mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-12 09:26:20 +00:00
Code Issues 1 Projects Releases Wiki Activity

gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.

Browse source 
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
This commit is contained in:
Wim Taymans 2007-09-04 16:18:48 +00:00
parent 417107b40e
commit 56e39e7c1c
4 changed files with 31 additions and 2 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

14
ChangeLog
View file

@ -1,3 +1,17 @@
2007-09-04 Wim Taymans <wim.taymans@gmail.com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_payload_audio_handle_event):
Return FALSE from the event handler to let the parent class handle the
event.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
* gst-libs/gst/rtp/gstbasertppayload.c:
Bump the MTU to 1400.
2007-09-03 Johan Dahlin <jdahlin@async.com.br>
* gst/typefind/gsttypefindfunctions.c (plugin_init):

3
gst-libs/gst/rtp/gstbasertpaudiopayload.c
View file

@ -621,7 +621,7 @@ static gboolean
gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
{
GstBaseRTPAudioPayload *basertpaudiopayload;
gboolean res = TRUE;
gboolean res = FALSE;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (gst_pad_get_parent (pad));
@ -642,6 +642,7 @@ gst_base_rtp_payload_audio_handle_event (GstPad * pad, GstEvent * event)
gst_object_unref (basertpaudiopayload);
/* return FALSE to let parent handle the remainder of the event */
return res;
}

14
gst-libs/gst/rtp/gstbasertpdepayload.c
View file

@ -56,6 +56,8 @@ struct _GstBaseRTPDepayloadPrivate
gdouble play_scale;
GstClockTime exttimestamp;
gboolean discont;
};
/* Filter signals and args */
@ -248,6 +250,7 @@ static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
@ -257,6 +260,9 @@ gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
if (filter->clock_rate == 0)
goto not_configured;
priv = filter->priv;
priv->discont = GST_BUFFER_IS_DISCONT (in);
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
/* let's send it out to processing */
@ -332,6 +338,9 @@ gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
GstFlowReturn ret;
GstCaps *srccaps;
GstBaseRTPDepayloadClass *bclass;
GstBaseRTPDepayloadPrivate *priv;
priv = filter->priv;
/* set the caps if any */
srccaps = GST_PAD_CAPS (filter->srcpad);
@ -344,6 +353,11 @@ gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter,
if (bclass->set_gst_timestamp && do_ts)
bclass->set_gst_timestamp (filter, timestamp, out_buf);
if (priv->discont) {
GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* push it */
GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (out_buf),

2
gst-libs/gst/rtp/gstbasertppayload.c
View file

@ -60,7 +60,7 @@ enum
* 1432 bytes or so. And that should be adjusted downward further for other
* encapsulations like PPPoE, so 1400 at most.
*/
#define DEFAULT_MTU 1024
#define DEFAULT_MTU 1400
#define DEFAULT_PT 96
#define DEFAULT_SSRC -1
#define DEFAULT_TIMESTAMP_OFFSET -1