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.
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.
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
Don't try to calculate silence samples, base class does this much
better now.
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
Calculate silence samples correctly.
* gst-libs/gst/audio/gstringbuffer.h:
Add _CAST macro.
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
add remaining symbols into correct setions
* gst-libs/gst/audio/gstringbuffer.c:
fix incomplete docs
* gst-libs/gst/audio/gstringbuffer.h:
comment out not yet implemented function
* gst-libs/gst/floatcast/floatcast.h:
* gst-libs/gst/netbuffer/gstnetbuffer.c:
add short descriptions
* gst-libs/gst/interfaces/propertyprobe.c:
fix return value docs
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
simplify debug logging
* gst-libs/gst/riff/riff-read.h:
sync function prototype and docs
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
remove left over symbol
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.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix#330748.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes#331903)
Handle EOS correctly.
Added docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Name (private) union, makes Forte compiler happy (this time
for real) (#324900).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Sun's Forte compiler doesn't seem to like anonymous structs,
so use same setup as in GstBaseSrc (fixes#324900).
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated TODO
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release):
Small cleanups.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Slave to the master clock when going to PLAYING and unslave when
going to PAUSED.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_advance):
* gst-libs/gst/audio/gstringbuffer.h:
Add some docs and cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.h:
Don't break ABI.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
Some more comments.
Handle missing required caps fields better.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_get_offset),
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (wait_segment), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Add flushing mode to the ringbuffer so that it in all cases does
not try to handle more audio. This makes sure it does not try to
block anymore when flushing and fixes a livelock.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all):
* gst-libs/gst/audio/gstringbuffer.h:
Added function to clear the ringbuffer.
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.
* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.
* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.
* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.
* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
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.
