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fc523e047c
Original commit message from CVS: * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire): Choose to allocate one less segment but require one additional segment as latency. * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire): No need to increment the number of segments in the source. * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_get_time), (clock_convert_external), (gst_base_audio_sink_resample_slaving), (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render), (gst_base_audio_sink_async_play): Remove adding latency when returning the internal time while subtracting it again when we use the value a little later. When calculating the end timestamp, we are making a rounding error with the current algorithm. Ensure that we don't accumulate these rounding errors when aligning samples by not resampling at all if we don't need to. Fixes #419351. Make the initial calibration of the clock slaving a little more predictable and accurate. Also handle the case where we don't do clock slaving. |
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| .. | ||
| .gitignore | ||
| audio.c | ||
| audio.def | ||
| audio.h | ||
| audio.vcproj | ||
| audiofilter.vcproj | ||
| gstaudioclock.c | ||
| gstaudioclock.h | ||
| gstaudiofilter.c | ||
| gstaudiofilter.h | ||
| gstaudiofiltertemplate.c | ||
| gstaudiosink.c | ||
| gstaudiosink.h | ||
| gstaudiosrc.c | ||
| gstaudiosrc.h | ||
| gstbaseaudiosink.c | ||
| gstbaseaudiosink.h | ||
| gstbaseaudiosrc.c | ||
| gstbaseaudiosrc.h | ||
| gstringbuffer.c | ||
| gstringbuffer.h | ||
| make_filter | ||
| Makefile.am | ||
| mixerutils.c | ||
| mixerutils.h | ||
| multichannel.c | ||
| multichannel.h | ||
| testchannels.c | ||
| TODO | ||