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gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_setcaps), (gst_base_audio_sink_event), (gst_base_audio_sink_preroll), (gst_base_audio_sink_render): Improve debugging. Post error when caps cannot be parsed. Resync on discontinuity in the stream. Clip samples to segment boundaries. return WRONG_STATE sooner when we are flushing. * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init), (gst_base_audio_src_get_time), (gst_base_audio_src_create): Make audiosrc operate in TIME. Set TIMESTAMP and DURATION on buffers.
This commit is contained in:
Wim Taymans
parent
8eb4e006a2
commit
2bc5ca1786
16
ChangeLog
16
ChangeLog
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@ -1,3 +1,19 @@
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2006-01-25 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
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(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
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Improve debugging.
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Post error when caps cannot be parsed.
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Resync on discontinuity in the stream.
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Clip samples to segment boundaries.
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return WRONG_STATE sooner when we are flushing.
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* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
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(gst_base_audio_src_get_time), (gst_base_audio_src_create):
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Make audiosrc operate in TIME.
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Set TIMESTAMP and DURATION on buffers.
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2006-01-24 Tim-Philipp Müller <tim at centricular dot net>
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* tests/examples/seek/seek.c: (main):
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@ -262,12 +262,12 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
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spec = &sink->ringbuffer->spec;
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GST_DEBUG ("release old ringbuffer");
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GST_DEBUG_OBJECT (sink, "release old ringbuffer");
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/* release old ringbuffer */
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gst_ring_buffer_release (sink->ringbuffer);
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GST_DEBUG ("parse caps");
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GST_DEBUG_OBJECT (sink, "parse caps");
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spec->buffer_time = sink->buffer_time;
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spec->latency_time = sink->latency_time;
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@ -278,7 +278,7 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
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gst_ring_buffer_debug_spec_buff (spec);
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GST_DEBUG ("acquire new ringbuffer");
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GST_DEBUG_OBJECT (sink, "acquire new ringbuffer");
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if (!gst_ring_buffer_acquire (sink->ringbuffer, spec))
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goto acquire_error;
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@ -297,12 +297,14 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
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/* ERRORS */
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parse_error:
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{
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GST_DEBUG ("could not parse caps");
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GST_DEBUG_OBJECT (sink, "could not parse caps");
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GST_ELEMENT_ERROR (sink, STREAM, FORMAT,
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("cannot parse audio format."), ("cannot parse audio format."));
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return FALSE;
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}
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acquire_error:
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{
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GST_DEBUG ("could not acquire ringbuffer");
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GST_DEBUG_OBJECT (sink, "could not acquire ringbuffer");
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return FALSE;
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}
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}
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@ -332,7 +334,9 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
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gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
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break;
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case GST_EVENT_EOS:
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/* need to start playback when we reach EOS */
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gst_ring_buffer_start (sink->ringbuffer);
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/* now wait till we played everything */
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break;
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default:
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break;
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@ -355,7 +359,7 @@ gst_base_audio_sink_preroll (GstBaseSink * bsink, GstBuffer * buffer)
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wrong_state:
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{
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GST_DEBUG ("ringbuffer in wrong state");
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GST_DEBUG_OBJECT (sink, "ringbuffer in wrong state");
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GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
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("sink not negotiated."), ("sink not negotiated."));
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return GST_FLOW_NOT_NEGOTIATED;
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@ -396,11 +400,10 @@ 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, render_time, duration;
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GstClockTimeDiff render_diff;
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GstClockTime time, stop, render_time, duration;
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GstBaseAudioSink *sink;
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GstRingBuffer *ringbuf;
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gint64 diff;
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gint64 diff, ctime, cstop;
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guint8 *data;
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guint size;
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guint samples;
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@ -412,6 +415,11 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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sink = GST_BASE_AUDIO_SINK (bsink);
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if (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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@ -431,80 +439,109 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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duration = GST_BUFFER_DURATION (buf);
