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Port to latest GstAggregator segment API

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The aggregator segment is now exposed on the src pad

https://bugzilla.gnome.org/show_bug.cgi?id=793944
This commit is contained in:
Mathieu Duponchelle 2018-03-01 00:34:06 +01:00
parent 318eb61e23
commit 22981e8a42
3 changed files with 44 additions and 39 deletions
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3
ext/gl/gstglmixer.c
View file

@ -364,7 +364,8 @@ gst_gl_mixer_class_init (GstGLMixerClass * klass)
gobject_class->get_property = gst_gl_mixer_get_property;
gobject_class->set_property = gst_gl_mixer_set_property;
gst_element_class_add_static_pad_template (element_class, &src_factory);
gst_element_class_add_static_pad_template_with_gtype (element_class,
&src_factory, GST_TYPE_AGGREGATOR_PAD);
gst_element_class_add_static_pad_template_with_gtype (element_class,
&sink_factory, GST_TYPE_GL_MIXER_PAD);

3
ext/gl/gstglstereomix.c
View file

@ -179,7 +179,8 @@ gst_gl_stereo_mix_class_init (GstGLStereoMixClass * klass)
GST_TYPE_GL_STEREO_DOWNMIX_MODE_TYPE, DEFAULT_DOWNMIX,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (element_class, &src_factory);
gst_element_class_add_static_pad_template_with_gtype (element_class,
&src_factory, GST_TYPE_AGGREGATOR_PAD);
gst_element_class_add_static_pad_template_with_gtype (element_class,
&sink_factory, GST_TYPE_GL_STEREO_MIX_PAD);

