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authorWim Taymans <wim.taymans@gmail.com>2007-09-17 02:01:41 +0000
committerWim Taymans <wim.taymans@gmail.com>2007-09-17 02:01:41 +0000
commitc0aa28ca5bfc2eaf6b9cbba1159e9490c7c118c4 (patch)
tree6cbc87e37fd54f7e593a59656ea09358473c391f
parent04d3b8290698e41034809e8baec11622ca128243 (diff)
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gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found): Link to the right pads regardless of which one was created first in the ssrc demuxer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsource.c: (calculate_jitter): Improve debugging. * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event), (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_internal_links): * gst/rtpmanager/gstrtpssrcdemux.h: Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
-rw-r--r--ChangeLog22
-rw-r--r--gst/rtpmanager/gstrtpbin.c7
-rw-r--r--gst/rtpmanager/gstrtpjitterbuffer.c6
-rw-r--r--gst/rtpmanager/gstrtpsession.c27
-rw-r--r--gst/rtpmanager/gstrtpssrcdemux.c36
-rw-r--r--gst/rtpmanager/gstrtpssrcdemux.h2
-rw-r--r--gst/rtpmanager/rtpsource.c2
7 files changed, 78 insertions, 24 deletions
diff --git a/ChangeLog b/ChangeLog
index 368134d3..00260f97 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,27 @@
2007-09-16 Wim Taymans <wim.taymans@gmail.com>
+ * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
+ Link to the right pads regardless of which one was created first in the
+ ssrc demuxer.
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+ * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
+ (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
+ * gst/rtpmanager/rtpsource.c: (calculate_jitter):
+ Improve debugging.
+
+ * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
+ (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
+ (gst_rtp_ssrc_demux_sink_event),
+ (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
+ (gst_rtp_ssrc_demux_rtcp_chain),
+ (gst_rtp_ssrc_demux_internal_links):
+ * gst/rtpmanager/gstrtpssrcdemux.h:
+ Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
+
+2007-09-16 Wim Taymans <wim.taymans@gmail.com>
+
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
Use lock to protect variable.
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c
index cdbdaf64..035f4cc9 100644
--- a/gst/rtpmanager/gstrtpbin.c
+++ b/gst/rtpmanager/gstrtpbin.c
@@ -1404,8 +1404,11 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
/* get pad and link */
GST_DEBUG_OBJECT (session->bin, "linking jitterbuffer");
+ padname = g_strdup_printf ("src_%d", ssrc);
+ srcpad = gst_element_get_pad (element, padname);
+ g_free (padname);
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
- gst_pad_link (pad, sinkpad);
+ gst_pad_link (srcpad, sinkpad);
gst_object_unref (sinkpad);
/* get the RTCP sync pad */
@@ -1434,7 +1437,7 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
no_stream:
{
GST_RTP_SESSION_UNLOCK (session);
- GST_DEBUG ("could not create stream");
+ GST_DEBUG_OBJECT (session->bin, "could not create stream");
return;
}
}
diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c
index 57be42ec..d27c2aa1 100644
--- a/gst/rtpmanager/gstrtpjitterbuffer.c
+++ b/gst/rtpmanager/gstrtpjitterbuffer.c
@@ -797,6 +797,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
guint16 seqnum;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
+ guint64 latency_ts;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
@@ -849,7 +850,6 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
* latency is set, we just pump it in the queue and let the other end push it
* out as fast as possible. */
if (priv->latency_ms && priv->drop_on_latency) {
- guint64 latency_ts;
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
@@ -1053,8 +1053,8 @@ again:
if (priv->next_seqnum != -1) {
/* we expected next_seqnum but received something else, that's a gap */
GST_WARNING_OBJECT (jitterbuffer,
- "Sequence number GAP detected -> %d instead of %d", priv->next_seqnum,
- seqnum);
+ "Sequence number GAP detected: expected %d instead of %d",
+ priv->next_seqnum, seqnum);
} else {
/* we don't know what the next_seqnum should be, wait for the last
* possible moment to push this buffer, maybe we get an earlier seqnum
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index 6d4cf8b0..83210ff8 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -781,11 +781,11 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
- GST_DEBUG_OBJECT (rtpsession, "reading receiving RTP packet");
-
if (rtpsession->recv_rtp_src) {
+ GST_DEBUG_OBJECT (rtpsession, "pushing received RTP packet");
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
} else {
+ GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
@@ -1114,10 +1114,22 @@ gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
}
ret = rtp_session_process_rtp (priv->session, buffer, ntpnstime);
+ if (ret != GST_FLOW_OK)
+ goto push_error;
+
+done:
gst_object_unref (rtpsession);
return ret;
+
+ /* ERRORS */
+push_error:
+ {
+ GST_DEBUG_OBJECT (rtpsession, "process returned %s",
+ gst_flow_get_name (ret));
+ goto done;
+ }
}
static GstFlowReturn
@@ -1286,10 +1298,21 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
}
ret = rtp_session_send_rtp (priv->session, buffer, ntpnstime);
