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gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
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.
This commit is contained in:
Wim Taymans
parent
410bb3fef1
commit
0990cbf274
5 changed files with 162 additions and 151 deletions
13
ChangeLog
13
ChangeLog
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@ -1,3 +1,16 @@
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2006-11-13 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_event), (gst_base_audio_sink_render):
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* gst-libs/gst/audio/gstbaseaudiosink.h:
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Make the clock sync code more accurate wrt resampling and playback
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at different rates.
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* gst-libs/gst/audio/gstringbuffer.c:
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(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
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* gst-libs/gst/audio/gstringbuffer.h:
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Use better algorithm to interpolate sample rates.
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2006-11-13 Michael Smith <msmith@fluendo.com>
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
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@ -444,27 +444,12 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
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case GST_EVENT_NEWSEGMENT:
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{
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gdouble rate;
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GValue src = { 0 };
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GValue dst = { 0 };
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/* we only need the rate */
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gst_event_parse_new_segment_full (event, NULL, &rate, NULL, NULL,
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NULL, NULL, NULL);
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g_value_init (&src, G_TYPE_DOUBLE);
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g_value_set_double (&src, rate);
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g_value_init (&dst, GST_TYPE_FRACTION);
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g_value_transform (&src, &dst);
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g_value_unset (&src);
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sink->abidata.ABI.rate_num = gst_value_get_fraction_numerator (&dst);
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sink->abidata.ABI.rate_denom = gst_value_get_fraction_denominator (&dst);
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sink->abidata.ABI.rate_accum = 0;
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GST_DEBUG_OBJECT (sink, "set rate to %f = %d / %d", rate,
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sink->abidata.ABI.rate_num, sink->abidata.ABI.rate_denom);
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g_value_unset (&dst);
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GST_DEBUG_OBJECT (sink, "new rate of %f", rate);
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break;
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}
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default:
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@ -530,19 +515,19 @@ gst_base_audio_sink_get_offset (GstBaseAudioSink * sink)
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static GstFlowReturn
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gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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{
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guint64 render_offset, in_offset;
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GstClockTime time, stop, render_time, duration;
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guint64 in_offset, clock_offset;
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GstClockTime time, stop, render_start, render_stop, sample_offset;
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GstBaseAudioSink *sink;
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GstRingBuffer *ringbuf;
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gint64 diff, ctime, cstop;
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gint64 diff, align, ctime, cstop;
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guint8 *data;
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guint size;
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guint samples, written;
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gint bps;
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gint accum;
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GstClockTime crate_num;
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GstClockTime crate_denom;
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gint rate_num;
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gint rate_denom;
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gint out_samples;
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GstClockTime cinternal, cexternal;
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GstClock *clock;
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gboolean sync;
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@ -562,27 +547,28 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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goto wrong_size;
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samples = size / bps;
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out_samples = samples;
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in_offset = GST_BUFFER_OFFSET (buf);
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time = GST_BUFFER_TIMESTAMP (buf);
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duration = GST_BUFFER_DURATION (buf);
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data = GST_BUFFER_DATA (buf);
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stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
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ringbuf->spec.rate);
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GST_DEBUG_OBJECT (sink,
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"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT,
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GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->segment.start));
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"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT
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", samples %u", GST_TIME_ARGS (time), in_offset,
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GST_TIME_ARGS (bsink->segment.start), samples);
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rate_num = sink->abidata.ABI.rate_num;
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rate_denom = sink->abidata.ABI.rate_denom;
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data = GST_BUFFER_DATA (buf);
