mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-11 19:06:33 +00:00
Merge branch 'master' into 0.11
Conflicts: gst-libs/gst/audio/gstaudioencoder.c gst/playback/gstplaybin2.c gst/videotestsrc/videotestsrc.c
This commit is contained in:
commit
19346c2c3b
2 changed files with 27 additions and 33 deletions
|
@ -454,20 +454,13 @@ gst_audio_encoder_finalize (GObject * object)
|
|||
* @buffer: encoded data
|
||||
* @samples: number of samples (per channel) represented by encoded data
|
||||
*
|
||||
* Collects encoded data and/or pushes encoded data downstream.
|
||||
* Source pad caps must be set when this is called. Depending on the nature
|
||||
* of the (framing of) the format, subclass can decide whether to push
|
||||
* encoded data directly or to collect various "frames" in a single buffer.
|
||||
* Note that the latter behaviour is recommended whenever the format is allowed,
|
||||
* as it incurs no additional latency and avoids otherwise generating a
|
||||
* a multitude of (small) output buffers. If not explicitly pushed,
|
||||
* any available encoded data is pushed at the end of each processing cycle,
|
||||
* i.e. which encodes as much data as available input data allows.
|
||||
* Collects encoded data and pushes encoded data downstream.
|
||||
* Source pad caps must be set when this is called.
|
||||
*
|
||||
* If @samples < 0, then best estimate is all samples provided to encoder
|
||||
* (subclass) so far. @buf may be NULL, in which case next number of @samples
|
||||
* are considered discarded, e.g. as a result of discontinuous transmission,
|
||||
* and a discontinuity is marked (note that @buf == NULL => push == TRUE).
|
||||
* and a discontinuity is marked.
|
||||
*
|
||||
* Returns: a #GstFlowReturn that should be escalated to caller (of caller)
|
||||
*
|
||||
|
@ -494,6 +487,26 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
|
|||
|
||||
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
|
||||
|
||||
GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
|
||||
buf ? gst_buffer_get_size (buf) : -1, samples);
|
||||
|
||||
/* mark subclass still alive and providing */
|
||||
if (G_LIKELY (buf))
|
||||
priv->got_data = TRUE;
|
||||
|
||||
if (priv->pending_events) {
|
||||
GList *pending_events, *l;
|
||||
|
||||
pending_events = priv->pending_events;
|
||||
priv->pending_events = NULL;
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "Pushing pending events");
|
||||
for (l = pending_events; l; l = l->next)
|
||||
gst_pad_push_event (enc->srcpad, l->data);
|
||||
g_list_free (pending_events);
|
||||
}
|
||||
|
||||
/* send after pending events, which likely includes newsegment event */
|
||||
if (G_UNLIKELY (enc->priv->tags)) {
|
||||
GstTagList *tags;
|
||||
#if 0
|
||||
|
@ -510,29 +523,10 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
|
|||
gst_pb_utils_add_codec_description_to_tag_list (tags, GST_TAG_AUDIO_CODEC,
|
||||
caps);
|
||||
#endif
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "sending tags %" GST_PTR_FORMAT, tags);
|
||||
gst_element_found_tags_for_pad (GST_ELEMENT (enc), enc->srcpad, tags);
|
||||
}
|
||||
|
||||
GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
|
||||
buf ? gst_buffer_get_size (buf) : -1, samples);
|
||||
|
||||
/* mark subclass still alive and providing */
|
||||
priv->got_data = TRUE;
|
||||
|
||||
if (priv->pending_events) {
|
||||
GList *pending_events, *l;
|
||||
|
||||
pending_events = priv->pending_events;
|
||||
priv->pending_events = NULL;
|
||||
|
||||
GST_DEBUG_OBJECT (enc, "Pushing pending events");
|
||||
for (l = priv->pending_events; l; l = l->next)
|
||||
gst_pad_push_event (enc->srcpad, l->data);
|
||||
g_list_free (pending_events);
|
||||
}
|
||||
|
||||
/* remove corresponding samples from input */
|
||||
if (samples < 0)
|
||||
samples = (enc->priv->offset / ctx->info.bpf);
|
||||
|
@ -1287,10 +1281,10 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
|
|||
gst_tag_list_remove_tag (tags, GST_TAG_AUDIO_CODEC);
|
||||
event = gst_event_new_tag (tags);
|
||||
|
||||
GST_OBJECT_LOCK (enc);
|
||||
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
|
||||
enc->priv->pending_events =
|
||||
g_list_append (enc->priv->pending_events, event);
|
||||
GST_OBJECT_UNLOCK (enc);
|
||||
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
|
||||
handled = TRUE;
|
||||
break;
|
||||
}
|
||||
|
|
|
@ -2612,7 +2612,7 @@ pad_added_cb (GstElement * decodebin, GstPad * pad, GstSourceGroup * group)
|
|||
G_CALLBACK (notify_tags_cb), ntdata, (GClosureNotify) g_free,
|
||||
(GConnectFlags) 0);
|
||||
g_object_set_data (G_OBJECT (sinkpad), "playbin.notify_tags_handler",
|
||||
(gpointer) notify_tags_handler);
|
||||
(gpointer) (guintptr) notify_tags_handler);
|
||||
|
||||
/* store the pad in the array */
|
||||
GST_DEBUG_OBJECT (playbin, "pad %p added to array", sinkpad);
|
||||
|
@ -2714,7 +2714,7 @@ pad_removed_cb (GstElement * decodebin, GstPad * pad, GstSourceGroup * group)
|
|||
gulong notify_tags_handler;
|
||||
|
||||
notify_tags_handler =
|
||||
(gulong) g_object_get_data (G_OBJECT (peer),
|
||||
(guintptr) g_object_get_data (G_OBJECT (peer),
|
||||
"playbin.notify_tags_handler");
|
||||
if (notify_tags_handler != 0)
|
||||
g_signal_handler_disconnect (G_OBJECT (peer), notify_tags_handler);
|
||||
|
|
Loading…
Reference in a new issue