mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-03-28 20:05:38 +00:00
Code Issues 1 Projects Releases Wiki Activity

aggregator: Add a timeout parameter to ::aggregate()

Browse source 
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
This commit is contained in:
Sebastian Dröge 2014-12-17 17:54:09 +01:00
parent 67ef96c82d
commit d508b39952
1 changed files with 11 additions and 9 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

20
gst/audiomixer/gstaudiomixer.c
View file

@ -233,7 +233,8 @@ static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad);
static GstFlowReturn
gst_audiomixer_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf);
static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg);
static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg,
gboolean timeout);
static GstClockTime
gst_audiomixer_get_next_time (GstAggregator * agg)
@ -1327,7 +1328,7 @@ gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
}
static GstFlowReturn
gst_audiomixer_aggregate (GstAggregator * agg)
gst_audiomixer_aggregate (GstAggregator * agg, gboolean timeout)
{
/* Get all pads that have data for us and store them in a
* new list.
@ -1401,7 +1402,6 @@ gst_audiomixer_aggregate (GstAggregator * agg)
} else {
next_offset = audiomixer->offset - audiomixer->blocksize;
}
next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
if (audiomixer->current_buffer) {
@ -1428,13 +1428,14 @@ gst_audiomixer_aggregate (GstAggregator * agg)
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (iter->data);
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (iter->data);
inbuf = gst_aggregator_pad_get_buffer (aggpad);
if (!inbuf)
continue;
g_assert (!pad->buffer || pad->buffer == inbuf);
/* New buffer? */
if (!pad->buffer || pad->buffer != inbuf) {
if (!pad->buffer) {
/* Takes ownership of buffer */
if (!gst_audio_mixer_fill_buffer (audiomixer, pad, inbuf)) {
dropped = TRUE;
@ -1451,11 +1452,13 @@ gst_audiomixer_aggregate (GstAggregator * agg)
}
/* At this point adata->output_offset >= audiomixer->offset or we have no buffer anymore */
g_assert (!pad->buffer || pad->output_offset >= audiomixer->offset);
if (pad->output_offset >= audiomixer->offset
&& pad->output_offset <
audiomixer->offset + audiomixer->blocksize && pad->buffer) {
GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
gst_audio_mixer_mix_buffer (audiomixer, pad, &outmap);
if (pad->output_offset >= next_offset) {
GST_DEBUG_OBJECT (pad,
"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
@ -1469,17 +1472,17 @@ gst_audiomixer_aggregate (GstAggregator * agg)
gst_buffer_unmap (outbuf, &outmap);
if (dropped) {
if (dropped && !timeout) {
/* We dropped a buffer, retry */
GST_INFO_OBJECT (audiomixer,
"A pad dropped a buffer, wait for the next one");
return GST_FLOW_OK;
}
if (!is_done && !is_eos) {
if (!is_done && !is_eos && !timeout) {
/* Get more buffers */
GST_INFO_OBJECT (audiomixer,
"We're not done yet for the current offset," " waiting for more data");
"We're not done yet for the current offset, waiting for more data");
return GST_FLOW_OK;
}
@ -1489,7 +1492,6 @@ gst_audiomixer_aggregate (GstAggregator * agg)
GST_DEBUG_OBJECT (audiomixer, "We're EOS");
GST_OBJECT_LOCK (agg);
for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (iter->data);