gboolean
gst_audio_aggregator_set_src_caps (GstAudioAggregator * aagg, GstCaps * caps)
{
GstAudioInfo info;
if (!gst_audio_info_from_caps (&info, caps)) {
GST_WARNING_OBJECT (aagg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps);
return FALSE;
}
GST_AUDIO_AGGREGATOR_LOCK (aagg);
GST_OBJECT_LOCK (aagg);
if (!gst_audio_info_is_equal (&info, &aagg->info)) {
GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
gst_caps_replace (&aagg->current_caps, caps);
memcpy (&aagg->info, &info, sizeof (info));
aagg->priv->send_caps = TRUE;
}
GST_OBJECT_UNLOCK (aagg);
GST_AUDIO_AGGREGATOR_UNLOCK (aagg);
/* send caps event later, after stream-start event */
return TRUE;
}
/* 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_adder_setcaps (GstAdder * adder, GstPad * pad, GstCaps * orig_caps)
{
GstCaps *caps;
GstAudioInfo info;
GstStructure *s;
gint channels;
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;
GST_OBJECT_LOCK (adder);
/* 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 (adder->current_caps != NULL) {
if (gst_audio_info_is_equal (&info, &adder->info)) {
GST_OBJECT_UNLOCK (adder);
gst_caps_unref (caps);
return TRUE;
} else {
GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
"current caps are %" GST_PTR_FORMAT, caps, adder->current_caps);
GST_OBJECT_UNLOCK (adder);
gst_pad_push_event (pad, gst_event_new_reconfigure ());
gst_caps_unref (caps);
return FALSE;
}
}
GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps);
adder->current_caps = gst_caps_ref (caps);
memcpy (&adder->info, &info, sizeof (info));
GST_OBJECT_UNLOCK (adder);
/* send caps event later, after stream-start event */
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 (adder, "invalid format set as caps");
return FALSE;
}
}
/* 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_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
GstBuffer ** buf)
{
GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
GstCaps *caps;
GstBuffer *buffer;
guint n, bpf;
guint64 period_time;
guint64 period_samples;
GST_DEBUG_OBJECT (interaudiosrc, "create");
buffer = NULL;
caps = NULL;
g_mutex_lock (&interaudiosrc->surface->mutex);
if (interaudiosrc->surface->audio_info.finfo) {
if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
&interaudiosrc->info)) {
caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
interaudiosrc->timestamp_offset +=
gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
interaudiosrc->n_samples = 0;
}
}
bpf = interaudiosrc->surface->audio_info.bpf;
period_time = interaudiosrc->surface->audio_period_time;
period_samples =
gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND);
if (bpf > 0)
n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
else
n = 0;
if (n > period_samples)
n = period_samples;
if (n > 0) {
buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
n * bpf);
} else {
buffer = gst_buffer_new ();
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
}
g_mutex_unlock (&interaudiosrc->surface->mutex);
if (caps) {
gboolean ret = gst_base_src_set_caps (src, caps);
gst_caps_unref (caps);
if (!ret) {
GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps);
if (buffer)
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
}
}
buffer = gst_buffer_make_writable (buffer);
bpf = interaudiosrc->info.bpf;
if (n < period_samples) {
GstMapInfo map;
GstMemory *mem;
GST_DEBUG_OBJECT (interaudiosrc,
"creating %" G_GUINT64_FORMAT " samples of silence",
period_samples - n);
mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL);
if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
gst_audio_format_fill_silence (interaudiosrc->info.finfo, map.data,
map.size);
gst_memory_unmap (mem, &map);
}
gst_buffer_prepend_memory (buffer, mem);
}
n = period_samples;
GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
GST_BUFFER_TIMESTAMP (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
interaudiosrc->info.rate);
GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
gst_util_uint64_scale (interaudiosrc->n_samples + n, GST_SECOND,
interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
if (interaudiosrc->n_samples == 0) {
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
}
interaudiosrc->n_samples += n;
*buf = buffer;
return GST_FLOW_OK;
}