static gboolean
gst_audio_aggregator_sink_event (GstAggregator * agg,
GstAggregatorPad * aggpad, GstEvent * event)
{
gboolean res = TRUE;
GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEGMENT:
{
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
if (segment->format != GST_FORMAT_TIME) {
GST_ERROR_OBJECT (agg, "Segment of type %s are not supported,"
" only TIME segments are supported",
gst_format_get_name (segment->format));
gst_event_unref (event);
event = NULL;
res = FALSE;
break;
}
GST_OBJECT_LOCK (agg);
if (segment->rate != agg->segment.rate) {
GST_ERROR_OBJECT (aggpad,
"Got segment event with wrong rate %lf, expected %lf",
segment->rate, agg->segment.rate);
res = FALSE;
gst_event_unref (event);
event = NULL;
} else if (segment->rate < 0.0) {
GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet");
res = FALSE;
gst_event_unref (event);
event = NULL;
} else {
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GST_OBJECT_LOCK (pad);
pad->priv->new_segment = TRUE;
GST_OBJECT_UNLOCK (pad);
}
GST_OBJECT_UNLOCK (agg);
break;
}
default:
break;
}
if (event != NULL)
return
GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event
(agg, aggpad, event);
return res;
}
static gboolean
gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
GstAggregator * aggregator)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
GST_OBJECT_LOCK (aggpad);
pad->priv->position = pad->priv->size = 0;
pad->priv->output_offset = pad->priv->next_offset = -1;
pad->priv->discont_time = GST_CLOCK_TIME_NONE;
gst_buffer_replace (&pad->priv->buffer, NULL);
GST_OBJECT_UNLOCK (aggpad);
return TRUE;
}
static GstBuffer *
gst_audio_aggregator_do_clip (GstAggregator * agg,
GstAggregatorPad * bpad, GstBuffer * buffer)
{
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad);
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&pad->info);
bpf = GST_AUDIO_INFO_BPF (&pad->info);
GST_OBJECT_LOCK (bpad);
buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
GST_OBJECT_UNLOCK (bpad);
return buffer;
}
/* the first caps we receive on any of the sinkpads will define the caps for all
* the other sinkpads because we can only mix streams with the same caps.
*/
static gboolean
gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad,
GstCaps * orig_caps)
{
GstAggregator *agg = GST_AGGREGATOR (audiomixer);
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (audiomixer);
GstCaps *caps;
GstAudioInfo info;
GstStructure *s;
gint channels = 0;
caps = gst_caps_copy (orig_caps);
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "channels", &channels))
if (channels <= 2)
gst_structure_remove_field (s, "channel-mask");
if (!gst_audio_info_from_caps (&info, caps))
goto invalid_format;
if (channels == 1) {
GstCaps *filter;
GstCaps *downstream_caps;
if (audiomixer->filter_caps)
filter = gst_caps_intersect_full (caps, audiomixer->filter_caps,
GST_CAPS_INTERSECT_FIRST);
else
filter = gst_caps_ref (caps);
downstream_caps = gst_pad_peer_query_caps (agg->srcpad, filter);
gst_caps_unref (filter);
if (downstream_caps) {
gst_caps_unref (caps);
caps = downstream_caps;
if (gst_caps_is_empty (caps)) {
gst_caps_unref (caps);
return FALSE;
}
caps = gst_caps_fixate (caps);
}
}
GST_OBJECT_LOCK (audiomixer);
/* don't allow reconfiguration for now; there's still a race between the
* different upstream threads doing query_caps + accept_caps + sending
* (possibly different) CAPS events, but there's not much we can do about
* that, upstream needs to deal with it. */
if (aagg->current_caps != NULL) {
if (gst_audio_info_is_equal (&info, &aagg->info)) {
GST_OBJECT_UNLOCK (audiomixer);
gst_caps_unref (caps);
gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
orig_caps);
return TRUE;
} else {
GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
"current caps are %" GST_PTR_FORMAT, caps, aagg->current_caps);
GST_OBJECT_UNLOCK (audiomixer);
gst_pad_push_event (pad, gst_event_new_reconfigure ());
gst_caps_unref (caps);
return FALSE;
}
} else {
gst_caps_replace (&aagg->current_caps, caps);
aagg->info = info;
gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (agg));
}
GST_OBJECT_UNLOCK (audiomixer);
gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
orig_caps);
GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
return TRUE;
/* ERRORS */
invalid_format:
{
gst_caps_unref (caps);
GST_WARNING_OBJECT (audiomixer, "invalid format set as caps");
return FALSE;
}
}
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;
}
}
