Commit graph

26 commits

Author SHA1 Message Date
Tim-Philipp Müller
d7fc45f42e docs: fix up some Since: markers 2011-11-07 23:05:44 +00:00
Felipe Contreras
3df415d4c7 baseaudiosink: make discont-wait configurable
Now we can configure how much time to wait before deciding that a
discont has happened.

Also, adds getter and setter to allow derived implementations to set
this value upon construction.

Suggestions and several improvements by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-07 11:58:46 +01:00
Felipe Contreras
369cf3f14a baseaudiosink: split drift-tolerance into alignment-threshold
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
2011-11-07 11:10:05 +01:00
Arun Raghavan
643e5f586c baseaudiosink: Allow subclasses to provide payloaders
This allows subclasses to provide a "payload" function to prepare
buffers for consumption. The immediate use for this is for sinks that
can handle compressed formats - parsers are directly connected to the
sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading
might be used.

API: GstBaseAudioSinkClass:payload()

https://bugzilla.gnome.org/show_bug.cgi?id=642730
2011-05-14 18:23:18 +05:30
Mark Nauwelaerts
e73f293ee5 baseaudiosink: arrange for running clock when rendering eos
Commit ba2e500bd9 ensured to provide
a running clock when EOS had finished rendering.  However,
other measures are needed (and were in place before) to ensure a
running clock when EOS still needs rendering (i.e. waiting).

So, specifically, re-introduce eos_rendering removed in aforementioned commit,
this time as a public variable so subclasses can be aware of the situation.

Fixes (part of) #645961.

API: GstBaseAudioSink:eos_rendering
2011-03-31 13:18:53 +02:00
Havard Graff
3067a83df2 baseaudiosink: Added getter and setter for drift tolerance. 2010-09-24 13:06:35 +02:00
Wim Taymans
47550f6984 baseaudiosink: whitespace fixes 2009-09-09 16:17:02 +02:00
Edward Hervey
20adaa1328 gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
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.
2008-12-30 17:55:07 +00:00
Sebastian Dröge
04d9ff9a24 gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
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.
2008-12-13 06:57:09 +00:00
Wim Taymans
7916e386ca gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
2008-04-28 08:51:38 +00:00
Wim Taymans
157a65b15e Expose methods for some object properties so that subclasses can more easily configure them.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_set_provide_clock),
(gst_base_audio_sink_get_provide_clock),
(gst_base_audio_sink_set_slave_method),
(gst_base_audio_sink_get_slave_method),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
(gst_base_audio_sink_none_slaving),
(gst_base_audio_sink_handle_slaving):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
Added slave method none, that completely disables slaving to the
internal clock.
API: gst_base_audio_sink_set_provide_clock()
API: gst_base_audio_sink_get_provide_clock()
API: gst_base_audio_sink_set_slave_method()
API: gst_base_audio_sink_get_slave_method()
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_set_provide_clock),
(gst_base_audio_src_get_provide_clock),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Expose methods for some object properties so that subclasses can more
easily configure them.
API: gst_base_audio_src_set_provide_clock()
API: gst_base_audio_src_get_provide_clock()
2007-11-21 13:04:17 +00:00
Wim Taymans
450030ebaf gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
2007-03-28 14:50:47 +00:00
Wim Taymans
0990cbf274 gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Make the clock sync code more accurate wrt resampling and playback
at different rates.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
* gst-libs/gst/audio/gstringbuffer.h:
Use better algorithm to interpolate sample rates.
2006-11-13 17:30:17 +00:00
Wim Taymans
1166abbc99 gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Extract rate from the NEWSEGMENT event.
Use commit_full to also take rate adjustment into account when writing
samples to the ringbuffer.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Added _commit_full() to also take rate into account.
Use simple interpolation algorithm to resample audio.
API: gst_ring_buffer_commit_full()
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
* tests/examples/seek/seek.c: (segment_done):
Don't try to seek with 0.0 rate, just pause instead.
Remove bogus debug line.
2006-10-18 13:42:49 +00:00
Wim Taymans
7367722509 Added docs for the audio libs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstaudioclock.c:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
Added docs for the audio libs.
2006-09-27 11:05:08 +00:00
Jan Schmidt
45e06fe704 gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
(gst_ring_buffer_samples_done):
* gst-libs/gst/audio/gstringbuffer.h:
Document better the fact that latency_time and buffer_time are values
stored in microseconds, and not the usual GStreamer nanoseconds.
Change the variables (compatibly) that store them from GstClockTime
to guint64 to make it more clear that they're not storing clock times.
Also, remove the bogus property description that says the user can
specify -1 to get the default value, since that's never been the case.
When computing the default segment size for the ring buffer, make it
an integer number of samples.
When the sub-class indicates a delay greater than the number of
samples we've written return 0 from the audio sink get_time method.
2006-06-03 21:06:49 +00:00
Stefan Kost
1a2642a1d2 Fix broken GObject macros
Original commit message from CVS:
* ext/pango/gsttextrender.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstbasertppayload.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideosink.h:
* gst/playback/gstplaybasebin.h:
* gst/tcp/gstmultifdsink.h:
* sys/v4l/gstv4lelement.h:
Fix broken GObject macros
2006-04-08 18:09:17 +00:00
Thomas Vander Stichele
5f83aa7dfa expand tabs
Original commit message from CVS:
expand tabs
2005-12-06 19:42:02 +00:00
Wim Taymans
a3cb4d4937 gst-libs/gst/audio/gstaudioclock.c: This clock can be slaved to a master clock now.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
This clock can be slaved to a master clock now.

* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Handle slaving the internal clock to the clock selected in the
pipeline.
Add property to make the basesink not provide a clock.

* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_wait):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We can use the clock in GstElement, no need to store it ourselves.
2005-11-22 18:32:09 +00:00
Wim Taymans
b17856db22 Fix sync again. Moved sample alignment to basesink.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (audioringbuffer_thread_func),
(gst_audioringbuffer_stop):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop),
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
(gst_ring_buffer_commit), (gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Fix sync again. Moved sample alignment to basesink.
2005-09-24 13:06:03 +00:00
Tim-Philipp Müller
b9b56ce7d3 gst-libs/gst/: Add padding (you will need to rebuild gst-plugins-base, gst-plugins and all applications afterwards!)
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.h:
* gst-libs/gst/net/gstnetbuffer.h:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Add padding (you will need to rebuild gst-plugins-base,
gst-plugins and all applications afterwards!)
2005-08-09 17:29:40 +00:00
Thomas Vander Stichele
9e8a11d3ce use overridable ERROR_CFLAGS; more macro splitting
Original commit message from CVS:
use overridable ERROR_CFLAGS; more macro splitting
2005-07-10 12:03:58 +00:00
Wim Taymans
2e2623748d gst-libs/gst/audio/: Fix compilation error.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
(gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ringbuffer_set_callback):
Fix compilation error.
Ringbuffer starts out as not running.
Free our clock in dispose.
When releasing the ringbuffer we need to renegotiate so
clear the pad caps.
2005-06-29 11:17:33 +00:00
Wim Taymans
fa8c2eb659 Make the base audiosink return an error when there is no audiobuffer negotiated.
Original commit message from CVS:
Make the base audiosink return an error when there is no
audiobuffer negotiated.
2005-05-06 16:18:24 +00:00
Wim Taymans
235ea5989c Make ringbuffer faster and more simple by removing the locks in the playback thread.
Original commit message from CVS:
Make ringbuffer faster and more simple by removing the locks
in the playback thread.
Add sample accurate playback based on buffer sample offsets.
Make the baseaudiosink provide a clock.
Parse caps in the base class.
Correctly handle seeking, flushing and state changes.
2005-04-28 16:15:42 +00:00
Wim Taymans
5a3941c762 An attempt at a set of audio base classes together with some design docs.
Original commit message from CVS:
* docs/design-audiosinks.txt:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_dispose), (gst_audioringbuffer_finalize),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_base_init),
(gst_audiosink_class_init), (gst_audiosink_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstaudiosink.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_base_init), (gst_baseaudiosink_class_init),
(gst_baseaudiosink_init), (gst_baseaudiosink_set_property),
(gst_baseaudiosink_get_property), (gst_baseaudiosink_setcaps),
(gst_baseaudiosink_get_times), (gst_baseaudiosink_event),
(gst_baseaudiosink_preroll), (gst_baseaudiosink_render),
(gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_class_init), (gst_ringbuffer_init),
(gst_ringbuffer_dispose), (gst_ringbuffer_finalize),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_play_unlocked),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_resume), (gst_ringbuffer_stop),
(gst_ringbuffer_callback), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
An attempt at a set of audio base classes together with some
design docs.
2005-04-20 10:19:54 +00:00