summaryrefslogtreecommitdiffstats
path: root/gst
diff options
context:
space:
mode:
Diffstat (limited to 'gst')
-rw-r--r--gst/rtpmanager/gstrtpsession.c14
-rw-r--r--gst/rtpmanager/rtpsession.c6
-rw-r--r--gst/rtpmanager/rtpsource.c17
3 files changed, 18 insertions, 19 deletions
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c
index a3359e45..b96a1dfc 100644
--- a/gst/rtpmanager/gstrtpsession.c
+++ b/gst/rtpmanager/gstrtpsession.c
@@ -1062,7 +1062,7 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
priv = rtpsession->priv;
if (rtpsession->recv_rtp_src) {
- GST_DEBUG_OBJECT (rtpsession, "pushing received RTP packet");
+ GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
} else {
GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
@@ -1085,7 +1085,7 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
- GST_DEBUG_OBJECT (rtpsession, "sending RTP packet");
+ GST_LOG_OBJECT (rtpsession, "sending RTP packet");
if (rtpsession->send_rtp_src) {
result = gst_pad_push (rtpsession->send_rtp_src, buffer);
@@ -1119,7 +1119,7 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
gst_caps_unref (caps);
}
gst_buffer_set_caps (buffer, caps);
- GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
+ GST_LOG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
} else {
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
@@ -1152,7 +1152,7 @@ gst_rtp_session_sync_rtcp (RTPSession * sess,
gst_caps_unref (caps);
}
gst_buffer_set_caps (buffer, caps);
- GST_DEBUG_OBJECT (rtpsession, "sending Sync RTCP");
+ GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
result = gst_pad_push (rtpsession->sync_src, buffer);
} else {
GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
@@ -1390,7 +1390,7 @@ gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
- GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
+ GST_LOG_OBJECT (rtpsession, "received RTP packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
@@ -1467,7 +1467,7 @@ gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer)
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
- GST_DEBUG_OBJECT (rtpsession, "received RTCP packet");
+ GST_LOG_OBJECT (rtpsession, "received RTCP packet");
current_time = gst_clock_get_time (priv->sysclock);
ret = rtp_session_process_rtcp (priv->session, buffer, current_time);
@@ -1614,7 +1614,7 @@ gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer)
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
- GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
+ GST_LOG_OBJECT (rtpsession, "received RTP packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c
index c7ab410f..59b1d8d0 100644
--- a/gst/rtpmanager/rtpsession.c
+++ b/gst/rtpmanager/rtpsession.c
@@ -812,7 +812,7 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
GstFlowReturn result = GST_FLOW_OK;
if (source == session->source) {
- GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
+ GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
@@ -824,7 +824,7 @@ source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
gst_buffer_unref (buffer);
} else {
- GST_DEBUG ("source %08x pushed receiver RTP packet", source->ssrc);
+ GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.process_rtp)
@@ -1772,7 +1772,7 @@ rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
if (!gst_rtp_buffer_validate (buffer))
goto invalid_packet;
- GST_DEBUG ("received RTP packet for sending");
+ GST_LOG ("received RTP packet for sending");
RTP_SESSION_LOCK (sess);
source = sess->source;
diff --git a/gst/rtpmanager/rtpsource.c b/gst/rtpmanager/rtpsource.c
index 7eab91e0..50170d10 100644
--- a/gst/rtpmanager/rtpsource.c
+++ b/gst/rtpmanager/rtpsource.c
@@ -708,13 +708,13 @@ push_packet (RTPSource * src, GstBuffer * buffer)
while (!g_queue_is_empty (src->packets)) {
GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
- GST_DEBUG ("pushing queued packet");
+ GST_LOG ("pushing queued packet");
if (src->callbacks.push_rtp)
src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
}
- GST_DEBUG ("pushing new packet");
+ GST_LOG ("pushing new packet");
/* push packet */
if (src->callbacks.push_rtp)
ret = src->callbacks.push_rtp (src, buffer, src->user_data);
@@ -763,7 +763,7 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
pt = gst_rtp_buffer_get_payload_type (buffer);
- GST_DEBUG ("SSRC %08x got payload %d", src->ssrc, pt);
+ GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
/* get clockrate */
if ((clock_rate = get_clock_rate (src, pt)) == -1)
@@ -802,7 +802,7 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
src->stats.prev_rtptime = src->stats.last_rtptime;
src->stats.last_rtptime = rtparrival;
- GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
+ GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
return;
@@ -937,7 +937,7 @@ rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
src->is_sender = TRUE;
src->validated = TRUE;
- GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
+ GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received);
/* calculate jitter for the stats */
@@ -1018,7 +1018,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
- GST_DEBUG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
+ GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime));
if (ext_rtptime > src->last_rtptime) {
@@ -1028,7 +1028,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
/* calc the diff so we can detect drift at the sender. This can also be used
* to guestimate the clock rate if the NTP time is locked to the RTP
* timestamps (as is the case when the capture device is providing the clock). */
- GST_DEBUG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
+ GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff));
}
@@ -1053,8 +1053,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
src->ssrc);
gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
}
- GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
- src->stats.packets_sent);
+ GST_LOG ("pushing RTP packet %" G_GUINT64_FORMAT, src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);
} else {
GST_WARNING ("no callback installed, dropping packet");
generated by cgit v1.2.1 (git 2.18.0) at 2024-12-24 03:28:52 +0000