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data = GST_BUFFER_DATA (buf);
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GST_DEBUG ("time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT,
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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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/* if not valid timestamp or we don't need to sync, try to play
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* sample ASAP */
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if (!GST_CLOCK_TIME_IS_VALID (time) || !bsink->sync) {
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render_offset = gst_base_audio_sink_get_offset (sink);
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GST_DEBUG ("Buffer of size %u has no time. Using render_offset=%"
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G_GUINT64_FORMAT, GST_BUFFER_SIZE (buf), render_offset);
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stop = -1;
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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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goto no_sync;
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}
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render_diff = time - bsink->segment.start;
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/* samples should be rendered based on their timestamp. All samples
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* arriving before the segment.start are to be thrown away */
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/* FIXME, for now we drop the sample completely, we should
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* in fact clip the sample. Same for the segment.stop, actually. */
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if (render_diff < 0)
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* arriving before the segment.start or after segment.stop are to be
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* thrown away. All samples should also be clipped to the segment
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* boundaries */
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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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/* see if some clipping happened */
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diff = ctime - time;
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if (diff > 0) {
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diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
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GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
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G_GUINT64_FORMAT " samples", GST_TIME_ARGS (ctime), diff);
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samples -= diff;
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data += samples * bps;
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time = ctime;
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}
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diff = stop - cstop;
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if (diff > 0) {
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diff = gst_util_uint64_scale_int (diff, ringbuf->spec.rate, GST_SECOND);
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GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
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G_GUINT64_FORMAT " samples", GST_TIME_ARGS (cstop), diff);
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samples -= diff;
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stop = cstop;
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}
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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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/* bring buffer timestamp to stream time */
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render_time = render_diff;
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/* adjust for rate */
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render_time /= ABS (bsink->segment.rate);
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/* adjust for accumulated segments */
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render_time += bsink->segment.accum;
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/* bring buffer timestamp to running time */
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render_time =
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gst_segment_to_running_time (&bsink->segment, GST_FORMAT_TIME, time);
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/* add base time to get absolute clock time */
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render_time +=
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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 = render_time * ringbuf->spec.rate / GST_SECOND;
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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 %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 (sink->next_sample != -1)
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if (sink->next_sample != -1) {
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diff = ABS ((gint64) render_offset - (gint64) sink->next_sample);
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else
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diff = ringbuf->spec.rate;
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GST_DEBUG ("render time %" GST_TIME_FORMAT
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", render offset %llu, diff %lld, samples %lu",
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GST_TIME_ARGS (render_time), render_offset, diff, samples);
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/* we tollerate a 10th of 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. */
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if (diff < ringbuf->spec.rate / DIFF_TOLERANCE) {
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GST_DEBUG ("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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/* we tollerate a 10th of 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. */
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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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GST_DEBUG_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 ("resync");
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GST_DEBUG_OBJECT (sink, "resync after discont");
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}
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crate = ((gdouble) crate_num) / crate_denom;
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GST_DEBUG_OBJECT (sink,
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"internal %" G_GUINT64_FORMAT ", %" G_GUINT64_FORMAT ", rate %g",
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cinternal, cexternal, crate);
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no_sync:
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/* clip length based on rate */
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samples = MIN (samples, samples / (crate * ABS (bsink->segment.rate)));
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samples = MIN (samples, samples / (crate * bsink->segment.abs_rate));
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/* the next sample should be current sample and its length */
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sink->next_sample = render_offset + samples;
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gst_ring_buffer_commit (ringbuf, render_offset, data, samples);
