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gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
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.
This commit is contained in:
Wim Taymans
parent
531c6fb462
commit
fc523e047c
4 changed files with 76 additions and 41 deletions
25
ChangeLog
25
ChangeLog
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@ -1,3 +1,28 @@
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2008-05-09 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
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Choose to allocate one less segment but require one additional segment
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as latency.
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* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
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No need to increment the number of segments in the source.
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_get_time), (clock_convert_external),
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(gst_base_audio_sink_resample_slaving),
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(gst_base_audio_sink_skew_slaving),
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(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
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(gst_base_audio_sink_async_play):
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Remove adding latency when returning the internal time while subtracting
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it again when we use the value a little later.
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When calculating the end timestamp, we are making a rounding error
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with the current algorithm. Ensure that we don't accumulate these
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rounding errors when aligning samples by not resampling at all if we
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don't need to. Fixes #419351.
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Make the initial calibration of the clock slaving a little more
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predictable and accurate. Also handle the case where we don't do
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clock slaving.
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2008-05-09 Sebastian Dröge <slomo@circular-chaos.org>
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Based on a patch by:
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@ -366,8 +366,8 @@ gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
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if (!result)
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goto could_not_prepare;
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/* allocate one more segment as we need some headroom */
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spec->segtotal++;
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/* set latency to one more segment as we need some headroom */
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spec->seglatency = spec->segtotal + 1;
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buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
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memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
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@ -360,9 +360,6 @@ gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec)
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if (!result)
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goto could_not_open;
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/* allocate one more segment as we need some headroom */
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spec->segtotal++;
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buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize);
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memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data));
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@ -371,7 +371,7 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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{
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guint64 raw, samples;
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guint delay;
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GstClockTime result, us_latency;
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GstClockTime result;
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if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
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return GST_CLOCK_TIME_NONE;
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@ -391,15 +391,9 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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result = gst_util_uint64_scale_int (samples, GST_SECOND,
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sink->ringbuffer->spec.rate);
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/* latency before starting the clock */
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us_latency = sink->priv->us_latency;
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result += us_latency;
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GST_DEBUG_OBJECT (sink,
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"processed samples: raw %llu, delay %u, real %llu, time %"
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GST_TIME_FORMAT ", upstream latency %" GST_TIME_FORMAT, raw, delay,
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samples, GST_TIME_ARGS (result), GST_TIME_ARGS (us_latency));
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GST_TIME_FORMAT, raw, delay, samples, GST_TIME_ARGS (result));
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return result;
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}
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@ -787,8 +781,7 @@ gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
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static GstClockTime
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clock_convert_external (GstClockTime external, GstClockTime cinternal,
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GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom,
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GstClockTime us_latency)
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GstClockTime cexternal, GstClockTime crate_num, GstClockTime crate_denom)
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{
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/* adjust for rate and speed */
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if (external >= cexternal) {
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@ -803,12 +796,6 @@ clock_convert_external (GstClockTime external, GstClockTime cinternal,
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else
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external = 0;
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}
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/* adjust for offset when slaving started */
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if (external > us_latency)
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external -= us_latency;
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else
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external = 0;
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return external;
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}
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@ -838,9 +825,9 @@ gst_base_audio_sink_resample_slaving (GstBaseAudioSink * sink,
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/* bring external time to internal time */
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render_start = clock_convert_external (render_start, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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crate_num, crate_denom);
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render_stop = clock_convert_external (render_stop, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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crate_num, crate_denom);
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GST_DEBUG_OBJECT (sink,
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"after slaving: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
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@ -875,6 +862,9 @@ gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
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etime = etime > cexternal ? etime - cexternal : 0;
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itime = itime > cinternal ? itime - cinternal : 0;
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/* do itime - etime.
