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gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
This commit is contained in:
Wim Taymans 2008-06-20 09:09:37 +00:00
parent 6391d87d09
commit d2f328f55b
2 changed files with 60 additions and 19 deletions
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11
ChangeLog
View file

@ -1,3 +1,14 @@
2008-06-20 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_render):
Report latency even if we are not live instead of hiding it.
Take ts-offset and render-delay of the basesink into account when
scheduling samples.
Rework the clipping code so that we can take the various offsets into
account and still do correct clipping.
2008-06-20 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:

68
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -351,8 +351,8 @@ gst_base_audio_sink_query (GstElement * element, GstQuery * query)
} else {
GST_DEBUG_OBJECT (basesink,
"peer or we are not live, don't care about latency");
min_latency = 0;
max_latency = -1;
min_latency = min_l;
max_latency = max_l;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
@ -1146,6 +1146,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstClockTimeDiff sync_offset, ts_offset;
GstBaseAudioSink *sink;
GstRingBuffer *ringbuf;
gint64 diff, align, ctime, cstop;
@ -1155,10 +1156,11 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
gint bps;
gint accum;
gint out_samples;
GstClockTime base_time = GST_CLOCK_TIME_NONE, latency;
GstClockTime base_time, render_delay, latency;
GstClock *clock;
gboolean sync, slaved, align_next;
GstFlowReturn ret;
GstSegment clip_seg;
sink = GST_BASE_AUDIO_SINK (bsink);
@ -1172,6 +1174,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
* that we can align the first sample of the ringbuffer to the base_time +
* latency. */
GST_OBJECT_LOCK (sink);
base_time = GST_ELEMENT_CAST (sink)->base_time;
if (G_UNLIKELY (sink->priv->sync_latency)) {
/* only do this once until we are set back to PLAYING */
sink->priv->sync_latency = FALSE;
@ -1194,8 +1197,6 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
in_offset = GST_BUFFER_OFFSET (buf);
time = GST_BUFFER_TIMESTAMP (buf);
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
ringbuf->spec.rate);
GST_DEBUG_OBJECT (sink,
"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT
@ -1212,16 +1213,48 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
GST_DEBUG_OBJECT (sink,
"Buffer of size %u has no time. Using render_start=%" G_GUINT64_FORMAT,
GST_BUFFER_SIZE (buf), render_start);
/* we don't have a start so we don't know stop either */
stop = -1;
goto no_sync;
}
/* let's calc stop based on the number of samples in the buffer instead
* of trusting the DURATION */
stop = time + gst_util_uint64_scale_int (samples, GST_SECOND,
ringbuf->spec.rate);
/* prepare the clipping segment. Since we will be subtracting ts-offset and
* device-delay later we scale the start and stop with those values so that we
* can correctly clip them */
clip_seg.format = GST_FORMAT_TIME;
clip_seg.start = bsink->segment.start;
clip_seg.stop = bsink->segment.stop;
clip_seg.duration = -1;
/* the sync offset is the combination of ts-offset and device-delay */
latency = gst_base_sink_get_latency (bsink);
ts_offset = gst_base_sink_get_ts_offset (bsink);
render_delay = gst_base_sink_get_render_delay (bsink);
sync_offset = ts_offset - render_delay + latency;
GST_DEBUG_OBJECT (sink,
"sync-offset %" G_GINT64_FORMAT ", render-delay %" GST_TIME_FORMAT
", ts-offset %" G_GINT64_FORMAT, sync_offset,
GST_TIME_ARGS (render_delay), ts_offset);
/* compensate for ts-offset and device-delay when negative we need to
* clip. */
if (sync_offset < 0) {
clip_seg.start += -sync_offset;
if (clip_seg.stop != -1)
clip_seg.stop += -sync_offset;
}
/* samples should be rendered based on their timestamp. All samples
* arriving before the segment.start or after segment.stop are to be
* thrown away. All samples should also be clipped to the segment
* boundaries */
/* let's calc stop based on the number of samples in the buffer instead
* of trusting the DURATION */
if (!gst_segment_clip (&bsink->segment, GST_FORMAT_TIME, time, stop, &ctime,
if (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop, &ctime,
&cstop))
goto out_of_segment;
@ -1271,26 +1304,23 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bsink));
GST_DEBUG_OBJECT (sink, "base_time %" GST_TIME_FORMAT,
GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time));
/* add base time to sync against the clock */
render_start += base_time;
render_stop += base_time;
/* compensate for latency */
latency = gst_base_sink_get_latency (bsink);
/* compensate for ts-offset and delay we know this will not underflow because we
* clipped above. */
GST_DEBUG_OBJECT (sink,
"compensating for latency %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
/* add latency to get the timestamp to sync against the pipeline clock */
render_start += latency;
render_stop += latency;
"compensating for sync-offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (sync_offset));
render_start += sync_offset;
render_stop += sync_offset;
GST_DEBUG_OBJECT (sink,
"after latency: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
"after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
if ((slaved = clock != sink->provided_clock)) {