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audiobasesink: Re-work GAP buffer and trick-mode handling
In trickmode no-audio mode, or when receiving a GAP buffer, discard the contents and render as a GAP event instead. Make sure when rendering a gap event that the ring buffer will restart on PAUSED->PLAYING by setting the eos_rendering flag. This mostly reverts commit 8557ee and replaces it. The problem with the previous approach is that it hangs in wait_preroll() on a PLAYING-PAUSED transition because it doesn't commit state properly. https://bugzilla.gnome.org/show_bug.cgi?id=735666
This commit is contained in:
Jan Schmidt
parent
d3f6d2b887
commit
ca231ce321
1 changed files with 69 additions and 82 deletions
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@ -182,9 +182,6 @@ static GstCaps *gst_audio_base_sink_fixate (GstBaseSink * bsink,
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static gboolean gst_audio_base_sink_query_pad (GstBaseSink * bsink,
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GstQuery * query);
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static GstFlowReturn
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gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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GstBuffer * buf, GstClockTime time, GstClockTime duration);
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/* static guint gst_audio_base_sink_signals[LAST_SIGNAL] = { 0 }; */
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@ -1032,6 +1029,7 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
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{
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GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
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GstFlowReturn ret;
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gboolean clear_force_start_flag = FALSE;
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/* For both gap and EOS events, make sure the ringbuffer is running
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* before trying to wait on the event! */
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@ -1046,6 +1044,9 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
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}
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gst_audio_base_sink_force_start (sink);
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/* Make sure the ringbuffer will start again if interrupted during event_wait() */
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g_atomic_int_set (&sink->eos_rendering, 1);
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clear_force_start_flag = TRUE;
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break;
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default:
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break;
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@ -1053,15 +1054,9 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
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ret = GST_BASE_SINK_CLASS (parent_class)->wait_event (bsink, event);
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if (ret != GST_FLOW_OK)
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return ret;
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goto done;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_GAP:{
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GstClockTime ts, dur;
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gst_event_parse_gap (event, &ts, &dur);
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gst_audio_base_sink_render_samples (bsink, NULL, ts, dur);
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break;
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}
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case GST_EVENT_EOS:
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/* now wait till we played everything */
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gst_audio_base_sink_drain (sink);
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@ -1069,6 +1064,10 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
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default:
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break;
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}
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done:
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if (clear_force_start_flag)
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g_atomic_int_set (&sink->eos_rendering, 0);
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return ret;
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}
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@ -1601,12 +1600,10 @@ gst_audio_base_sink_get_alignment (GstAudioBaseSink * sink,
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return align;
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}
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/* buf may be NULL, to render silence/skip */
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static GstFlowReturn
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gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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GstBuffer * buf, GstClockTime time, GstClockTime duration)
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gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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{
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GstClockTime stop, render_start, render_stop, sample_offset;
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GstClockTime time, stop, render_start, render_stop, sample_offset;
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GstClockTimeDiff sync_offset, ts_offset;
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GstAudioBaseSinkClass *bclass;
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GstAudioBaseSink *sink;
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@ -1615,7 +1612,8 @@ gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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guint64 ctime, cstop;
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gsize offset;
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GstMapInfo info;
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guint samples = 0, written;
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gsize size;
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guint samples, written;
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gint bpf, rate;
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gint accum;
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gint out_samples;
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@ -1654,7 +1652,7 @@ gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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/* Before we go on, let's see if we need to payload the data. If yes, we also
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* need to unref the output buffer before leaving. */
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if (buf && bclass->payload) {
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if (bclass->payload) {
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out = bclass->payload (sink, buf);
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if (!out)
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@ -1666,26 +1664,36 @@ gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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bpf = GST_AUDIO_INFO_BPF (&ringbuf->spec.info);
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rate = GST_AUDIO_INFO_RATE (&ringbuf->spec.info);
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if (buf) {
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gsize size = gst_buffer_get_size (buf);
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if (G_UNLIKELY (size % bpf) != 0)
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goto wrong_size;
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size = gst_buffer_get_size (buf);
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if (G_UNLIKELY (size % bpf) != 0)
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goto wrong_size;
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samples = size / bpf;
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} else if (duration != GST_CLOCK_TIME_NONE) {
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/* Convert the passed duration to a number of samples */
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samples = gst_util_uint64_scale_int (duration, rate, GST_SECOND);
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}
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samples = size / bpf;
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out_samples = samples;
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/* Last ditch attempt to ensure that we only place silence if
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* we are in trickmode no-audio mode by dropping the buffer contents.
