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audioaggregator: Fix mixup of running times and segment positions

Browse source 
We have to queue buffers based on their running time, not based on
the segment position.

Also return running time from GstAggregator::get_next_time() instead of
a segment position, as required by the API.

Also only update the segment position after we pushed a buffer, otherwise
we're going to push down a segment event with the next position already.

https://bugzilla.gnome.org/show_bug.cgi?id=753196
This commit is contained in:
Sebastian Dröge 2015-09-11 21:37:08 +02:00
parent 97fe89f351
commit 637106e287
1 changed files with 100 additions and 38 deletions
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138
gst/audiomixer/gstaudioaggregator.c
View file

@ -53,11 +53,12 @@ struct _GstAudioAggregatorPadPrivate
cached values. */
guint position, size;
guint64 output_offset; /* Offset in output segment that
collect.pos refers to in the
guint64 output_offset; /* Sample offset in output segment relative to
segment.start that collect.pos refers to in the
current buffer. */
guint64 next_offset; /* Next expected offset in the input segment */
guint64 next_offset; /* Next expected sample offset in the input segment
relative to segment.start */
/* Last time we noticed a discont */
GstClockTime discont_time;
@ -145,7 +146,8 @@ struct _GstAudioAggregatorPrivate
/* counters to keep track of timestamps */
/* Readable with object lock, writable with both aag lock and object lock */
gint64 offset;
gint64 offset; /* Sample offset starting from 0 at segment.start */
};
#define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex);
@ -195,10 +197,16 @@ gst_audio_aggregator_get_next_time (GstAggregator * agg)
GstClockTime next_time;
GST_OBJECT_LOCK (agg);
if (agg->segment.position == -1)
if (agg->segment.position == -1 || agg->segment.position < agg->segment.start)
next_time = agg->segment.start;
else
next_time = agg->segment.position;
if (agg->segment.stop != -1 && next_time > agg->segment.stop)
next_time = agg->segment.stop;
next_time =
gst_segment_to_running_time (&agg->segment, GST_FORMAT_TIME, next_time);
GST_OBJECT_UNLOCK (agg);
return next_time;
@ -742,6 +750,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
guint64 start_offset, end_offset;
gint rate, bpf;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
g_assert (pad->priv->buffer == NULL);
@ -767,7 +776,12 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND,
rate);
start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND);
/* Clipping should've ensured this */
g_assert (start_time >= aggpad->segment.start);
start_offset =
gst_util_uint64_scale (start_time - aggpad->segment.start, rate,
GST_SECOND);
end_offset = start_offset + pad->priv->size;
if (GST_BUFFER_IS_DISCONT (inbuf)
@ -822,8 +836,8 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
if (pad->priv->output_offset == -1) {
GstClockTime start_running_time;
GstClockTime end_running_time;
guint64 start_running_time_offset;
guint64 end_running_time_offset;
guint64 start_output_offset;
guint64 end_output_offset;
start_running_time =
gst_segment_to_running_time (&aggpad->segment,
@ -831,12 +845,40 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
end_running_time =
gst_segment_to_running_time (&aggpad->segment,
GST_FORMAT_TIME, end_time);
start_running_time_offset =
gst_util_uint64_scale (start_running_time, rate, GST_SECOND);
end_running_time_offset =
gst_util_uint64_scale (end_running_time, rate, GST_SECOND);
if (end_running_time_offset < aagg->priv->offset) {
/* Convert to position in the output segment */
start_output_offset =
gst_segment_to_position (&agg->segment, GST_FORMAT_TIME,
start_running_time);
if (start_output_offset != -1)
start_output_offset =
gst_util_uint64_scale (start_output_offset - agg->segment.start, rate,
GST_SECOND);
end_output_offset =
gst_segment_to_position (&agg->segment, GST_FORMAT_TIME,
end_running_time);
if (end_output_offset != -1)
end_output_offset =
gst_util_uint64_scale (end_output_offset - agg->segment.start, rate,
GST_SECOND);
if (start_output_offset == -1 && end_output_offset == -1) {
