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audioencoder: Delay sending of serialized events to finish_frame()

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This makes sure that the caps are already set before any serialized
events are sent downstream.
This commit is contained in:
Sebastian Dröge 2011-09-26 15:42:14 +02:00
parent 11e375486e
commit 61ffd7cb42
1 changed files with 51 additions and 3 deletions
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54
gst-libs/gst/audio/gstaudioencoder.c
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@ -243,6 +243,8 @@ struct _GstAudioEncoderPrivate
/* pending tags */
GstTagList *tags;
/* pending serialized sink events, will be sent from finish_frame() */
GList *pending_events;
};
static void gst_audio_encoder_finalize (GObject * object);
@ -394,6 +396,10 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
if (enc->priv->tags)
gst_tag_list_free (enc->priv->tags);
enc->priv->tags = NULL;
g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (enc->priv->pending_events);
enc->priv->pending_events = NULL;
}
gst_segment_init (&enc->segment, GST_FORMAT_TIME);
@ -485,6 +491,19 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
/* mark subclass still alive and providing */
priv->got_data = TRUE;
if (priv->pending_events) {
GList *pending_events, *l;
GST_OBJECT_LOCK (enc);
pending_events = priv->pending_events;
priv->pending_events = NULL;
GST_OBJECT_UNLOCK (enc);
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);
@ -1192,6 +1211,13 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
klass->flush (enc);
/* and get (re)set for the sequel */
gst_audio_encoder_reset (enc, FALSE);
GST_OBJECT_LOCK (enc);
g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (enc->priv->pending_events);
enc->priv->pending_events = NULL;
GST_OBJECT_UNLOCK (enc);
break;
case GST_EVENT_EOS:
@ -1209,7 +1235,10 @@ 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_pad_push_event (enc->srcpad, event);
GST_OBJECT_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_OBJECT_UNLOCK (enc);
handled = TRUE;
break;
}
@ -1241,8 +1270,27 @@ gst_audio_encoder_sink_event (GstPad * pad, GstEvent * event)
if (!handled)
handled = gst_audio_encoder_sink_eventfunc (enc, event);
if (!handled)
ret = gst_pad_event_default (pad, event);
if (!handled) {
/* Forward non-serialized events and EOS/FLUSH_STOP immediately.
* For EOS this is required because no buffer or serialized event
* will come after EOS and nothing could trigger another
* _finish_frame() call.
*
* For FLUSH_STOP this is required because it is expected
* to be forwarded immediately and no buffers are queued anyway.
*/
if (!GST_EVENT_IS_SERIALIZED (event)
|| GST_EVENT_TYPE (event) == GST_EVENT_EOS
|| GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
ret = gst_pad_event_default (pad, event);
} else {
GST_OBJECT_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_OBJECT_UNLOCK (enc);
ret = TRUE;
}
}
GST_DEBUG_OBJECT (enc, "event handled");