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-rw-r--r--gst-libs/gst/base/gstaggregator.c10
-rw-r--r--gst-libs/gst/base/gstaggregator.h3
-rw-r--r--gst-libs/gst/video/gstvideoaggregator.c4
-rw-r--r--gst/audiomixer/gstaudiomixer.c20
-rw-r--r--gst/compositor/compositor.c3
-rw-r--r--tests/check/libs/aggregator.c2
6 files changed, 26 insertions, 16 deletions
diff --git a/gst-libs/gst/base/gstaggregator.c b/gst-libs/gst/base/gstaggregator.c
index bb80527f3..16690f456 100644
--- a/gst-libs/gst/base/gstaggregator.c
+++ b/gst-libs/gst/base/gstaggregator.c
@@ -486,12 +486,14 @@ gst_aggregator_get_next_time (GstAggregator * self)
/* called with the src STREAM lock */
static gboolean
-_wait_and_check (GstAggregator * self)
+_wait_and_check (GstAggregator * self, gboolean * timeout)
{
GstClockTime latency_max, latency_min;
GstClockTime start;
gboolean live;
+ *timeout = FALSE;
+
gst_aggregator_get_latency (self, &live, &latency_min, &latency_max);
if (gst_aggregator_iterate_sinkpads (self,
@@ -561,6 +563,7 @@ _wait_and_check (GstAggregator * self)
/* we timed out */
if (status == GST_CLOCK_OK || status == GST_CLOCK_EARLY) {
SRC_STREAM_UNLOCK (self);
+ *timeout = TRUE;
return TRUE;
}
}
@@ -575,6 +578,7 @@ aggregate_func (GstAggregator * self)
{
GstAggregatorPrivate *priv = self->priv;
GstAggregatorClass *klass = GST_AGGREGATOR_GET_CLASS (self);
+ gboolean timeout = FALSE;
if (self->priv->running == FALSE) {
GST_DEBUG_OBJECT (self, "Not running anymore");
@@ -583,12 +587,12 @@ aggregate_func (GstAggregator * self)
GST_LOG_OBJECT (self, "Checking aggregate");
while (priv->send_eos && priv->running) {
- if (!_wait_and_check (self))
+ if (!_wait_and_check (self, &timeout))
continue;
GST_TRACE_OBJECT (self, "Actually aggregating!");
- priv->flow_return = klass->aggregate (self);
+ priv->flow_return = klass->aggregate (self, timeout);
if (priv->flow_return == GST_FLOW_EOS) {
_push_eos (self);
diff --git a/gst-libs/gst/base/gstaggregator.h b/gst-libs/gst/base/gstaggregator.h
index 19b04e982..d11545cb5 100644
--- a/gst-libs/gst/base/gstaggregator.h
+++ b/gst-libs/gst/base/gstaggregator.h
@@ -237,7 +237,8 @@ struct _GstAggregatorClass {
GstPadMode mode,
gboolean active);
- GstFlowReturn (*aggregate) (GstAggregator * aggregator);
+ GstFlowReturn (*aggregate) (GstAggregator * aggregator,
+ gboolean timeout);
gboolean (*stop) (GstAggregator * aggregator);
diff --git a/gst-libs/gst/video/gstvideoaggregator.c b/gst-libs/gst/video/gstvideoaggregator.c
index a711beea8..898de598f 100644
--- a/gst-libs/gst/video/gstvideoaggregator.c
+++ b/gst-libs/gst/video/gstvideoaggregator.c
@@ -1230,7 +1230,7 @@ gst_videoaggregator_get_next_time (GstAggregator * agg)
}
static GstFlowReturn
-gst_videoaggregator_aggregate (GstAggregator * agg)
+gst_videoaggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
GstFlowReturn ret;
GstVideoAggregator *vagg = GST_VIDEO_AGGREGATOR (agg);
@@ -1275,7 +1275,7 @@ gst_videoaggregator_aggregate (GstAggregator * agg)
output_end_time);
}
- if (res == GST_FLOW_NEEDS_DATA) {
+ if (res == GST_FLOW_NEEDS_DATA && !timeout) {
GST_DEBUG_OBJECT (vagg, "Need more data for decisions");
ret = GST_FLOW_OK;
goto done;
diff --git a/gst/audiomixer/gstaudiomixer.c b/gst/audiomixer/gstaudiomixer.c
index 9cac2d7b9..5929b2dcc 100644
--- a/gst/audiomixer/gstaudiomixer.c
+++ b/gst/audiomixer/gstaudiomixer.c
@@ -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);
diff --git a/gst/compositor/compositor.c b/gst/compositor/compositor.c
index 04d01a14c..c5d09011c 100644
--- a/gst/compositor/compositor.c
+++ b/gst/compositor/compositor.c
@@ -280,6 +280,9 @@ gst_compositor_pad_prepare_frame (GstVideoAggregatorPad * pad,
static GstAllocationParams params = { 0, 15, 0, 0, };
gint width, height;
+ if (!pad->buffer)
+ return TRUE;
+
if (!gst_video_frame_map (frame, &pad->buffer_vinfo, pad->buffer,
GST_MAP_READ)) {
GST_WARNING_OBJECT (vagg, "Could not map input buffer");
diff --git a/tests/check/libs/aggregator.c b/tests/check/libs/aggregator.c
index 46a657fa1..5e0c388d2 100644
--- a/tests/check/libs/aggregator.c
+++ b/tests/check/libs/aggregator.c
@@ -63,7 +63,7 @@ struct _GstTestAggregatorClass
};
static GstFlowReturn
-gst_test_aggregator_aggregate (GstAggregator * aggregator)
+gst_test_aggregator_aggregate (GstAggregator * aggregator, gboolean timeout)
{
GstIterator *iter;
gboolean all_eos = TRUE;
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