Increase default drift tolerance to 40ms to avoid glitches with decoders
or formats where there's a lot of timestamp jitter for some reason or
another (in this case: asf/wma), at least until we implement timestamp
smoothing.
Our calibration against the pipeline clock is done with the adjusted
ringbuffer time, so take the adjustement into account. Fixes some audio dropouts
when reusing audio sinks after switching clocks and slaving methods in a
pipeline.
When we are calibrating the internal clock against the external clock take into
account the time offset applied to our internal clock because we will subtract
that in the render_function again.
When we start and we need to produce the first sample, go to the next sample
that will be written into the ringbuffer instead of trying to go to sample 0.
We relied on rather small ringbuffer sizes to correctly go to the current
sample, which breaks whith large buffers.
Fixes#600945
Add drift-tolerance property (defaulting to 20ms) to handle resync after clock
drift or timestamp drift instead of relying on the latency-time value for clock
drift and 500ms for timestamp drift.
Remove warning about discont timestamp and simply resync. The warning is in some
cases not correct and is triggered more frequently now that we lower the
tolerance value.
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
Check for pulsesink < 0.10.17 because it includes code that is now included in
baseaudiosink. Disable that code in baseaudiosink to be compatible with the
older version.
Take the time of the clock so that the last_time field is set. This is important
for sinks that restart their internal ringbuffer after a caps change and need to
know the last know position.
When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
also make sure that the clock is updated with the elapsed time so that it
alsways increments even when the ringbuffer goes back to 0. When this happened
we need to adjust the sample position for the reset ringbuffer.
Fixes#594136
The old one did the mistake of not actually advancing the ringbuffer, it just
adjusted the segbase, introducing the whole lenght of the ringbuffer as an
extra delay in the pipeline.
Also make sure that the resync can never go back in time, producing the same
timestamps that has already been produced, as this can cause severe problems
for sinks and other synching mechanisms.
Fixes#594256
Add section docs for multichannel, so that it has a short desc in the toc too.
Move the section docs in adio up, so that the follow the copyright like
elsewhere.
Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.
When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().
Add a debug category and some debug lines to the audio clock.
API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
* gst-libs/gst/audio/gstaudioclock.h:
Make gst_audio_clock_new use const gchar* to ease the wrapping of
C++ bindings. Fixes#566723.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c:
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c:
* win32/MANIFEST:
* win32/common/audio-enumtypes.c:
(gst_audio_channel_position_get_type),
(gst_ring_buffer_state_get_type),
(gst_ring_buffer_seg_state_get_type),
(gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
* win32/common/audio-enumtypes.h:
* win32/common/multichannel-enumtypes.c:
* win32/common/multichannel-enumtypes.h:
* win32/vs6/grammar.dsp:
* win32/vs6/libgstaudio.dsp:
* win32/vs7/libgstaudio.vcproj:
* win32/vs8/libgstaudio.vcproj:
Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
audio- in order to wrap all enums declarations of that library.
This modification should not matter since that header file is not a
public header (it will be included by public headers).
Modify win32 crap^Wfiles accordingly.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Complete Sebastien's commit from the 13th by exporting the
_slave_method_get_type() methods.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps):
Pause the write thread before deactivating and releasing the ringbuffer
to avoid a deadlock when we do gapless playback with different sample
rates in playbin2. Fixes#564929.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
Make GstAudioSrcSlaveMethod get_type() function non-static
as it's public now.
* win32/common/libgstaudio.def:
* win32/common/libgstnetbuffer.def:
Add some missing functions to the list of exported symbols.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_slave_method_get_type),
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_slave_method_get_type),
(gst_base_audio_src_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
public API. This is needed for the C++ bindings to be able
to use this base classes. Fixes bug #564200, #564206.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_callback):
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
for the latency to expire, fixes#559567.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Implement a separate activate functions to start monitoring the segments
or, in pull mode, pulling in data.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
(gst_base_audio_sink_activate_pull),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Implement pad and element convert query function.
Activate the ringbuffer.
Use the segment last_stop value as the offset to pull.
Use new basesink _do_preroll() method to preroll in the pulling thread.
Take appropriate locking in the pulling thread.
* gst-libs/gst/audio/gstringbuffer.h:
Update some docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
(gst_ring_buffer_activate), (gst_ring_buffer_is_active):
* gst-libs/gst/audio/gstringbuffer.h:
Add methods to more accuratly control the pulling thread of a
ringbuffer.
Add format conversion helper code to the ringbuffer.
API: GstRingBuffer:gst_ring_buffer_activate()
API: GstRingBuffer:gst_ring_buffer_is_active()
API: GstRingBuffer:gst_ring_buffer_convert()
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Signal thread startup earlier so that we can immediatly go into pull
mode when we have to and block on preroll.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read):
In pull mode we want the callback to prepull a buffer we can preroll on
even when we are not yet playing.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix debug statements (space between '%' and actual format).
