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audiobasesink: drop GAP buffers, or all buffers in trickmode no-audio mode

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Make the base audio sink throw away buffers marked GAP, or all
incoming buffers when performing a trick play with
GST_SEGMENT_TRICKMODE_NO_AUDIO flag set, and make sure to start
the ringbuffer when that happens so the clock starts running.

Preserve the timing calculations when rendering, so state is all
updated the same, but just don't render samples.

https://bugzilla.gnome.org/show_bug.cgi?id=735666
This commit is contained in:
Jan Schmidt 2015-01-27 02:04:22 +11:00
parent 9491157f1a
commit 8557eead82
1 changed files with 84 additions and 47 deletions
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131
gst-libs/gst/audio/gstaudiobasesink.c
View file

@ -182,6 +182,9 @@ static GstCaps *gst_audio_base_sink_fixate (GstBaseSink * bsink,
static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
GstQuery * query);
static GstFlowReturn
gst_audio_base_sink_render_samples (GstBaseSink * bsink,
GstBuffer * buf, GstClockTime time, GstClockTime duration);
/* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
@ -1062,7 +1065,8 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
buffer = gst_buffer_new_and_alloc (n_samples * spec->info.bpf);
if (n_samples != 0) {
gst_buffer_map (buffer, &minfo, GST_MAP_WRITE);
if (!gst_buffer_map (buffer, &minfo, GST_MAP_WRITE))
return FALSE;
gst_audio_format_fill_silence (spec->info.finfo, minfo.data,
minfo.size);
gst_buffer_unmap (buffer, &minfo);
@ -1617,11 +1621,12 @@ gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
return align;
}
/* buf may be NULL, to render silence/skip */
static GstFlowReturn
gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
gst_audio_base_sink_render_samples (GstBaseSink * bsink,
GstBuffer * buf, GstClockTime time, GstClockTime duration)
{
guint64 in_offset;
GstClockTime time, stop, render_start, render_stop, sample_offset;
GstClockTime stop, render_start, render_stop, sample_offset;
GstClockTimeDiff sync_offset, ts_offset;
GstAudioBaseSinkClass *bclass;
GstAudioBaseSink *sink;
@ -1630,8 +1635,7 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
guint64 ctime, cstop;
gsize offset;
GstMapInfo info;
gsize size;
guint samples, written;
guint samples = 0, written;
gint bpf, rate;
gint accum;
gint out_samples;
@ -1670,7 +1674,7 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
/* Before we go on, let's see if we need to payload the data. If yes, we also
* need to unref the output buffer before leaving. */
if (bclass->payload) {
if (buf && bclass->payload) {
out = bclass->payload (sink, buf);
if (!out)
@ -1682,19 +1686,31 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
size = gst_buffer_get_size (buf);
if (G_UNLIKELY (size % bpf) != 0)
goto wrong_size;
if (buf) {
gsize size = gst_buffer_get_size (buf);
if (G_UNLIKELY (size % bpf) != 0)
goto wrong_size;
samples = size / bpf;
samples = size / bpf;
} else if (duration != GST_CLOCK_TIME_NONE) {
/* Convert the passed duration to a number of samples */
samples = gst_util_uint64_scale_int (duration, rate, GST_SECOND);
}
out_samples = samples;
in_offset = GST_BUFFER_OFFSET (buf);
time = GST_BUFFER_TIMESTAMP (buf);
/* Last ditch attempt to ensure that we only place silence if
* we are in trickmode no-audio mode by dropping the buffer contents.
* We then continue calculations based on the passed times/duration */
if (G_UNLIKELY ((bsink->segment.flags & GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO)
|| (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))) {
GST_DEBUG_OBJECT (bsink,
"Received GAP or ignoring audio for trickplay. Dropping contents");
buf = NULL;
}
GST_DEBUG_OBJECT (sink,
"time %" GST_TIME_FORMAT ", offset %" G_GUINT64_FORMAT ", start %"
GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time), in_offset,
"time %" GST_TIME_FORMAT ", start %"
GST_TIME_FORMAT ", samples %u", GST_TIME_ARGS (time),
GST_TIME_ARGS (bsink->segment.start), samples);
offset = 0;
@ -1704,8 +1720,8 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
if (!GST_CLOCK_TIME_IS_VALID (time)) {
render_start = gst_audio_base_sink_get_offset (sink);
render_stop = render_start + samples;
GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
" Using render_start=%" G_GUINT64_FORMAT, size, render_start);
GST_DEBUG_OBJECT (sink, "Samples have no timestamp."
" Using render_start=%" G_GUINT64_FORMAT, render_start);
/* we don't have a start so we don't know stop either */
stop = -1;
goto no_align;
@ -1896,10 +1912,12 @@ gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
}
/* always resync after a discont */
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT) ||
GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_RESYNC))) {
GST_DEBUG_OBJECT (sink, "resync after discont/resync");
goto no_align;
if (G_LIKELY (buf)) {
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT) ||
GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_RESYNC))) {
GST_DEBUG_OBJECT (sink, "resync after discont/resync");
goto no_align;
}
}
/* resync when we don't know what to align the sample with */
@ -1939,37 +1957,48 @@ no_align:
/* we need to accumulate over different runs for when we get interrupted */
accum = 0;
align_next = TRUE;
gst_buffer_map (buf, &info, GST_MAP_READ);
do {
written =
gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
info.data + offset, samples, out_samples, &accum);
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &info, GST_MAP_READ);
do {
written =
gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
info.data + offset, samples, out_samples, &accum);
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (G_LIKELY (written == samples))
break;
GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
/* if we wrote all, we're done */
if (G_LIKELY (written == samples))
break;
/* else something interrupted us and we wait for preroll. */
if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
goto stopping;
/* else something interrupted us and we wait for preroll. */
if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
goto stopping;
/* if we got interrupted, we cannot assume that the next sample should
* be aligned to this one */
align_next = FALSE;
/* if we got interrupted, we cannot assume that the next sample should
* be aligned to this one */
align_next = FALSE;
/* update the output samples. FIXME, this will just skip them when pausing
* during trick mode */
if (out_samples > written) {
out_samples -= written;
accum = 0;
} else
break;
/* update the output samples. FIXME, this will just skip them when pausing
* during trick mode */
if (out_samples > written) {
out_samples -= written;
accum = 0;
} else
break;
samples -= written;
offset += written * bpf;
} while (TRUE);
gst_buffer_unmap (buf, &info);
samples -= written;
offset += written * bpf;
} while (TRUE);
gst_buffer_unmap (buf, &info);
} else {
GstClockTime buffer_time = ringbuf->spec.buffer_time * GST_USECOND;
GST_DEBUG_OBJECT (sink, "Discarded audio content. Starting ringbuffer");
gst_audio_base_sink_force_start (sink);
/* Wait until the time when we would have written out this timestamp */
if (sink->priv->eos_time != -1 && sink->priv->eos_time > buffer_time)
gst_base_sink_wait (GST_BASE_SINK (sink),
sink->priv->eos_time - buffer_time, NULL);
}
if (G_LIKELY (align_next))
sink->next_sample = sample_offset;
@ -2046,6 +2075,14 @@ sync_latency_failed:
}
}
static GstFlowReturn
gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
{
GstClockTime ts = GST_BUFFER_TIMESTAMP (buf);
GstClockTime dur = GST_BUFFER_DURATION (buf);
return gst_audio_base_sink_render_samples (bsink, buf, ts, dur);
}
/**
* gst_audio_base_sink_create_ringbuffer:
* @sink: a #GstAudioBaseSink.