77
gst-libs/gst/audio/gstaudioaggregator.c
View file

@ -448,18 +448,18 @@ static GstClockTime
gst_audio_aggregator_get_next_time (GstAggregator * agg)
{
GstClockTime next_time;
GstSegment *segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
GST_OBJECT_LOCK (agg);
if (agg->segment.position == -1 || agg->segment.position < agg->segment.start)
next_time = agg->segment.start;
if (segment->position == -1 || segment->position < segment->start)
next_time = segment->start;
else
next_time = agg->segment.position;
next_time = segment->position;
if (agg->segment.stop != -1 && next_time > agg->segment.stop)
next_time = agg->segment.stop;
if (segment->stop != -1 && next_time > segment->stop)
next_time = segment->stop;
next_time =
gst_segment_to_running_time (&agg->segment, GST_FORMAT_TIME, next_time);
next_time = gst_segment_to_running_time (segment, GST_FORMAT_TIME, next_time);
GST_OBJECT_UNLOCK (agg);
return next_time;
@ -929,7 +929,7 @@ gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event)
}
GST_OBJECT_LOCK (agg);
dest_format = agg->segment.format;
dest_format = GST_AGGREGATOR_PAD (agg->srcpad)->segment.format;
GST_OBJECT_UNLOCK (agg);
if (seek_format != dest_format) {
result = FALSE;
@ -980,10 +980,10 @@ gst_audio_aggregator_sink_event (GstAggregator * agg,
}
GST_OBJECT_LOCK (agg);
if (segment->rate != agg->segment.rate) {
if (segment->rate != GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate) {
GST_ERROR_OBJECT (aggpad,
"Got segment event with wrong rate %lf, expected %lf",
segment->rate, agg->segment.rate);
segment->rate, GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate);
res = FALSE;
gst_event_unref (event);
event = NULL;
@ -1166,8 +1166,9 @@ gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query)
switch (format) {
case GST_FORMAT_TIME:
gst_query_set_position (query, format,
gst_segment_to_stream_time (&agg->segment, GST_FORMAT_TIME,
agg->segment.position));
gst_segment_to_stream_time (&GST_AGGREGATOR_PAD (agg->
srcpad)->segment, GST_FORMAT_TIME,
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position));
res = TRUE;
break;
case GST_FORMAT_BYTES:
@ -1227,7 +1228,7 @@ gst_audio_aggregator_reset (GstAudioAggregator * aagg)
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
agg->segment.position = -1;
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
aagg->priv->offset = -1;
gst_audio_info_init (&GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)->info);
gst_caps_replace (&aagg->current_caps, NULL);
@ -1263,7 +1264,7 @@ gst_audio_aggregator_flush (GstAggregator * agg)
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
agg->segment.position = -1;
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
aagg->priv->offset = -1;
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_OBJECT_UNLOCK (aagg);
@ -1410,6 +1411,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
GstClockTime segment_pos;
guint64 start_output_offset = -1;
guint64 end_output_offset = -1;
GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
start_running_time =
gst_segment_to_running_time (&aggpad->segment,
@ -1420,19 +1422,19 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
/* Convert to position in the output segment */
segment_pos =
gst_segment_position_from_running_time (&agg->segment, GST_FORMAT_TIME,
gst_segment_position_from_running_time (agg_segment, GST_FORMAT_TIME,
start_running_time);
if (GST_CLOCK_TIME_IS_VALID (segment_pos))
start_output_offset =
gst_util_uint64_scale (segment_pos - agg->segment.start, rate,
gst_util_uint64_scale (segment_pos - agg_segment->start, rate,
GST_SECOND);
segment_pos =
gst_segment_position_from_running_time (&agg->segment, GST_FORMAT_TIME,
gst_segment_position_from_running_time (agg_segment, GST_FORMAT_TIME,
end_running_time);
if (GST_CLOCK_TIME_IS_VALID (segment_pos))
end_output_offset =
gst_util_uint64_scale (segment_pos - agg->segment.start, rate,
gst_util_uint64_scale (segment_pos - agg_segment->start, rate,
GST_SECOND);
if (start_output_offset == -1 && end_output_offset == -1) {
@ -1668,6 +1670,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
gboolean is_done = TRUE;
guint blocksize;
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
@ -1679,11 +1682,11 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_OBJECT_LOCK (agg);
/* Update position from the segment start/stop if needed */
if (agg->segment.position == -1) {
if (agg->segment.rate > 0.0)
agg->segment.position = agg->segment.start;
if (agg_segment->position == -1) {
if (agg_segment->rate > 0.0)
agg_segment->position = agg_segment->start;
else
agg->segment.position = agg->segment.stop;
agg_segment->position = agg_segment->stop;
}
if (G_UNLIKELY (srcpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
@ -1692,12 +1695,12 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
"Got timeout before receiving any caps, don't output anything");
/* Advance position */
if (agg->segment.rate > 0.0)
agg->segment.position += aagg->priv->output_buffer_duration;
else if (agg->segment.position > aagg->priv->output_buffer_duration)
agg->segment.position -= aagg->priv->output_buffer_duration;
if (agg_segment->rate > 0.0)
agg_segment->position += aagg->priv->output_buffer_duration;
else if (agg_segment->position > aagg->priv->output_buffer_duration)
agg_segment->position -= aagg->priv->output_buffer_duration;
else
agg->segment.position = 0;
agg_segment->position = 0;
GST_OBJECT_UNLOCK (agg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
@ -1713,7 +1716,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
if (aagg->priv->offset == -1) {
aagg->priv->offset =
gst_util_uint64_scale (agg->segment.position - agg->segment.start, rate,
gst_util_uint64_scale (agg_segment->position - agg_segment->start, rate,
GST_SECOND);
GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT,
aagg->priv->offset);
@ -1724,7 +1727,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
blocksize = MAX (1, blocksize);
/* FIXME: Reverse mixing does not work at all yet */
if (agg->segment.rate > 0.0) {
if (agg_segment->rate > 0.0) {
next_offset = aagg->priv->offset + blocksize;
} else {
next_offset = aagg->priv->offset - blocksize;
@ -1732,7 +1735,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
/* Use the sample counter, which will never accumulate rounding errors */
next_timestamp =
agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
agg_segment->start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
if (aagg->priv->current_buffer == NULL) {
@ -1749,7 +1752,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_LOG_OBJECT (agg,
"Starting to mix %u samples for offset %" G_GINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, blocksize,
aagg->priv->offset, GST_TIME_ARGS (agg->segment.position));
aagg->priv->offset, GST_TIME_ARGS (agg_segment->position));
for (iter = element->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
@ -1904,7 +1907,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
G_GINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
next_timestamp =
agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
agg_segment->start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
if (next_offset > aagg->priv->offset)
@ -1914,16 +1917,16 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
/* set timestamps on the output buffer */
GST_OBJECT_LOCK (agg);
if (agg->segment.rate > 0.0) {
GST_BUFFER_PTS (outbuf) = agg->segment.position;
if (agg_segment->rate > 0.0) {
GST_BUFFER_PTS (outbuf) = agg_segment->position;
GST_BUFFER_OFFSET (outbuf) = aagg->priv->offset;
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
GST_BUFFER_DURATION (outbuf) = next_timestamp - agg->segment.position;
GST_BUFFER_DURATION (outbuf) = next_timestamp - agg_segment->position;
} else {
GST_BUFFER_PTS (outbuf) = next_timestamp;
GST_BUFFER_OFFSET (outbuf) = next_offset;
GST_BUFFER_OFFSET_END (outbuf) = aagg->priv->offset;
GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp;
GST_BUFFER_DURATION (outbuf) = agg_segment->position - next_timestamp;
}
GST_OBJECT_UNLOCK (agg);
@ -1944,7 +1947,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);
aagg->priv->offset = next_offset;
agg->segment.position = next_timestamp;
agg_segment->position = next_timestamp;
/* If there was a timeout and there was a gap in data in out of the streams,
* then it's a very good time to for a resync with the timestamps.