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soundtouch: Don't assume output buffer timestamps

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There's no guarantee whatsoever that the first buffer to output will
start at the segment.start.

Instead, wait for the first buffer after a segment, and use that timestamp
This commit is contained in:
Edward Hervey 2012-08-08 15:06:09 +02:00
parent c6b16aed08
commit c4ccca8795
1 changed files with 16 additions and 27 deletions
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43
ext/soundtouch/gstpitch.cc
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@ -180,7 +180,7 @@ gst_pitch_init (GstPitch * pitch)
pitch->out_seg_rate = 1.0;
pitch->seg_arate = 1.0;
pitch->pitch = 1.0;
pitch->next_buffer_time = 0;
pitch->next_buffer_time = GST_CLOCK_TIME_NONE;
pitch->next_buffer_offset = 0;
pitch->priv->st->setRate (pitch->rate);
@ -665,8 +665,6 @@ gst_pitch_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
static gboolean
gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
{
GstFormat conv_format;
gint64 next_offset = 0, next_time = 0;
gdouble out_seg_rate, our_arate;
gfloat stream_time_ratio;
GstSegment seg;
@ -715,28 +713,10 @@ gst_pitch_process_segment (GstPitch * pitch, GstEvent ** event)
GST_OBJECT_UNLOCK (pitch);
seg.start = (gint64) (seg.start / stream_time_ratio);
if (seg.stop != (guint64) -1)
if (seg.stop != (guint64) - 1)
seg.stop = (gint64) (seg.stop / stream_time_ratio);
seg.time = (gint64) (seg.time / stream_time_ratio);
conv_format = GST_FORMAT_TIME;
if (!gst_pitch_convert (pitch, seg.format, seg.start, &conv_format, &next_time)) {
GST_LOG_OBJECT (pitch->sinkpad,
"could not convert segment start value to time");
return FALSE;
}
conv_format = GST_FORMAT_DEFAULT;
if (!gst_pitch_convert (pitch, seg.format, seg.start, &conv_format,
&next_offset)) {
GST_LOG_OBJECT (pitch->sinkpad,
"could not convert segment start value to offset");
return FALSE;
}
pitch->next_buffer_time = next_time;
pitch->next_buffer_offset = next_offset;
gst_event_unref (*event);
*event = gst_event_new_segment (&seg);
@ -758,7 +738,7 @@ gst_pitch_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
gst_pitch_flush_buffer (pitch, FALSE);
pitch->priv->st->clear ();
pitch->next_buffer_offset = 0;
pitch->next_buffer_time = 0;
pitch->next_buffer_time = GST_CLOCK_TIME_NONE;
pitch->min_latency = pitch->max_latency = 0;
break;
case GST_EVENT_EOS:
@ -837,13 +817,22 @@ gst_pitch_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
pitch = GST_PITCH (parent);
priv = GST_PITCH_GET_PRIVATE (pitch);
gst_object_sync_values (GST_OBJECT (pitch), pitch->next_buffer_time);
timestamp = GST_BUFFER_TIMESTAMP (buffer);
// Remember the first time and corresponding offset
if (!GST_CLOCK_TIME_IS_VALID (pitch->next_buffer_time)) {
GstFormat out_format = GST_FORMAT_DEFAULT;
pitch->next_buffer_time = timestamp;
gst_pitch_convert (pitch, GST_FORMAT_TIME, timestamp, &out_format,
&pitch->next_buffer_offset);
}
gst_object_sync_values (GST_OBJECT (pitch), pitch->next_buffer_time);
/* push the received samples on the soundtouch buffer */
GST_LOG_OBJECT (pitch, "incoming buffer (%d samples)",
(gint) (gst_buffer_get_size (buffer) / pitch->sample_size));
GST_LOG_OBJECT (pitch, "incoming buffer (%d samples) %" GST_TIME_FORMAT,
(gint) (gst_buffer_get_size (buffer) / pitch->sample_size),
GST_TIME_ARGS (timestamp));
if (GST_PITCH_GET_PRIVATE (pitch)->pending_segment) {
GstEvent *event =