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/* if not valid timestamp or we can't clip or sync, try to play
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* sample ASAP */
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if (!GST_CLOCK_TIME_IS_VALID (time)) {
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render_offset = gst_base_audio_sink_get_offset (sink);
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stop = -1;
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render_start = gst_base_audio_sink_get_offset (sink);
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render_stop = render_start + samples;
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GST_DEBUG_OBJECT (sink,
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"Buffer of size %u has no time. Using render_offset=%" G_GUINT64_FORMAT,
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GST_BUFFER_SIZE (buf), render_offset);
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"Buffer of size %u has no time. Using render_start=%" G_GUINT64_FORMAT,
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GST_BUFFER_SIZE (buf), render_start);
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goto no_sync;
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}
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@ -592,9 +578,6 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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* boundaries */
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/* let's calc stop based on the number of samples in the buffer instead
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* of trusting the DURATION */
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stop =
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time + gst_util_uint64_scale_int (samples, GST_SECOND,
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ringbuf->spec.rate);
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if (!gst_segment_clip (&bsink->segment, GST_FORMAT_TIME, time, stop, &ctime,
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&cstop))
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goto out_of_segment;
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@ -628,45 +611,45 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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if (!sync) {
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/* no sync needed, play sample ASAP */
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render_offset = gst_base_audio_sink_get_offset (sink);
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stop = -1;
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render_start = gst_base_audio_sink_get_offset (sink);
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render_stop = render_start + samples;
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GST_DEBUG_OBJECT (sink,
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"no sync needed. Using render_offset=%" G_GUINT64_FORMAT,
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render_offset);
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"no sync needed. Using render_start=%" G_GUINT64_FORMAT, render_start);
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goto no_sync;
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}
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/* bring buffer timestamp to running time */
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render_time =
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/* bring buffer start and stop times to running time */
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render_start =
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gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
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GST_DEBUG_OBJECT (sink, "running time %" GST_TIME_FORMAT,
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GST_TIME_ARGS (render_time));
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render_stop =
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gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, stop);
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GST_DEBUG_OBJECT (sink,
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"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
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GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
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/* get calibration parameters to compensate for speed and offset differences
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* when we are slaved */
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gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
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&crate_num, &crate_denom);
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/* add base time to get absolute clock time */
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render_time +=
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clock_offset =
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(gst_element_get_base_time (GST_ELEMENT_CAST (bsink)) - cexternal) +
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cinternal;
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/* and bring the time to the offset in the buffer */
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render_offset =
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gst_util_uint64_scale_int (render_time, ringbuf->spec.rate, GST_SECOND);
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GST_DEBUG_OBJECT (sink, "render time %" GST_TIME_FORMAT
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", render offset %" G_GUINT64_FORMAT ", samples %u",
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GST_TIME_ARGS (render_time), render_offset, samples);
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GST_DEBUG_OBJECT (sink, "clock offset %" GST_TIME_FORMAT " %" G_GUINT64_FORMAT
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"/%" G_GUINT64_FORMAT, GST_TIME_ARGS (clock_offset), crate_num,
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crate_denom);
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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 (clock != 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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/* and bring the time to the rate corrected offset in the buffer */
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render_start = gst_util_uint64_scale_int (render_start + clock_offset,
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ringbuf->spec.rate, GST_SECOND);
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render_stop = gst_util_uint64_scale_int (render_stop + clock_offset,
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ringbuf->spec.rate, GST_SECOND);
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GST_DEBUG_OBJECT (sink,
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"render: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
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GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
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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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@ -680,11 +663,16 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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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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if (render_offset >= sink->next_sample)
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diff = render_offset - sink->next_sample;
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if (bsink->segment.rate >= 1.0)
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sample_offset = render_start;
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else
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diff = sink->next_sample - render_offset;
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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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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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diff = sink->next_sample - sample_offset;
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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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@ -694,8 +682,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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GST_DEBUG_OBJECT (sink,
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"align with prev sample, %" G_GINT64_FORMAT " < %d", 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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/* calc align with previous sample */
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align = sink->next_sample - sample_offset;
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} else {
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/* bring sample diff to seconds for error message */
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diff = gst_util_uint64_scale_int (diff, GST_SECOND, ringbuf->spec.rate);
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@ -706,50 +694,41 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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("Unexpected discontinuity in audio timestamps of more "
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"than half a second (%" GST_TIME_FORMAT "), resyncing",
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GST_TIME_ARGS (diff)));
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align = 0;
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}
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/* apply alignment */
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render_start += align;
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/* only align stop if we are not slaved */
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if (clock != sink->provided_clock) {
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GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
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goto no_align;
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}
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render_stop += align;
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no_align:
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/* check if we have a clock rate adjustment to do */
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if (crate_num != 1 || crate_num != 1) {
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gint64 num, denom;
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/* make clock rate fit in int */
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while ((crate_num | crate_denom) > G_MAXINT) {
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crate_num /= 2;
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crate_denom /= 2;
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}
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/* full 64bit multiplication */
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num = ((gint64) crate_denom) * rate_num;
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denom = ((gint64) crate_num) * rate_denom;
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/* make result fit in int again */
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while ((num | denom) > G_MAXINT) {
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num /= 2;
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denom /= 2;
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}
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rate_num = num;
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rate_denom = denom;
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GST_DEBUG_OBJECT (sink,
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"clock rate: internal %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT
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" %d/%d", cinternal, cexternal, rate_num, rate_denom);
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}
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/* number of target samples is difference between start and stop */
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out_samples = render_stop - render_start;
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no_sync:
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GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT " %d/%d",
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sink->next_sample, samples, out_samples);
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/* the next sample should be current sample and its length, this will be
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* updated as we write samples to the ringbuffer. */
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sink->next_sample = render_offset;
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GST_DEBUG_OBJECT (sink, "rendering at %" G_GUINT64_FORMAT, sink->next_sample);
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/* we render the first or last sample first, depending on the rate */
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if (bsink->segment.rate >= 1.0)
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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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/* we need to accumulate over different runs for when we get interrupted */
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accum = 0;
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do {
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written =
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gst_ring_buffer_commit_full (ringbuf, &sink->next_sample, data, samples,
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rate_num, rate_denom, &sink->abidata.ABI.rate_accum);
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gst_ring_buffer_commit_full (ringbuf, &sample_offset, data, samples,
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out_samples, &accum);
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GST_DEBUG_OBJECT (sink, "wrote %u of %u, resampled %" G_GUINT64_FORMAT,
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written, samples, sink->next_sample - render_offset);
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GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
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/* if we wrote all, we're done */
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if (written == samples)
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break;
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@ -762,6 +741,8 @@ no_sync:
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data += written * bps;
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} while (TRUE);
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sink->next_sample = sample_offset;
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GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
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sink->next_sample);
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@ -106,11 +106,6 @@ struct _GstBaseAudioSink {
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/*< private >*/
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union {
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struct {
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gint rate_num;
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gint rate_denom;
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gint rate_accum;
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} ABI;
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/* adding + 0 to mark ABI change to be undone later */
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gpointer _gst_reserved[GST_PADDING + 0];
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} abidata;
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|
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@ -1181,77 +1181,84 @@ G_STMT_START { \
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/* simple copy */ \
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if (!skip) \
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memcpy (d, s, towrite); \
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*sample += towrite / bps; \
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in_samples -= towrite / bps; \
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out_samples -= towrite / bps; \
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s += towrite; \
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GST_DEBUG ("copy %u bytes", towrite); \
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} G_STMT_END
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/* in_samples >= out_samples, rate > 1.0 */
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#define FWD_UP_SAMPLES(s,se,d,de) \
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G_STMT_START { \
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guint8 *ds = d; \
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guint8 *sb = s, *db = d; \
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while (s <= se && d < de) { \
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if (!skip) \
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memcpy (d, s, bps); \
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s += bps; \
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*accum += denom; \
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while (*accum > 0) { \
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*accum -= num; \
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*accum += outr; \
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if ((*accum << 1) >= inr) { \
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*accum -= inr; \
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d += bps; \
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} \
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} \
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*sample += (d - ds) / bps; \
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GST_DEBUG ("fwd_up %u bytes", d - ds); \
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in_samples -= (s - sb)/bps; \
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out_samples -= (d - db)/bps; \
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GST_DEBUG ("fwd_up end %d/%d",*accum,*toprocess); \
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} G_STMT_END
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/* out_samples > in_samples, for rates smaller than 1.0 */
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#define FWD_DOWN_SAMPLES(s,se,d,de) \
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G_STMT_START { \
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guint8 *ds = d; \
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guint8 *sb = s, *db = d; \
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while (s <= se && d < de) { \
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if (!skip) \
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memcpy (d, s, bps); \
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d += bps; \
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*accum -= num; \
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while (*accum < 0) { \
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*accum += denom; \
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*accum += inr; \
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if ((*accum << 1) >= outr) { \
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*accum -= outr; \
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s += bps; \
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} \
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} \
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*sample += (d - ds) / bps; \
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GST_DEBUG ("fwd_down %u bytes", d - ds); \
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in_samples -= (s - sb)/bps; \
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out_samples -= (d - db)/bps; \
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GST_DEBUG ("fwd_down end %d/%d",*accum,*toprocess); \
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} G_STMT_END
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#define REV_UP_SAMPLES(s,se,d,de) \
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G_STMT_START { \
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guint8 *ds = d; \
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guint8 *sb = se, *db = d; \
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while (s <= se && d < de) { \
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if (!skip) \
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memcpy (d, se, bps); \
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se -= bps; \
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*accum += denom; \
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while (*accum > 0) { \
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*accum += num; \
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*accum += outr; \
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while ((*accum << 1) >= inr) { \