+ if (ret != GST_FLOW_OK)
+ goto push_error;
+done:
gst_object_unref (rtpsession);
return ret;
+
+ /* ERRORS */
+push_error:
+ {
+ GST_DEBUG_OBJECT (rtpsession, "process returned %s",
+ gst_flow_get_name (ret));
+ goto done;
+ }
}
/* Create sinkpad to receive RTP packets from senders. This will also create a
diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c
index 2f1e9e67..a5956d1c 100644
--- a/gst/rtpmanager/gstrtpssrcdemux.c
+++ b/gst/rtpmanager/gstrtpssrcdemux.c
@@ -96,6 +96,9 @@ static GstElementDetails gst_rtp_ssrc_demux_details = {
"Wim Taymans <wim@fluendo.com>"
};
+#define GST_PAD_LOCK(obj) (g_mutex_lock ((obj)->padlock))
+#define GST_PAD_UNLOCK(obj) (g_mutex_unlock ((obj)->padlock))
+
/* signals */
enum
{
@@ -159,6 +162,7 @@ find_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
return NULL;
}
+/* with PAD_LOCK */
static GstRtpSsrcDemuxPad *
create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc,
GstClockTime timestamp)
@@ -202,9 +206,6 @@ create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc,
demux->srcpads = g_slist_prepend (demux->srcpads, demuxpad);
- /* unlock to perform the remainder and to fire our signal */
- GST_OBJECT_UNLOCK (demux);
-
/* copy caps from input */
gst_pad_set_caps (rtp_pad, GST_PAD_CAPS (demux->rtp_sink));
gst_pad_use_fixed_caps (rtp_pad);
@@ -227,8 +228,6 @@ create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc,
g_signal_emit (G_OBJECT (demux),
gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD], 0, ssrc, rtp_pad);
- GST_OBJECT_LOCK (demux);
-
return demuxpad;
}
@@ -304,6 +303,8 @@ gst_rtp_ssrc_demux_init (GstRtpSsrcDemux * demux,
gst_rtp_ssrc_demux_rtcp_sink_event);
gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtcp_sink);
+ demux->padlock = g_mutex_new ();
+
gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
}
@@ -327,6 +328,7 @@ gst_rtp_ssrc_demux_finalize (GObject * object)
GstRtpSsrcDemux *demux;
demux = GST_RTP_SSRC_DEMUX (object);
+ g_mutex_free (demux->padlock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@@ -349,14 +351,14 @@ gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event)
GSList *walk;
res = TRUE;
- GST_OBJECT_LOCK (demux);
+ GST_PAD_LOCK (demux);
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
gst_event_ref (event);
res &= gst_pad_push_event (pad->rtp_pad, event);
}
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
gst_event_unref (event);
break;
}
@@ -381,13 +383,13 @@ gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad, GstEvent * event)
GSList *walk;
res = TRUE;
- GST_OBJECT_LOCK (demux);
+ GST_PAD_LOCK (demux);
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
res &= gst_pad_push_event (pad->rtcp_pad, event);
}
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
break;
}
}
@@ -412,7 +414,7 @@ gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf)
GST_DEBUG_OBJECT (demux, "received buffer of SSRC %08x", ssrc);
- GST_OBJECT_LOCK (demux);
+ GST_PAD_LOCK (demux);
dpad = find_demux_pad_for_ssrc (demux, ssrc);
if (dpad == NULL) {
if (!(dpad =
@@ -420,7 +422,7 @@ gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf)
GST_BUFFER_TIMESTAMP (buf))))
goto create_failed;
}
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
/* push to srcpad */
ret = gst_pad_push (dpad->rtp_pad, buf);
@@ -440,7 +442,7 @@ create_failed:
{
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Could not create new pad"));
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
@@ -479,14 +481,14 @@ gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad, GstBuffer * buf)
GST_DEBUG_OBJECT (demux, "received RTCP of SSRC %08x", ssrc);
- GST_OBJECT_LOCK (demux);
+ GST_PAD_LOCK (demux);
dpad = find_demux_pad_for_ssrc (demux, ssrc);
if (dpad == NULL) {
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
if (!(dpad = create_demux_pad_for_ssrc (demux, ssrc, -1)))
goto create_failed;
}
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
/* push to srcpad */
ret = gst_pad_push (dpad->rtcp_pad, buf);
@@ -506,7 +508,7 @@ create_failed:
{
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
("Could not create new pad"));
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
@@ -539,7 +541,7 @@ gst_rtp_ssrc_demux_internal_links (GstPad * pad)
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
- GST_OBJECT_LOCK (demux);
+ GST_PAD_LOCK (demux);
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
@@ -555,7 +557,7 @@ gst_rtp_ssrc_demux_internal_links (GstPad * pad)
break;
}
}
- GST_OBJECT_UNLOCK (demux);
+ GST_PAD_UNLOCK (demux);
gst_object_unref (demux);
return res;
diff --git a/gst/rtpmanager/gstrtpssrcdemux.h b/gst/rtpmanager/gstrtpssrcdemux.h
index 88aeed80..10c569d9 100644
--- a/gst/rtpmanager/gstrtpssrcdemux.h
+++ b/gst/rtpmanager/gstrtpssrcdemux.h
@@ -40,6 +40,8 @@ struct _GstRtpSsrcDemux
GstPad *rtp_sink;
GstPad *rtcp_sink;
+
+ GMutex *padlock;
GSList *srcpads;
};
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 4ffc6bbc..1d7eb24a 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -284,6 +284,8 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
pt = gst_rtp_buffer_get_payload_type (buffer);
+ GST_DEBUG ("SSRC %08x got payload %d", src->ssrc, pt);
+
/* get clockrate */
if ((clock_rate = get_clock_rate (src, pt)) == -1)
goto no_clock_rate;