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samples = gst_ring_buffer_commit (ringbuf, render_offset, data, samples);
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if (samples == -1)
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goto stopping;
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if (GST_CLOCK_TIME_IS_VALID (time) && time + duration >= bsink->segment.stop) {
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GST_DEBUG ("start playback because we are at the end of segment");
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if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
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GST_DEBUG_OBJECT (sink,
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"start playback because we are at the end of segment");
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gst_ring_buffer_start (ringbuf);
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}
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@ -512,26 +549,32 @@ no_sync:
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out_of_segment:
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{
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GST_DEBUG ("dropping sample out of segment time %" GST_TIME_FORMAT
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", start %" GST_TIME_FORMAT,
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GST_TIME_ARGS (time), GST_TIME_ARGS (bsink->segment.start));
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GST_DEBUG_OBJECT (sink,
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"dropping sample out of segment time %" GST_TIME_FORMAT ", start %"
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GST_TIME_FORMAT, GST_TIME_ARGS (time),
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GST_TIME_ARGS (bsink->segment.start));
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return GST_FLOW_OK;
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}
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wrong_state:
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{
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GST_DEBUG ("ringbuffer not negotiated");
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GST_DEBUG_OBJECT (sink, "ringbuffer not negotiated");
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GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
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("sink not negotiated."), ("sink not negotiated."));
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return GST_FLOW_NOT_NEGOTIATED;
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}
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wrong_size:
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{
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GST_DEBUG ("wrong size");
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GST_DEBUG_OBJECT (sink, "wrong size");
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GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
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("sink received buffer of wrong size."),
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("sink received buffer of wrong size."));
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return GST_FLOW_ERROR;
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}
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stopping:
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{
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GST_DEBUG_OBJECT (sink, "ringbuffer is stopping");
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return GST_FLOW_WRONG_STATE;
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}
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}
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GstRingBuffer *
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@ -138,6 +138,8 @@ gst_base_audio_src_init (GstBaseAudioSrc * baseaudiosrc,
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gst_pad_set_fixatecaps_function (GST_BASE_SRC_PAD (baseaudiosrc),
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gst_base_audio_src_fixate);
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gst_base_src_set_format (GST_BASE_SRC (baseaudiosrc), GST_FORMAT_TIME);
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}
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static GstClock *
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@ -161,7 +163,8 @@ gst_base_audio_src_get_time (GstClock * clock, GstBaseAudioSrc * src)
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samples = gst_ring_buffer_samples_done (src->ringbuffer);
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result = samples * GST_SECOND / src->ringbuffer->spec.rate;
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result = gst_util_uint64_scale_int (samples, GST_SECOND,
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src->ringbuffer->spec.rate);
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return result;
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}
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@ -319,11 +322,14 @@ gst_base_audio_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
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guint len, samples;
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guint res;
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guint64 sample;
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GstRingBuffer *ringbuffer;
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if (!gst_ring_buffer_is_acquired (src->ringbuffer))
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ringbuffer = src->ringbuffer;
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if (!gst_ring_buffer_is_acquired (ringbuffer))
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goto wrong_state;
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buf = gst_buffer_new_and_alloc (src->ringbuffer->spec.segsize);
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buf = gst_buffer_new_and_alloc (ringbuffer->spec.segsize);
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data = GST_BUFFER_DATA (buf);
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len = GST_BUFFER_SIZE (buf);
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@ -334,13 +340,17 @@ gst_base_audio_src_create (GstPushSrc * psrc, GstBuffer ** outbuf)
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sample = 0;
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}
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samples = len / src->ringbuffer->spec.bytes_per_sample;
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samples = len / ringbuffer->spec.bytes_per_sample;
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res = gst_ring_buffer_read (src->ringbuffer, sample, data, samples);
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res = gst_ring_buffer_read (ringbuffer, sample, data, samples);
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if (res == -1)
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goto stopped;
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GST_BUFFER_TIMESTAMP (buf) = gst_util_uint64_scale_int (sample,
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GST_SECOND, ringbuffer->spec.rate);
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src->next_sample = sample + samples;
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GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (src->next_sample,
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GST_SECOND, ringbuffer->spec.rate) - GST_BUFFER_TIMESTAMP (buf);
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gst_buffer_set_caps (buf, GST_PAD_CAPS (GST_BASE_SRC_PAD (psrc)));
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