static gboolean
aggregate_func (GstAggregator * self)
{
GstAggregatorPrivate *priv = self->priv;
GstAggregatorClass *klass = GST_AGGREGATOR_GET_CLASS (self);
GST_LOG_OBJECT (self, "Checking aggregate");
while (priv->send_eos && gst_aggregator_iterate_sinkpads (self,
(GstAggregatorPadForeachFunc) _check_all_pads_with_data_or_eos,
NULL) && priv->running) {
GST_TRACE_OBJECT (self, "Actually aggregating!");
priv->flow_return = klass->aggregate (self);
if (priv->flow_return == GST_FLOW_EOS) {
g_main_context_wakeup (self->priv->mcontext);
_remove_all_sources (self);
_push_eos (self);
}
if (priv->flow_return == GST_FLOW_FLUSHING &&
g_atomic_int_get (&priv->flush_seeking))
priv->flow_return = GST_FLOW_OK;
GST_LOG_OBJECT (self, "flow return is %s",
gst_flow_get_name (priv->flow_return));
if (priv->flow_return != GST_FLOW_OK)
break;
}
return G_SOURCE_REMOVE;
}
static gboolean
_stop (GstAggregator * agg)
{
_reset_flow_values (agg);
gst_aggregator_iterate_sinkpads (agg,
(GstAggregatorPadForeachFunc) _flush_pad, NULL);
return TRUE;
}
static gboolean
gst_gl_mixer_stop (GstAggregator * agg)
{
GstGLMixer *mix = GST_GL_MIXER (agg);
GstGLMixerClass *mixer_class = GST_GL_MIXER_GET_CLASS (mix);
if (!GST_AGGREGATOR_CLASS (parent_class)->stop (agg))
return FALSE;
GST_OBJECT_LOCK (agg);
g_ptr_array_free (mix->frames, TRUE);
mix->frames = NULL;
g_ptr_array_free (mix->array_buffers, TRUE);
mix->array_buffers = NULL;
GST_OBJECT_UNLOCK (agg);
if (mixer_class->reset)
mixer_class->reset (mix);
if (mix->fbo) {
gst_gl_context_del_fbo (mix->context, mix->fbo, mix->depthbuffer);
mix->fbo = 0;
mix->depthbuffer = 0;
}
if (mix->download) {
gst_object_unref (mix->download);
mix->download = NULL;
}
gst_aggregator_iterate_sinkpads (GST_AGGREGATOR (mix), _clean_upload, NULL);
if (mix->priv->query) {
gst_query_unref (mix->priv->query);
mix->priv->query = NULL;
}
if (mix->priv->pool) {
gst_object_unref (mix->priv->pool);
mix->priv->pool = NULL;
}
if (mix->display) {
gst_object_unref (mix->display);
mix->display = NULL;
}
if (mix->context) {
gst_object_unref (mix->context);
mix->context = NULL;
}
gst_gl_mixer_reset (mix);
return TRUE;
}
static gboolean
_stop (GstAggregator * agg)
{
_reset_flow_values (agg);
gst_aggregator_iterate_sinkpads (agg,
(GstAggregatorPadForeachFunc) _stop_pad, NULL);
if (agg->priv->tags)
gst_tag_list_unref (agg->priv->tags);
agg->priv->tags = NULL;
return TRUE;
}
static void
_reset_gl (GstGLContext * context, GstGLVideoMixer * video_mixer)
{
const GstGLFuncs *gl = GST_GL_BASE_MIXER (video_mixer)->context->gl_vtable;
if (video_mixer->vao) {
gl->DeleteVertexArrays (1, &video_mixer->vao);
video_mixer->vao = 0;
}
if (video_mixer->vbo_indices) {
gl->DeleteBuffers (1, &video_mixer->vbo_indices);
video_mixer->vbo_indices = 0;
}
gst_aggregator_iterate_sinkpads (GST_AGGREGATOR (video_mixer), _reset_pad_gl,
NULL);
}
static gboolean
gst_aggregator_query_latency (GstAggregator * self, GstQuery * query)
{
LatencyData data;
data.min = 0;
data.max = GST_CLOCK_TIME_NONE;
data.live = FALSE;
/* query upstream's latency */
gst_aggregator_iterate_sinkpads (self,
(GstAggregatorPadForeachFunc) _latency_query, &data);
if (data.live && GST_CLOCK_TIME_IS_VALID (self->timeout) &&
self->timeout > data.max) {
GST_ELEMENT_WARNING (self, CORE, NEGOTIATION,
("%s", "Timeout too big"),
("The requested timeout value is too big for the latency in the "
"current pipeline. Limiting to %" G_GINT64_FORMAT, data.max));
self->timeout = data.max;
}
self->priv->latency_live = data.live;
self->priv->latency_min = data.min;
self->priv->latency_max = data.max;
/* add our own */
if (GST_CLOCK_TIME_IS_VALID (self->timeout)) {
if (GST_CLOCK_TIME_IS_VALID (data.min))
data.min += self->timeout;
if (GST_CLOCK_TIME_IS_VALID (data.max))
data.max += self->timeout;
}
GST_DEBUG_OBJECT (self, "configured latency live:%s min:%" G_GINT64_FORMAT
" max:%" G_GINT64_FORMAT, data.live ? "true" : "false", data.min,
data.max);
gst_query_set_latency (query, data.live, data.min, data.max);
return TRUE;
}
static void
aggregate_func (GstAggregator * self)
{
GstAggregatorPrivate *priv = self->priv;
GstAggregatorClass *klass = GST_AGGREGATOR_GET_CLASS (self);
if (self->priv->running == FALSE) {
GST_DEBUG_OBJECT (self, "Not running anymore");
return;
}
QUEUE_POP (self);
GST_LOG_OBJECT (self, "Checking aggregate");
while (priv->send_eos && gst_aggregator_iterate_sinkpads (self,
(GstAggregatorPadForeachFunc)
_check_all_pads_with_data_or_eos_or_timeout, NULL) && priv->running) {
GST_TRACE_OBJECT (self, "Actually aggregating!");
priv->flow_return = klass->aggregate (self);
if (priv->flow_return == GST_FLOW_EOS) {
QUEUE_FLUSH (self);
_push_eos (self);
}
if (priv->flow_return == GST_FLOW_FLUSHING &&
g_atomic_int_get (&priv->flush_seeking))
priv->flow_return = GST_FLOW_OK;
GST_LOG_OBJECT (self, "flow return is %s",
gst_flow_get_name (priv->flow_return));
if (priv->flow_return != GST_FLOW_OK)
break;
}
}
static gboolean
_forward_event_to_all_sinkpads (GstAggregator * self, GstEvent * event,
gboolean flush)
{
EventData evdata;
evdata.event = event;
evdata.result = TRUE;
evdata.flush = flush;
/* We first need to set all pads as flushing in a first pass
* as flush_start flush_stop is sometimes sent synchronously
* while we send the seek event */
if (flush)
gst_aggregator_iterate_sinkpads (self,
(GstAggregatorPadForeachFunc) _set_flush_pending, NULL);
gst_pad_forward (self->srcpad, (GstPadForwardFunction) event_forward_func,
&evdata);
gst_event_unref (event);
return evdata.result;
}
static GstFlowReturn
gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout)
{
/* Get all pads that have data for us and store them in a
* new list.
*
* Calculate the current output offset/timestamp and
* offset_end/timestamp_end. Allocate a silence buffer
* for this and store it.
*
* For all pads:
* 1) Once per input buffer (cached)
* 1) Check discont (flag and timestamp with tolerance)
* 2) If discont or new, resync. That means:
* 1) Drop all start data of the buffer that comes before
* the current position/offset.
* 2) Calculate the offset (output segment!) that the first
* frame of the input buffer corresponds to. Base this on
* the running time.
*
* 2) If the current pad's offset/offset_end overlaps with the output
* offset/offset_end, mix it at the appropiate position in the output
* buffer and advance the pad's position. Remember if this pad needs
* a new buffer to advance behind the output offset_end.
*
* 3) If we had no pad with a buffer, go EOS.
*
* 4) If we had at least one pad that did not advance behind output
* offset_end, let collected be called again for the current
* output offset/offset_end.
*/
GstElement *element;
GstAudioAggregator *aagg;
GList *iter;
GstFlowReturn ret;
GstBuffer *outbuf = NULL;
gint64 next_offset;
gint64 next_timestamp;
gint rate, bpf;
gboolean dropped = FALSE;
gboolean is_eos = TRUE;
gboolean is_done = TRUE;
guint blocksize;
element = GST_ELEMENT (agg);
aagg = GST_AUDIO_AGGREGATOR (agg);
/* Sync pad properties to the stream time */
gst_aggregator_iterate_sinkpads (agg,
(GstAggregatorPadForeachFunc) GST_DEBUG_FUNCPTR (sync_pad_values), NULL);
GST_AUDIO_AGGREGATOR_LOCK (aagg);
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;
else
agg->segment.position = agg->segment.stop;
}
if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
if (timeout) {
GST_DEBUG_OBJECT (aagg,
"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;
else
agg->segment.position = 0;
GST_OBJECT_UNLOCK (agg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
} else {
GST_OBJECT_UNLOCK (agg);
goto not_negotiated;
}
}
if (aagg->priv->send_caps) {
GST_OBJECT_UNLOCK (agg);
gst_aggregator_set_src_caps (agg, aagg->current_caps);
GST_OBJECT_LOCK (agg);
aagg->priv->send_caps = FALSE;
}
rate = GST_AUDIO_INFO_RATE (&aagg->info);
bpf = GST_AUDIO_INFO_BPF (&aagg->info);
if (aagg->priv->offset == -1) {
aagg->priv->offset =
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);
}
blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
rate, GST_SECOND);
blocksize = MAX (1, blocksize);
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */
/* FIXME: Reverse mixing does not work at all yet */
if (agg->segment.rate > 0.0) {
next_offset = aagg->priv->offset + blocksize;
} else {
next_offset = aagg->priv->offset - blocksize;
}
next_timestamp =
agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND,
rate);
if (aagg->priv->current_buffer == NULL) {
GST_OBJECT_UNLOCK (agg);
aagg->priv->current_buffer =
GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg,
blocksize);
/* Be careful, some things could have changed ? */
GST_OBJECT_LOCK (agg);
GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP);
}
outbuf = aagg->priv->current_buffer;
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));
