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gst/: Segment update fix.
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_init), (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_set_clock), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): * gst/audioresample/gstaudioresample.c: Segment update fix.
This commit is contained in:
Wim Taymans
parent
f405e12b4a
commit
0f2336cff6
3 changed files with 67 additions and 20 deletions
10
ChangeLog
10
ChangeLog
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@ -1,3 +1,13 @@
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2005-11-21 Wim Taymans <wim@fluendo.com>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
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(gst_base_audio_sink_provide_clock),
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(gst_base_audio_sink_set_clock), (gst_base_audio_sink_render),
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(gst_base_audio_sink_change_state):
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* gst/audioresample/gstaudioresample.c:
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Segment update fix.
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2005-11-21 Andy Wingo <wingo@pobox.com>
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* *.h:
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@ -65,6 +65,7 @@ static GstStateChangeReturn gst_base_audio_sink_change_state (GstElement *
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element, GstStateChange transition);
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static GstClock *gst_base_audio_sink_provide_clock (GstElement * elem);
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static void gst_base_audio_sink_set_clock (GstElement * elem, GstClock * clock);
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static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
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GstBaseAudioSink * sink);
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static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data,
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@ -118,6 +119,8 @@ gst_base_audio_sink_class_init (GstBaseAudioSinkClass * klass)
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_change_state);
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gstelement_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_provide_clock);
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gstelement_class->set_clock =
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GST_DEBUG_FUNCPTR (gst_base_audio_sink_set_clock);
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gstbasesink_class->event = GST_DEBUG_FUNCPTR (gst_base_audio_sink_event);
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gstbasesink_class->preroll = GST_DEBUG_FUNCPTR (gst_base_audio_sink_preroll);
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@ -134,10 +137,8 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
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baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
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baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
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baseaudiosink->clock = gst_audio_clock_new ("clock",
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(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
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}
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static void
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@ -166,15 +167,27 @@ gst_base_audio_sink_provide_clock (GstElement * elem)
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sink = GST_BASE_AUDIO_SINK (elem);
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#if 1
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clock = GST_CLOCK_CAST (gst_object_ref (sink->clock));
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#else
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clock = gst_system_clock_obtain ();
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#endif
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return clock;
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}
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static void
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gst_base_audio_sink_set_clock (GstElement * elem, GstClock * clock)
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{
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GstBaseAudioSink *sink;
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sink = GST_BASE_AUDIO_SINK (elem);
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GST_OBJECT_LOCK (sink);
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if (clock != sink->clock) {
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gst_clock_set_master (sink->clock, clock);
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} else {
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gst_clock_set_master (sink->clock, NULL);
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}
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GST_OBJECT_UNLOCK (sink);
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}
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static GstClockTime
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gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
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{
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@ -385,6 +398,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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guint size;
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guint samples;
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gint bps;
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gdouble crate;
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GstClockTime cinternal, cexternal;
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sink = GST_BASE_AUDIO_SINK (bsink);
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@ -408,7 +423,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * 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_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->segment_start));
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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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@ -417,23 +432,30 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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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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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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* 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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* in fact clip the sample. Same for the segment.stop, actually. */
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if (render_diff < 0)
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goto out_of_segment;
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gst_clock_get_calibration (sink->clock, &cinternal, &cexternal, &crate);
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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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/* 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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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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render_time += bsink->segment.accum;
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/* add base time to get absolute clock time */
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render_time += gst_element_get_base_time (GST_ELEMENT_CAST (bsink));
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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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@ -461,14 +483,14 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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no_sync:
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/* clip length based on rate */
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samples = MIN (samples, samples / ABS (bsink->segment_rate));
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samples = MIN (samples, samples / (crate * ABS (bsink->segment.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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if (GST_CLOCK_TIME_IS_VALID (time) && time + duration >= bsink->segment_stop) {
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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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gst_ring_buffer_start (ringbuf);
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}
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@ -479,7 +501,7 @@ 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_TIME_ARGS (time), 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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@ -544,6 +566,23 @@ gst_base_audio_sink_change_state (GstElement * element,
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gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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{
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GstClockTime time;
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gdouble rate;
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time = gst_clock_get_internal_time (sink->clock);
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GST_DEBUG_OBJECT (sink, "time: %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
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gst_clock_get_calibration (sink->clock, NULL, NULL, &rate);
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/* Does not work yet.
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gst_clock_set_calibration (sink->clock,
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time, element->base_time, rate);
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*/
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break;
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}
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
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break;
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default:
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break;
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@ -556,8 +595,6 @@ gst_base_audio_sink_change_state (GstElement * element,
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gst_ring_buffer_pause (sink->ringbuffer);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
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gst_ring_buffer_stop (sink->ringbuffer);
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gst_ring_buffer_release (sink->ringbuffer);
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gst_pad_set_caps (GST_BASE_SINK_PAD (sink), NULL);
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break;
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@ -399,7 +399,7 @@ static GstFlowReturn
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outsize, outsamples);
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GST_BUFFER_OFFSET (outbuf) = audioresample->offset;
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GST_BUFFER_TIMESTAMP (outbuf) = base->segment_start +
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GST_BUFFER_TIMESTAMP (outbuf) = base->segment.start +
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audioresample->offset * GST_SECOND / audioresample->o_rate;
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audioresample->offset += outsamples;
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@ -408,7 +408,7 @@ static GstFlowReturn
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/* we calculate DURATION as the difference between "next" timestamp
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* and current timestamp so we ensure a contiguous stream, instead of
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* having rounding errors. */
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GST_BUFFER_DURATION (outbuf) = base->segment_start +
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GST_BUFFER_DURATION (outbuf) = base->segment.start +
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audioresample->offset * GST_SECOND / audioresample->o_rate -
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GST_BUFFER_TIMESTAMP (outbuf);
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