/* Outside output segment, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
pad->priv->position = 0;
pad->priv->size = 0;
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad, "Buffer outside output segment");
return FALSE;
}
/* Calculate end_output_offset if it was outside the output segment */
if (end_output_offset == -1)
end_output_offset = start_output_offset + pad->priv->size;
if (end_output_offset < aagg->priv->offset) {
/* Before output segment, drop */
gst_buffer_unref (inbuf);
pad->priv->buffer = NULL;
@ -845,12 +887,25 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GUINT64_FORMAT, end_running_time_offset, aagg->priv->offset);
G_GUINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
if (start_running_time_offset < aagg->priv->offset) {
guint diff = aagg->priv->offset - start_running_time_offset;
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) {
guint diff;
if (start_output_offset == -1 && end_output_offset < pad->priv->size) {
diff = pad->priv->size - end_output_offset + aagg->priv->offset;
} else if (start_output_offset == -1) {
start_output_offset = end_output_offset - pad->priv->size;
if (start_output_offset < aagg->priv->offset)
diff = aagg->priv->offset - start_output_offset;
else
diff = 0;
} else {
diff = aagg->priv->offset - start_output_offset;
}
pad->priv->position += diff;
if (pad->priv->position >= pad->priv->size) {
@ -862,14 +917,16 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg,
pad->priv->output_offset = -1;
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT, end_running_time_offset,
aagg->priv->offset);
" < %" G_GUINT64_FORMAT, end_output_offset, aagg->priv->offset);
return FALSE;
}
}
pad->priv->output_offset =
MAX (start_running_time_offset, aagg->priv->offset);
if (start_output_offset == -1 || start_output_offset < aagg->priv->offset)
pad->priv->output_offset = aagg->priv->offset;
else
pad->priv->output_offset = start_output_offset;
GST_DEBUG_OBJECT (pad,
"Buffer resynced: Pad offset %" G_GUINT64_FORMAT
", current audio aggregator offset %" G_GUINT64_FORMAT,
@ -1066,13 +1123,10 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_OBJECT_UNLOCK (agg);
gst_aggregator_set_src_caps (agg, aagg->current_caps);
GST_OBJECT_LOCK (agg);
aagg->priv->offset = gst_util_uint64_scale (agg->segment.position,
GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
aagg->priv->send_caps = FALSE;
}
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
@ -1090,7 +1144,9 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
next_offset = aagg->priv->offset - blocksize;
}
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
next_timestamp =
agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
if (aagg->priv->current_buffer == NULL) {
GST_OBJECT_UNLOCK (agg);
@ -1248,7 +1304,9 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
G_GUINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
next_timestamp =
agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
if (next_offset > aagg->priv->offset)
gst_buffer_resize (outbuf, 0, (next_offset - aagg->priv->offset) * bpf);
@ -1269,6 +1327,23 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp;
}
GST_OBJECT_UNLOCK (agg);
/* send it out */
GST_LOG_OBJECT (aagg,
"pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
GST_BUFFER_OFFSET (outbuf));
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
ret = gst_aggregator_finish_buffer (agg, aagg->priv->current_buffer);
aagg->priv->current_buffer = NULL;
GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret));
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (agg);
aagg->priv->offset = next_offset;
agg->segment.position = next_timestamp;
@ -1285,22 +1360,9 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
GST_OBJECT_UNLOCK (pad);
}
}
GST_OBJECT_UNLOCK (agg);
/* send it out */
GST_LOG_OBJECT (aagg,
"pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
GST_BUFFER_OFFSET (outbuf));
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
ret = gst_aggregator_finish_buffer (agg, aagg->priv->current_buffer);
aagg->priv->current_buffer = NULL;
GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret));
return ret;
/* ERRORS */
not_negotiated: