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audiobasesink: add some G_LIKELY

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This commit is contained in:
Wim Taymans 2012-03-09 17:15:38 +01:00
parent 94869bff38
commit 7296ef7c63
1 changed files with 48 additions and 42 deletions
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90
gst-libs/gst/audio/gstaudiobasesink.c
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@ -1670,7 +1670,7 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
/* compensate for ts-offset and device-delay when negative we need to
* clip. */
if (sync_offset < 0) {
if (G_UNLIKELY (sync_offset < 0)) {
clip_seg.start += -sync_offset;
if (clip_seg.stop != -1)
clip_seg.stop += -sync_offset;
@ -1680,13 +1680,13 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
* arriving before the segment.start or after segment.stop are to be
* thrown away. All samples should also be clipped to the segment
* boundaries */
if (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop, &ctime,
&cstop))
if (G_UNLIKELY (!gst_segment_clip (&clip_seg, GST_FORMAT_TIME, time, stop,
&ctime, &cstop)))
goto out_of_segment;
/* see if some clipping happened */
diff = ctime - time;
if (diff > 0) {
if (G_UNLIKELY (diff > 0)) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping start to %" GST_TIME_FORMAT " %"
@ -1696,7 +1696,7 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
time = ctime;
}
diff = stop - cstop;
if (diff > 0) {
if (G_UNLIKELY (diff > 0)) {
/* bring clipped time to samples */
diff = gst_util_uint64_scale_int (diff, rate, GST_SECOND);
GST_DEBUG_OBJECT (sink, "clipping stop to %" GST_TIME_FORMAT " %"
@ -1706,12 +1706,12 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
}
/* figure out how to sync */
if ((clock = GST_ELEMENT_CLOCK (bsink)))
if (G_LIKELY ((clock = GST_ELEMENT_CLOCK (bsink))))
sync = bsink->sync;
else
sync = FALSE;
if (!sync) {
if (G_UNLIKELY (!sync)) {
/* no sync needed, play sample ASAP */
render_start = gst_audio_base_sink_get_offset (sink);
render_stop = render_start + samples;
@ -1732,32 +1732,32 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
/* store the time of the last sample, we'll use this to perform sync on the
* last sample when draining the buffer */
if (bsink->segment.rate >= 0.0) {
if (G_LIKELY (bsink->segment.rate >= 0.0)) {
sink->priv->eos_time = render_stop;
} else {
sink->priv->eos_time = render_start;
}
/* compensate for ts-offset and delay we know this will not underflow because we
* clipped above. */
GST_DEBUG_OBJECT (sink,
"compensating for sync-offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (sync_offset));
render_start += sync_offset;
render_stop += sync_offset;
if (G_UNLIKELY (sync_offset != 0)) {
/* compensate for ts-offset and delay we know this will not underflow because we
* clipped above. */
GST_DEBUG_OBJECT (sink,
"compensating for sync-offset %" GST_TIME_FORMAT,
GST_TIME_ARGS (sync_offset));
render_start += sync_offset;
render_stop += sync_offset;
}
GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time));
if (base_time != 0) {
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;
/* add base time to sync against the clock */
render_start += base_time;
render_stop += base_time;
}
GST_DEBUG_OBJECT (sink,
"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)) {
if (G_UNLIKELY ((slaved = (clock != sink->provided_clock)))) {
/* handle clock slaving */
gst_audio_base_sink_handle_slaving (sink, render_start, render_stop,
&render_start, &render_stop);
@ -1774,16 +1774,20 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
/* bring to position in the ringbuffer */
time_offset = GST_AUDIO_CLOCK_CAST (sink->provided_clock)->time_offset;
GST_DEBUG_OBJECT (sink,
"time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
if (render_start > time_offset)
render_start -= time_offset;
else
render_start = 0;
if (render_stop > time_offset)
render_stop -= time_offset;
else
render_stop = 0;
if (G_UNLIKELY (time_offset != 0)) {
GST_DEBUG_OBJECT (sink,
"apply time offset %" GST_TIME_FORMAT, GST_TIME_ARGS (time_offset));
if (render_start > time_offset)
render_start -= time_offset;
else
render_start = 0;
if (render_stop > time_offset)
render_stop -= time_offset;
else
render_stop = 0;
}
/* in some clock slaving cases, all late samples end up at 0 first,
* and subsequent ones align with that until threshold exceeded,
@ -1798,9 +1802,9 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
/* positive playback rate, first sample is render_start, negative rate, first
* sample is render_stop. When no rate conversion is active, render exactly
* the amount of input samples to avoid aligning to rounding errors. */
if (bsink->segment.rate >= 0.0) {
if (G_LIKELY (bsink->segment.rate >= 0.0)) {
sample_offset = render_start;
if (bsink->segment.rate == 1.0)
if (G_LIKELY (bsink->segment.rate == 1.0))
render_stop = sample_offset + samples;
} else {
sample_offset = render_stop;
@ -1828,7 +1832,8 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
render_start += align;
/* only align stop if we are not slaved to resample */
if (slaved && sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE) {
if (G_UNLIKELY (slaved
&& sink->priv->slave_method == GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE)) {
GST_DEBUG_OBJECT (sink, "no stop time align needed: we are slaved");
goto no_align;
}
@ -1839,7 +1844,7 @@ no_align:
out_samples = render_stop - render_start;
/* we render the first or last sample first, depending on the rate */
if (bsink->segment.rate >= 0.0)
if (G_LIKELY (bsink->segment.rate >= 0.0))
sample_offset = render_start;
else
sample_offset = render_stop;
@ -1858,7 +1863,7 @@ no_align:
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (written == samples)
if (G_LIKELY (written == samples))
break;
/* else something interrupted us and we wait for preroll. */
@ -1882,7 +1887,7 @@ no_align:
} while (TRUE);
gst_buffer_unmap (buf, &info);
if (align_next)
if (G_LIKELY (align_next))
sink->next_sample = sample_offset;
else
sink->next_sample = -1;
@ -1890,7 +1895,8 @@ no_align:
GST_DEBUG_OBJECT (sink, "next sample expected at %" G_GUINT64_FORMAT,
sink->next_sample);
if (GST_CLOCK_TIME_IS_VALID (stop) && stop >= bsink->segment.stop) {
if (G_UNLIKELY (GST_CLOCK_TIME_IS_VALID (stop)
&& stop >= bsink->segment.stop)) {
GST_DEBUG_OBJECT (sink,
"start playback because we are at the end of segment");
gst_audio_ring_buffer_start (ringbuf);