock.
* sys/winks/kshelpers.c (CHECK_OPTIONS_FLAG,
ks_options_flags_to_string):
Reorder the flags to match the headerfile order, and make the string
a bit more compact.
* sys/winks/ksvideohelpers.c (ks_video_probe_filter_for_caps):
Avoid leaking KSPROPERTY_PIN_DATARANGES.
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:
* ext/celt/gstceltdec.c: (celt_dec_chain_parse_data):
Correctly take the granulepos from upstream if possible and
correctly handle the granulepos in various calculations: the
granulepos is the sample number of the _last_ sample in a frame, not
the first.
* ext/celt/gstceltenc.c: (gst_celt_enc_sinkevent),
(gst_celt_enc_encode), (gst_celt_enc_chain),
(gst_celt_enc_change_state):
* ext/celt/gstceltenc.h:
Handle non-zero start timestamps in the encoder and detect/handle
stream discontinuities. Fixes bug #547075.
Original commit message from CVS:
* ext/faac/gstfaac.c: (gst_faac_init), (gst_faac_sink_event),
(gst_faac_chain), (gst_faac_change_state):
* ext/faac/gstfaac.h:
Add code for calculating proper timestamp/duration for the trailing
encoded buffers that faac will output when receiving EOS.
Original commit message from CVS:
* sys/winks/ksvideohelpers.c (ks_video_media_type_free):
Avoid leaking the KSDATARANGE member of each KsVideoMediaType.
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:
* sys/winks/gstksvideodevice.c (gst_ks_video_device_class_init,
gst_ks_video_device_set_state):
Don't set the pin state to KSSTATE_RUN from the streaming thread.
Skip KSSTATE_ACQUIRE when changing pin state downwards.
Be nice and specify G_PARAM_STATIC_STRINGS.
Remove unused finalize method.
* sys/winks/gstksvideosrc.c (DEFAULT_ENABLE_QUIRKS, PROP_ENABLE_QUIRKS,
enable_quirks, gst_ks_video_src_class_init, gst_ks_video_src_init,
gst_ks_video_src_finalize, gst_ks_video_src_get_property,
gst_ks_video_src_set_property, gst_ks_video_src_reset,
gst_ks_video_src_apply_driver_quirks, gst_ks_video_src_change_state,
gst_ks_video_src_set_caps):
First driver quirk: work around Logitech's hostile driver software to
improve stability and performance. See comments for details.
Provide a property to disable driver quirks (enabled by default).
Be nice and specify G_PARAM_STATIC_STRINGS.
Remove unused dispose method.
Tweak include order.
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:
* configure.ac:
* sys/Makefile.am:
* sys/winks/Makefile.am:
* sys/winks/gstksclock.c:
* sys/winks/gstksclock.h:
* sys/winks/gstksvideodevice.c:
* sys/winks/gstksvideodevice.h:
* sys/winks/gstksvideosrc.c:
* sys/winks/gstksvideosrc.h:
* sys/winks/kshelpers.c:
* sys/winks/kshelpers.h:
* sys/winks/ksvideohelpers.c:
* sys/winks/ksvideohelpers.h:
New plugin for low-latency video capture on Windows (#519935).
Uses Kernel Streaming, the lowest level API for doing video capture
on Windows (more or less just raw ioctls).
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:
* sys/dshowdecwrapper/gstdshowaudiodec.c:
Flip mpeg1/mpeg2 arrays for mpeg audio. Detect which type the audio
is correctly, instead of backwards. No functional changes, since this
mistake was completely self-consistent.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
Add, but commented out xml/element-dc1394.xml. Its documented, but
I can't get it to be build.
* ext/celt/gstceltdec.c:
* ext/celt/gstceltenc.c:
Fix doc warnings and reformat the doc block.
Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/acmenc/acmenc.c:
Add new windows ACM encoder wrapper.
Original commit message from CVS:
* sys/dshowdecwrapper/gstdshowaudiodec.c:
* sys/dshowdecwrapper/gstdshowaudiodec.h:
* sys/dshowdecwrapper/gstdshowvideodec.c:
* sys/dshowdecwrapper/gstdshowvideodec.h:
* sys/dshowvideosink/dshowvideosink.cpp:
* sys/dshowvideosink/dshowvideosink.h:
Initialise COM with default flags.
Only deinitialise if the initialisation was successful.
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:
* ext/x264/gstx264enc.c: (gst_x264_enc_reset),
(gst_x264_enc_chain), (gst_x264_enc_encode_frame):
* ext/x264/gstx264enc.h:
Do not deal with duplicated input (timestamps). If needed,
a generic element can do so.
Do not manipulate input timestamps on the way out,
since that shifts the timeline and A/V sync.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Integrate new properties into documentation.
* ext/x264/gstx264enc.c: (gst_x264_enc_class_init),
(gst_x264_enc_init), (gst_x264_enc_init_encoder),
(gst_x264_enc_set_property), (gst_x264_enc_get_property):
Fix up API prior to eventual plugin move.
API: GstX264Enc:pass (provides more options, and changed to enum)
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:
* ext/x264/gstx264enc.c:
* tests/check/Makefile.am:
* tests/check/elements/x264enc.c: (setup_x264enc),
(cleanup_x264enc), (GST_START_TEST), (x264enc_suite), (main):
Add documentation and unit test for x264enc.
Original commit message from CVS:
* ext/x264/gstx264enc.c: (gst_x264_enc_init),
(gst_x264_enc_header_buf), (gst_x264_enc_encode_frame):
Allocate some buffers in more adaptive and economical fashion.
Original commit message from CVS:
* configure.ac:
Check for sufficiently up-to-date x264 API.
* ext/x264/gstx264enc.c: (gst_x264_enc_pass_get_type),
(gst_x264_enc_base_init), (gst_x264_enc_class_init),
(gst_x264_enc_init), (gst_x264_enc_init_encoder),
(gst_x264_enc_set_property), (gst_x264_enc_get_property):
* ext/x264/gstx264enc.h:
Expose some more parameters of the x264 encoder as properties.
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:
* ext/x264/gstx264enc.c: (gst_x264_enc_log_callback),
(gst_x264_enc_finalize), (gst_x264_enc_header_buf),
(gst_x264_enc_set_src_caps), (gst_x264_enc_sink_set_caps),
(gst_x264_enc_flush_frames):
Coding style and layout; re-order some functions in more
typical and natural flow.
Original commit message from CVS:
* ext/x264/Makefile.am:
* ext/x264/gstx264enc.c: (gst_x264_enc_header_buf),
(gst_x264_enc_sink_set_caps), (gst_x264_enc_base_init),
(gst_x264_enc_class_init), (gst_x264_enc_log_callback),
(gst_x264_enc_init), (gst_x264_enc_init_encoder),
(gst_x264_enc_finalize), (gst_x264_enc_chain),
(gst_x264_enc_encode_frame), (plugin_init):
* ext/x264/gstx264enc.h:
Use video format library and GST_WRITE_*_BE macros where applicable.
Use finalize in stead of dispose.
Set up debug category and log callback.
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/sndfile/gstsf.c: (plugin_init):
* sys/dvb/gstdvbsrc.c: (gst_dvbsrc_plugin_init):
* sys/oss4/oss4-audio.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
Original commit message from CVS:
* ext/twolame/gsttwolame.c: (gst_two_lame_sink_setcaps),
(gst_two_lame_chain):
* ext/twolame/gsttwolame.h:
Allow raw float samples as input for encoding.
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.