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* We then continue calculations based on the passed times/duration */
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time = GST_BUFFER_TIMESTAMP (buf);
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/* Last ditch attempt to ensure that we only play silence if
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* we are in trickmode no-audio mode (or if a buffer is marked as a GAP)
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* by dropping the buffer contents and rendering as a gap event instead */
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if (G_UNLIKELY ((bsink->segment.flags & GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO)
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|| (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))) {
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GstClockTime duration;
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GstEvent *event;
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GstBaseSinkClass *bclass;
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GST_DEBUG_OBJECT (bsink,
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"Received GAP or ignoring audio for trickplay. Dropping contents");
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buf = NULL;
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duration = gst_util_uint64_scale_int (samples, GST_SECOND, rate);
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event = gst_event_new_gap (time, duration);
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bclass = GST_BASE_SINK_GET_CLASS (bsink);
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ret = bclass->wait_event (bsink, event);
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gst_event_unref (event);
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/* Ensure we'll resync on the next buffer as if discont */
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sink->next_sample = -1;
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goto done;
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}
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GST_DEBUG_OBJECT (sink,
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@ -1700,8 +1708,8 @@ gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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if (!GST_CLOCK_TIME_IS_VALID (time)) {
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render_start = gst_audio_base_sink_get_offset (sink);
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render_stop = render_start + samples;
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GST_DEBUG_OBJECT (sink, "Samples have no timestamp."
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" Using render_start=%" G_GUINT64_FORMAT, render_start);
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GST_DEBUG_OBJECT (sink, "Buffer of size %" G_GSIZE_FORMAT " has no time."
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" Using render_start=%" G_GUINT64_FORMAT, size, render_start);
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/* we don't have a start so we don't know stop either */
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stop = -1;
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goto no_align;
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@ -1892,12 +1900,10 @@ gst_audio_base_sink_render_samples (GstBaseSink * bsink,
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}
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/* always resync after a discont */
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if (G_LIKELY (buf)) {
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if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT) ||
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GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_RESYNC))) {
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GST_DEBUG_OBJECT (sink, "resync after discont/resync");
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goto no_align;
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}
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if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT) ||
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GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_RESYNC))) {
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GST_DEBUG_OBJECT (sink, "resync after discont/resync");
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goto no_align;
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}
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/* resync when we don't know what to align the sample with */
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@ -1937,48 +1943,37 @@ no_align:
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/* we need to accumulate over different runs for when we get interrupted */
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accum = 0;
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align_next = TRUE;
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if (G_LIKELY (buf)) {
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gst_buffer_map (buf, &info, GST_MAP_READ);
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do {
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written =
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gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
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info.data + offset, samples, out_samples, &accum);
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gst_buffer_map (buf, &info, GST_MAP_READ);
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do {
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written =
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gst_audio_ring_buffer_commit (ringbuf, &sample_offset,
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info.data + offset, samples, out_samples, &accum);
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GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
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/* if we wrote all, we're done */
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if (G_LIKELY (written == samples))
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break;
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GST_DEBUG_OBJECT (sink, "wrote %u of %u", written, samples);
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/* if we wrote all, we're done */
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if (G_LIKELY (written == samples))
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break;
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/* else something interrupted us and we wait for preroll. */
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if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
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goto stopping;
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/* else something interrupted us and we wait for preroll. */
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if ((ret = gst_base_sink_wait_preroll (bsink)) != GST_FLOW_OK)
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goto stopping;
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/* if we got interrupted, we cannot assume that the next sample should
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* be aligned to this one */
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align_next = FALSE;
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/* if we got interrupted, we cannot assume that the next sample should
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* be aligned to this one */
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align_next = FALSE;
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/* update the output samples. FIXME, this will just skip them when pausing
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* during trick mode */
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if (out_samples > written) {
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out_samples -= written;
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accum = 0;
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} else
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break;
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/* update the output samples. FIXME, this will just skip them when pausing
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* during trick mode */
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if (out_samples > written) {
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out_samples -= written;
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accum = 0;
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} else
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break;
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samples -= written;
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offset += written * bpf;
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} while (TRUE);
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gst_buffer_unmap (buf, &info);
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} else {
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GstClockTime buffer_time = ringbuf->spec.buffer_time * GST_USECOND;
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GST_DEBUG_OBJECT (sink, "Discarded audio content. Starting ringbuffer");
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gst_audio_base_sink_force_start (sink);
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/* Wait until the time when we would have written out this timestamp */
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if (sink->priv->eos_time != -1 && sink->priv->eos_time > buffer_time)
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gst_base_sink_wait (GST_BASE_SINK (sink),
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sink->priv->eos_time - buffer_time, NULL);
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}
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samples -= written;
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offset += written * bpf;
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} while (TRUE);
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gst_buffer_unmap (buf, &info);
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if (G_LIKELY (align_next))
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sink->next_sample = sample_offset;
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@ -2055,14 +2050,6 @@ sync_latency_failed:
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}
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}
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static GstFlowReturn
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gst_audio_base_sink_render (GstBaseSink * bsink, GstBuffer * buf)
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{
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GstClockTime ts = GST_BUFFER_TIMESTAMP (buf);
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GstClockTime dur = GST_BUFFER_DURATION (buf);
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return gst_audio_base_sink_render_samples (bsink, buf, ts, dur);
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}
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/**
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* gst_audio_base_sink_create_ringbuffer:
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* @sink: a #GstAudioBaseSink.
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