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* positive value means external clock goes slower
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* negative value means external clock goes faster */
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skew = GST_CLOCK_DIFF (etime, itime);
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if (sink->priv->avg_skew == -1) {
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/* first observation */
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@ -945,9 +935,9 @@ gst_base_audio_sink_skew_slaving (GstBaseAudioSink * sink,
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/* convert, ignoring speed */
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render_start = clock_convert_external (render_start, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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crate_num, crate_denom);
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render_stop = clock_convert_external (render_stop, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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crate_num, crate_denom);
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*srender_start = render_start;
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*srender_stop = render_stop;
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@ -967,9 +957,9 @@ gst_base_audio_sink_none_slaving (GstBaseAudioSink * sink,
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/* convert, ignoring speed */
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render_start = clock_convert_external (render_start, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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crate_num, crate_denom);
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render_stop = clock_convert_external (render_stop, cinternal, cexternal,
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crate_num, crate_denom, sink->priv->us_latency);
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crate_num, crate_denom);
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*srender_start = render_start;
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*srender_stop = render_stop;
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@ -1153,26 +1143,34 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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render_stop = gst_util_uint64_scale_int (render_stop,
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ringbuf->spec.rate, GST_SECOND);
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/* positive playback rate, first sample is render_start, negative rate, first
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* sample is render_stop. When no rate conversion is active, render exactly
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* the amount of input samples to avoid aligning to rounding errors. */
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if (bsink->segment.rate >= 0.0) {
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sample_offset = render_start;
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if (bsink->segment.rate == 1.0)
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render_stop = sample_offset + samples;
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} else {
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sample_offset = render_stop;
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if (bsink->segment.rate == -1.0)
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render_start = sample_offset + samples;
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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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/* resync when we don't know what to align the sample with */
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if (G_UNLIKELY (sink->next_sample == -1)) {
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GST_DEBUG_OBJECT (sink,
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"no align possible: no previous sample position known");
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goto no_align;
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}
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/* positive playback rate, first sample is render_start, negative rate, first
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* sample is render_stop */
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if (bsink->segment.rate >= 0.0)
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sample_offset = render_start;
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else
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sample_offset = render_stop;
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/* now try to align the sample to the previous one */
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/* now try to align the sample to the previous one, first see how big the
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* difference is. */
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if (sample_offset >= sink->next_sample)
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diff = sample_offset - sink->next_sample;
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else
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@ -1422,12 +1420,22 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink)
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/* if we are slaved to a clock, we need to set the initial
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* calibration */
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/* get external and internal time to set as calibration params */
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etime = gst_clock_get_time (clock);
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itime = gst_clock_get_internal_time (sink->provided_clock);
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sink->priv->avg_skew = -1;
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sink->next_sample = -1;
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etime = GST_ELEMENT_CAST (sink)->base_time;
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itime = gst_base_audio_sink_get_time (sink->provided_clock, sink);
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switch (sink->priv->slave_method) {
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case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
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case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
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/* adjust with upstream latency, when we are prerolled, our internal clock
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* should exactly have been the time of the upstream latency */
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etime += sink->priv->us_latency;
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break;
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case GST_BASE_AUDIO_SINK_SLAVE_NONE:
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/* no slaving, base_time corresponds to our 0 time */
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itime = 0;
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default:
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break;
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}
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GST_DEBUG_OBJECT (sink,
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"internal time: %" GST_TIME_FORMAT " external time: %" GST_TIME_FORMAT,
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GST_TIME_ARGS (itime), GST_TIME_ARGS (etime));
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@ -1439,14 +1447,19 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink)
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switch (sink->priv->slave_method) {
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case GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE:
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/* only set as master if we need to resample */
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/* only set as master when we are resampling */
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GST_DEBUG_OBJECT (sink, "Setting clock as master");
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gst_clock_set_master (sink->provided_clock, clock);
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break;
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case GST_BASE_AUDIO_SINK_SLAVE_SKEW:
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case GST_BASE_AUDIO_SINK_SLAVE_NONE:
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default:
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break;
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}
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sink->priv->avg_skew = -1;
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sink->next_sample = -1;
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/* start ringbuffer so we can start slaving right away when we need to */
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gst_ring_buffer_start (sink->ringbuffer);
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