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gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render): When the audio clock is slaved to another clock, never try to align samples but trust the rate interpolation algorithm.
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19cd03c607
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2 changed files with 42 additions and 26 deletions
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@ -1,3 +1,10 @@
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2006-07-24 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
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When the audio clock is slaved to another clock, never try to align
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samples but trust the rate interpolation algorithm.
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2006-07-24 Wim Taymans <wim@fluendo.com>
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* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
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@ -200,6 +200,7 @@ gst_base_audio_sink_provide_clock (GstElement * elem)
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return clock;
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/* ERRORS */
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wrong_state:
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{
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GST_DEBUG_OBJECT (sink, "ringbuffer not acquired");
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@ -490,11 +491,6 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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sink = GST_BASE_AUDIO_SINK (bsink);
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if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
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/* always resync after a discont */
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sink->next_sample = -1;
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}
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ringbuf = sink->ringbuffer;
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/* can't do anything when we don't have the device */
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@ -581,30 +577,43 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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", render offset %llu, samples %lu",
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GST_TIME_ARGS (render_time), render_offset, samples);
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/* roundoff errors in timestamp conversion */
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if (G_LIKELY (sink->next_sample != -1)) {
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diff = ABS ((gint64) render_offset - (gint64) sink->next_sample);
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/* we tollerate half a second diff before we start resyncing. This
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* should be enough to compensate for various rounding errors in the timestamp
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* and sample offset position. We always resync if we got a discont anyway and
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* non-discont should be aligned by definition. */
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if (diff < ringbuf->spec.rate / DIFF_TOLERANCE) {
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GST_DEBUG_OBJECT (sink,
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"align with prev sample, %" G_GINT64_FORMAT " < %lu", diff,
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ringbuf->spec.rate / DIFF_TOLERANCE);
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/* just align with previous sample then */
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render_offset = sink->next_sample;
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} else {
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/* timestamps drifted apart from previous samples too much, we need to
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* resync. */
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GST_WARNING_OBJECT (sink,
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"resync after discont with previous sample of diff: %lu", diff);
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}
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} else {
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GST_DEBUG_OBJECT (sink, "resync after discont");
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/* never try to align samples when we are slaved to another clock, just
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* trust the rate control algorithm to align the two clocks. We don't take
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* the LOCK to read the clock because it does not really matter here and the
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* clock is not changed while playing normally. */
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if (GST_ELEMENT_CLOCK (sink) != sink->provided_clock) {
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GST_DEBUG_OBJECT (sink, "no align needed: we are slaved");
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goto no_align;
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}
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/* always resync after a discont */
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if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))) {
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GST_DEBUG_OBJECT (sink, "resync after discont");
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goto no_align;
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}
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/* now try to align the sample to the previous one */
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diff = ABS ((gint64) render_offset - (gint64) sink->next_sample);
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/* we tollerate half a second diff before we start resyncing. This
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* should be enough to compensate for various rounding errors in the timestamp
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* and sample offset position. We always resync if we got a discont anyway and
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* non-discont should be aligned by definition. */
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if (diff < ringbuf->spec.rate / DIFF_TOLERANCE) {
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GST_DEBUG_OBJECT (sink,
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"align with prev sample, %" G_GINT64_FORMAT " < %lu", diff,
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ringbuf->spec.rate / DIFF_TOLERANCE);
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/* just align with previous sample then */
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render_offset = sink->next_sample;
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} else {
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/* timestamps drifted apart from previous samples too much, we need to
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* resync. We log this as an element warning. */
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GST_ELEMENT_WARNING (sink, CORE, CLOCK, (NULL),
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("Unexpected discontinuity in audio timestamps of more than half a second"));
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}
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no_align:
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crate =
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gst_guint64_to_gdouble (crate_num) / gst_guint64_to_gdouble (crate_denom);
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GST_DEBUG_OBJECT (sink,
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