for (iter = element->sinkpads; iter; iter = iter->next) {
GstBuffer *inbuf;
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
gboolean drop_buf = FALSE;
gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad);
if (!pad_eos)
is_eos = FALSE;
inbuf = gst_aggregator_pad_get_buffer (aggpad);
GST_OBJECT_LOCK (pad);
if (!inbuf) {
if (timeout) {
if (pad->priv->output_offset < next_offset) {
gint64 diff = next_offset - pad->priv->output_offset;
GST_LOG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT " frames (%"
GST_TIME_FORMAT ")", diff,
GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND,
GST_AUDIO_INFO_RATE (&aagg->info))));
}
} else if (!pad_eos) {
is_done = FALSE;
}
GST_OBJECT_UNLOCK (pad);
continue;
}
g_assert (!pad->priv->buffer || pad->priv->buffer == inbuf);
/* New buffer? */
if (!pad->priv->buffer) {
/* Takes ownership of buffer */
if (!gst_audio_aggregator_fill_buffer (aagg, pad, inbuf)) {
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
} else {
gst_buffer_unref (inbuf);
}
if (!pad->priv->buffer && !dropped && pad_eos) {
GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
GST_OBJECT_UNLOCK (pad);
continue;
}
g_assert (pad->priv->buffer);
/* This pad is lacking behind, we need to update the offset
* and maybe drop the current buffer */
if (pad->priv->output_offset < aagg->priv->offset) {
gint64 diff = aagg->priv->offset - pad->priv->output_offset;
gint64 odiff = diff;
if (pad->priv->position + diff > pad->priv->size)
diff = pad->priv->size - pad->priv->position;
pad->priv->position += diff;
pad->priv->output_offset += diff;
if (pad->priv->position == pad->priv->size) {
GST_LOG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT
", dropping %" GST_PTR_FORMAT,
GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND,
GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer);
/* Buffer done, drop it */
gst_buffer_replace (&pad->priv->buffer, NULL);
dropped = TRUE;
GST_OBJECT_UNLOCK (pad);
gst_aggregator_pad_drop_buffer (aggpad);
continue;
}
}
if (pad->priv->output_offset >= aagg->priv->offset
&& pad->priv->output_offset <
aagg->priv->offset + blocksize && pad->priv->buffer) {
GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer,
outbuf);
if (pad->priv->output_offset >= next_offset) {
GST_DEBUG_OBJECT (pad,
"Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
G_GINT64_FORMAT, pad->priv->output_offset, next_offset);
} else {
is_done = FALSE;
}
}
GST_OBJECT_UNLOCK (pad);
if (drop_buf)
gst_aggregator_pad_drop_buffer (aggpad);
}
GST_OBJECT_UNLOCK (agg);
if (dropped) {
/* We dropped a buffer, retry */
GST_INFO_OBJECT (aagg, "A pad dropped a buffer, wait for the next one");
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
}
if (!is_done && !is_eos) {
/* Get more buffers */
GST_INFO_OBJECT (aagg,
"We're not done yet for the current offset," " waiting for more data");
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_OK;
}
if (is_eos) {
gint64 max_offset = 0;
GST_DEBUG_OBJECT (aagg, "We're EOS");
GST_OBJECT_LOCK (agg);
for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
max_offset = MAX ((gint64) max_offset, (gint64) pad->priv->output_offset);
}
GST_OBJECT_UNLOCK (agg);
/* This means EOS or nothing mixed in at all */
if (aagg->priv->offset == max_offset) {
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return GST_FLOW_EOS;
}
if (max_offset <= next_offset) {
GST_DEBUG_OBJECT (aagg,
"Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
G_GINT64_FORMAT, max_offset, next_offset);
next_offset = max_offset;
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);
}
}
/* set timestamps on the output buffer */
GST_OBJECT_LOCK (agg);
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;
} 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_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;
/* 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.
*/
if (timeout) {
for (iter = element->sinkpads; iter; iter = iter->next) {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data);
GST_OBJECT_LOCK (pad);
if (pad->priv->output_offset < aagg->priv->offset)
pad->priv->output_offset = -1;
GST_OBJECT_UNLOCK (pad);
}
}
GST_OBJECT_UNLOCK (agg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
return ret;
/* ERRORS */
not_negotiated:
{
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
GST_ELEMENT_ERROR (aagg, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
