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gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
This commit is contained in:
Wim Taymans 2006-09-28 15:08:15 +00:00
parent 6425f71b72
commit 9945d7a468
2 changed files with 28 additions and 4 deletions
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7
ChangeLog
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@ -1,3 +1,10 @@
2006-09-28 Wim Taymans <wim@fluendo.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
When we have a timestamp, we can still perform clipping.
When we have no clock, we must play the sample ASAP.
2006-09-28 Wim Taymans <wim@fluendo.com> 2006-09-28 Wim Taymans <wim@fluendo.com>
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):

25
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -514,6 +514,8 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
GstClockTime crate_num; GstClockTime crate_num;
GstClockTime crate_denom; GstClockTime crate_denom;
GstClockTime cinternal, cexternal; GstClockTime cinternal, cexternal;
GstClock *clock;
gboolean sync;
sink = GST_BASE_AUDIO_SINK (bsink); sink = GST_BASE_AUDIO_SINK (bsink);
@ -540,9 +542,9 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
"time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT, "time %" GST_TIME_FORMAT ", offset %llu, start %" GST_TIME_FORMAT,
GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->segment.start)); GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->segment.start));
/* if not valid timestamp or we don't need to sync, try to play /* if not valid timestamp or we can't clip or sync, try to play
* sample ASAP */ * sample ASAP */
if (!GST_CLOCK_TIME_IS_VALID (time) || !bsink->sync) { if (!GST_CLOCK_TIME_IS_VALID (time)) {
render_offset = gst_base_audio_sink_get_offset (sink); render_offset = gst_base_audio_sink_get_offset (sink);
stop = -1; stop = -1;
GST_DEBUG_OBJECT (sink, GST_DEBUG_OBJECT (sink,
@ -585,6 +587,22 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
stop = cstop; stop = cstop;
} }
/* figure out how to sync */
if ((clock = GST_ELEMENT_CLOCK (bsink)))
sync = bsink->sync;
else
sync = FALSE;
if (!sync) {
/* no sync needed, play sample ASAP */
render_offset = gst_base_audio_sink_get_offset (sink);
stop = -1;
GST_DEBUG_OBJECT (sink,
"no sync needed. Using render_offset=%" G_GUINT64_FORMAT,
render_offset);
goto no_sync;
}
gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal, gst_clock_get_calibration (sink->provided_clock, &cinternal, &cexternal,
&crate_num, &crate_denom); &crate_num, &crate_denom);
@ -603,12 +621,11 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
", render offset %llu, samples %lu", ", render offset %llu, samples %lu",
GST_TIME_ARGS (render_time), render_offset, samples); GST_TIME_ARGS (render_time), render_offset, samples);
/* never try to align samples when we are slaved to another clock, just /* never try to align samples when we are slaved to another clock, just
* trust the rate control algorithm to align the two clocks. We don't take * trust the rate control algorithm to align the two clocks. We don't take
* the LOCK to read the clock because it does not really matter here and the * the LOCK to read the clock because it does not really matter here and the
* clock is not changed while playing normally. */ * clock is not changed while playing normally. */
if (GST_ELEMENT_CLOCK (sink) != sink->provided_clock) { if (clock != sink->provided_clock) {
GST_DEBUG_OBJECT (sink, "no align needed: we are slaved"); GST_DEBUG_OBJECT (sink, "no align needed: we are slaved");
goto no_align; goto no_align;
} }