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*accum -= inr; \
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d += bps; \
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} \
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} \
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*sample += (d - ds) / bps; \
|
||||
GST_DEBUG ("rev_up %u bytes", d - ds); \
|
||||
in_samples -= (sb - se)/bps; \
|
||||
out_samples -= (d - db)/bps; \
|
||||
GST_DEBUG ("rev_up end %d/%d",*accum,*toprocess); \
|
||||
} G_STMT_END
|
||||
|
||||
#define REV_DOWN_SAMPLES(s,se,d,de) \
|
||||
G_STMT_START { \
|
||||
guint8 *ds = d; \
|
||||
guint8 *sb = se, *db = d; \
|
||||
while (s <= se && d < de) { \
|
||||
if (!skip) \
|
||||
memcpy (d, se, bps); \
|
||||
d += bps; \
|
||||
*accum += num; \
|
||||
while (*accum < 0) { \
|
||||
*accum += denom; \
|
||||
*accum += inr; \
|
||||
while ((*accum << 1) >= outr) { \
|
||||
*accum -= outr; \
|
||||
se -= bps; \
|
||||
} \
|
||||
} \
|
||||
*sample += (d - ds) / bps; \
|
||||
GST_DEBUG ("rev_down %u bytes", d - ds); \
|
||||
in_samples -= (sb - se)/bps; \
|
||||
out_samples -= (d - db)/bps; \
|
||||
GST_DEBUG ("rev_down end %d/%d",*accum,*toprocess); \
|
||||
} G_STMT_END
|
||||
|
||||
/**
|
||||
|
|
@ -1259,9 +1266,8 @@ G_STMT_START { \
|
|||
* @buf: the #GstRingBuffer to commit
|
||||
* @sample: the sample position of the data
|
||||
* @data: the data to commit
|
||||
* @len: the number of samples in the data to commit
|
||||
* @num: the numerator of the speed
|
||||
* @denom: the denominator of the speed
|
||||
* @in_samples: the number of samples in the data to commit
|
||||
* @out_sampled: the number of samples to write to the ringbuffer
|
||||
* @accum: accumulator for rate conversion.
|
||||
*
|
||||
* Commit @len samples pointed to by @data to the ringbuffer @buf.
|
||||
|
|
@ -1276,9 +1282,6 @@ G_STMT_START { \
|
|||
* @len does not need to be a multiple of the segment size of the ringbuffer (if
|
||||
* @num and @denom are both 1) although it is recommended for optimal performance.
|
||||
*
|
||||
* @sample will be updated with the next position in the ringbuffer. This
|
||||
* position will take into account any resampling done when writing the samples.
|
||||
*
|
||||
* Returns: The number of samples written to the ringbuffer or -1 on error. The
|
||||
* number of samples written can be less than @len when @buf was interrupted
|
||||
* with a flush or stop.
|
||||
|
|
@ -1289,28 +1292,43 @@ G_STMT_START { \
|
|||
*/
|
||||
guint
|
||||
gst_ring_buffer_commit_full (GstRingBuffer * buf, guint64 * sample,
|
||||
guchar * data, guint len, gint num, gint denom, gint * accum)
|
||||
guchar * data, gint in_samples, gint out_samples, gint * accum)
|
||||
{
|
||||
gint segdone;
|
||||
gint segsize, segtotal, bps, sps;
|
||||
guint8 *dest, *data_end;
|
||||
gint writeseg, sampleoff;
|
||||
gint *toprocess;
|
||||
gint inr, outr;
|
||||
gboolean reverse;
|
||||
|
||||
if (G_UNLIKELY (len == 0))
|
||||
if (G_UNLIKELY (in_samples == 0 || out_samples == 0))
|
||||
return 0;
|
||||
|
||||
g_return_val_if_fail (GST_IS_RING_BUFFER (buf), -1);
|
||||
g_return_val_if_fail (buf->data != NULL, -1);
|
||||
g_return_val_if_fail (data != NULL, -1);
|
||||
g_return_val_if_fail (denom != 0, -1);
|
||||
|
||||
dest = GST_BUFFER_DATA (buf->data);
|
||||
segsize = buf->spec.segsize;
|
||||
segtotal = buf->spec.segtotal;
|
||||
bps = buf->spec.bytes_per_sample;
|
||||
sps = buf->samples_per_seg;
|
||||
/* data_end points to the last sample we have to write, not past it. */
|
||||
data_end = data + (bps * (len - 1));
|
||||
|
||||
reverse = out_samples < 0;
|
||||
out_samples = ABS (out_samples);
|
||||
|
||||
if (in_samples >= out_samples)
|
||||
toprocess = &in_samples;
|
||||
else
|
||||
toprocess = &out_samples;
|
||||
|
||||
inr = in_samples - 1;
|
||||
outr = out_samples - 1;
|
||||
|
||||
/* data_end points to the last sample we have to write, not past it. This is
|
||||
* needed to properly handle reverse playback: it points to the last sample. */
|
||||
data_end = data + (bps * inr);
|
||||
|
||||
/* figure out the segment and the offset inside the segment where
|
||||
* the first sample should be written. */
|
||||
|
|
@ -1318,7 +1336,7 @@ gst_ring_buffer_commit_full (GstRingBuffer * buf, guint64 * sample,
|
|||
sampleoff = (*sample % sps) * bps;
|
||||
|
||||
/* write out all samples */
|
||||
while (data <= data_end) {
|
||||
while (*toprocess > 0) {
|
||||
gint avail;
|
||||
guint8 *d, *d_end;
|
||||
gint ws;
|
||||
|
|
@ -1359,26 +1377,27 @@ gst_ring_buffer_commit_full (GstRingBuffer * buf, guint64 * sample,
|
|||
|
||||
/* we can write now */
|
||||
ws = writeseg % segtotal;
|
||||
avail = segsize - sampleoff;
|
||||
avail = MIN (segsize - sampleoff, bps * out_samples);
|
||||
|
||||
d = dest + (ws * segsize) + sampleoff;
|
||||
d_end = d + avail;
|
||||
*sample += avail / bps;
|
||||
|
||||
GST_DEBUG_OBJECT (buf, "write @%p seg %d, sps %d, off %d, avail %d",
|
||||
dest + ws * segsize, ws, sps, sampleoff, avail);
|
||||
|
||||
if (G_LIKELY (num == 1 && denom == 1)) {
|
||||
if (G_LIKELY (inr == outr && !reverse)) {
|
||||
/* no rate conversion, simply copy samples */
|
||||
FWD_SAMPLES (data, data_end, d, d_end);
|
||||
} else if (num >= 0) {
|
||||
if (num >= denom)
|
||||
} else if (!reverse) {
|
||||
if (inr >= outr)
|
||||
/* forward speed up */
|
||||
FWD_UP_SAMPLES (data, data_end, d, d_end);
|
||||
else
|
||||
/* forward slow down */
|
||||
FWD_DOWN_SAMPLES (data, data_end, d, d_end);
|
||||
} else {
|
||||
if (-num >= denom)
|
||||
if (inr >= outr)
|
||||
/* reverse speed up */
|
||||
REV_UP_SAMPLES (data, data_end, d, d_end);
|
||||
else
|
||||
|
|
@ -1390,9 +1409,11 @@ gst_ring_buffer_commit_full (GstRingBuffer * buf, guint64 * sample,
|
|||
writeseg++;
|
||||
sampleoff = 0;
|
||||
}
|
||||
/* we consumed all samples here */
|
||||
data = data_end + bps;
|
||||
|
||||
done:
|
||||
return (len - 1) - ((data_end - data) / bps);
|
||||
return inr - ((data_end - data) / bps);
|
||||
|
||||
/* ERRORS */
|
||||
not_started:
|
||||
|
|
@ -1422,9 +1443,9 @@ gst_ring_buffer_commit (GstRingBuffer * buf, guint64 sample, guchar * data,
|
|||
guint len)
|
||||
{
|
||||
guint res;
|
||||
guint64 spos = sample;
|
||||
guint64 samplep = sample;
|
||||
|
||||
res = gst_ring_buffer_commit_full (buf, &spos, data, len, 1, 1, NULL);
|
||||
res = gst_ring_buffer_commit_full (buf, &samplep, data, len, len, NULL);
|
||||
|
||||
return res;
|
||||
}
|
||||
|
|
|
|||
|
|
@ -337,9 +337,10 @@ void gst_ring_buffer_clear_all (GstRingBuffer *buf);
|
|||
/* commit samples */
|
||||
guint gst_ring_buffer_commit (GstRingBuffer *buf, guint64 sample,
|
||||
guchar *data, guint len);
|
||||
guint gst_ring_buffer_commit_full (GstRingBuffer *buf, guint64 *sample,
|
||||
guchar *data, guint len,
|
||||
gint num, gint denom, gint *accum);
|
||||
guint gst_ring_buffer_commit_full (GstRingBuffer * buf, guint64 *sample,
|
||||
guchar * data, gint in_samples,
|
||||
gint out_samples, gint * accum);
|
||||
|
||||
/* read samples */
|
||||
guint gst_ring_buffer_read (GstRingBuffer *buf, guint64 sample,
|
||||
guchar *data, guint len);
|
||||
|
|
|
|||
Loading…
Reference in a new issue