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authorPeter Kjellerstedt <pkj@axis.com>2008-07-03 14:31:10 +0000
committerTim-Philipp Müller <tim.muller@collabora.co.uk>2009-08-11 02:30:36 +0100
commite2f49d9ccf103fdd25212ce3c261344dc47202b7 (patch)
tree9c2f28ea0bdd88244f4efa1a2127f80c7449db44 /gst/rtpmanager/rtpsource.c
parentca15984e14415a6be286c4744020aa4f9b258979 (diff)
gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
Original commit message from CVS: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp): * gst/rtpmanager/rtpsession.c: (source_push_rtp), (rtp_session_send_rtp): * gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter), (rtp_source_process_rtp), (rtp_source_send_rtp): Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
Diffstat (limited to 'gst/rtpmanager/rtpsource.c')
-rw-r--r--gst/rtpmanager/rtpsource.c17
1 files changed, 8 insertions, 9 deletions
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");