diff options
author | Peter Kjellerstedt <pkj@axis.com> | 2008-07-03 14:31:10 +0000 |
---|---|---|
committer | Tim-Philipp Müller <tim.muller@collabora.co.uk> | 2009-08-11 02:30:36 +0100 |
commit | e2f49d9ccf103fdd25212ce3c261344dc47202b7 (patch) | |
tree | 9c2f28ea0bdd88244f4efa1a2127f80c7449db44 /gst/rtpmanager/rtpsource.c | |
parent | ca15984e14415a6be286c4744020aa4f9b258979 (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.c | 17 |
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"); |