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Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_resample_slaving), (gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render), (gst_base_audio_sink_async_play), (gst_base_audio_sink_change_state): Change the way in which the ringbuffer is started when dealing with a slaved clock and latency. We now sync to the clock until we reach upstream latency before starting the ringbuffer. This has the effect that we can accurately align the master and slave clocks and let the rate correction code take care of the initial drift or rounding errors instead of leaving them uncorrected with the old approach. |
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