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:
* tests/check/elements/audioresample.c: (setup_audioresample),
(fail_unless_perfect_stream), (test_perfect_stream_instance),
(test_discont_stream_instance):
Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
Add debugging for coherence.
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.
