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audioencoder: use default event function

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Implement a default event function so that subclasses can call it without having
to return FALSE (and make it impossible to report errors).
This commit is contained in:
Wim Taymans 2012-02-27 12:49:52 +01:00
parent 9f1732fba7
commit 15eb385412
1 changed files with 34 additions and 36 deletions
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70
gst-libs/gst/audio/gstaudioencoder.c
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@ -303,6 +303,8 @@ static gboolean gst_audio_encoder_sink_activate_mode (GstPad * pad,
static GstCaps *gst_audio_encoder_getcaps_default (GstAudioEncoder * enc,
GstCaps * filter);
static gboolean gst_audio_encoder_sink_event_default (GstAudioEncoder * enc,
GstEvent * event);
static gboolean gst_audio_encoder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_audio_encoder_sink_setcaps (GstAudioEncoder * enc,
@ -352,6 +354,7 @@ gst_audio_encoder_class_init (GstAudioEncoderClass * klass)
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
klass->getcaps = gst_audio_encoder_getcaps_default;
klass->event = gst_audio_encoder_sink_event_default;
}
static void
@ -1250,10 +1253,10 @@ gst_audio_encoder_getcaps_default (GstAudioEncoder * enc, GstCaps * filter)
}
static gboolean
gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
gst_audio_encoder_sink_event_default (GstAudioEncoder * enc, GstEvent * event)
{
GstAudioEncoderClass *klass;
gboolean handled = FALSE;
gboolean res;
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
@ -1270,6 +1273,7 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
} else {
GST_DEBUG_OBJECT (enc, "received SEGMENT %" GST_SEGMENT_FORMAT, &seg);
GST_DEBUG_OBJECT (enc, "unsupported format; ignoring");
res = TRUE;
break;
}
@ -1281,10 +1285,13 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
/* and follow along with segment */
enc->segment = seg;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = TRUE;
break;
}
case GST_EVENT_FLUSH_START:
res = gst_pad_push_event (enc->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
@ -1301,12 +1308,18 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
enc->priv->pending_events = NULL;
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = gst_pad_push_event (enc->srcpad, event);
break;
case GST_EVENT_EOS:
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_audio_encoder_drain (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
/* forward immediately because no buffer or serialized event
* will come after EOS and nothing could trigger another
* _finish_frame() call. */
res = gst_pad_push_event (enc->srcpad, event);
break;
case GST_EVENT_TAG:
@ -1335,7 +1348,7 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
handled = TRUE;
res = TRUE;
break;
}
@ -1344,17 +1357,26 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
GstCaps *caps;
gst_event_parse_caps (event, &caps);
gst_audio_encoder_sink_setcaps (enc, caps);
res = gst_audio_encoder_sink_setcaps (enc, caps);
gst_event_unref (event);
handled = TRUE;
break;
}
default:
/* Forward non-serialized events immediately. */
if (!GST_EVENT_IS_SERIALIZED (event)) {
res =
gst_pad_event_default (enc->sinkpad, GST_OBJECT_CAST (enc), event);
} else {
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
res = TRUE;
}
break;
}
return handled;
return res;
}
static gboolean
@ -1363,8 +1385,7 @@ gst_audio_encoder_sink_event (GstPad * pad, GstObject * parent,
{
GstAudioEncoder *enc;
GstAudioEncoderClass *klass;
gboolean handled = FALSE;
gboolean ret = TRUE;
gboolean ret;
enc = GST_AUDIO_ENCODER (parent);
klass = GST_AUDIO_ENCODER_GET_CLASS (enc);
@ -1373,34 +1394,11 @@ gst_audio_encoder_sink_event (GstPad * pad, GstObject * parent,
GST_EVENT_TYPE_NAME (event));
if (klass->event)
handled = klass->event (enc, event);
ret = klass->event (enc, event);
else
ret = FALSE;
if (!handled)
handled = gst_audio_encoder_sink_eventfunc (enc, 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, parent, event);
} else {
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
ret = TRUE;
}
}
GST_DEBUG_OBJECT (enc, "event handled");
GST_DEBUG_OBJECT (enc, "event result %d", ret);
return ret;
}