From e6d85e6a1e51eb39155591dd5dc1a4b06f1267ed Mon Sep 17 00:00:00 2001 From: Peter Kjellerstedt Date: Thu, 3 Jul 2008 14:31:10 +0000 Subject: 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. --- ChangeLog | 13 +++++++++++++ gst/rtpmanager/gstrtpsession.c | 14 +++++++------- gst/rtpmanager/rtpsession.c | 6 +++--- gst/rtpmanager/rtpsource.c | 17 ++++++++--------- 4 files changed, 31 insertions(+), 19 deletions(-) diff --git a/ChangeLog b/ChangeLog index 8f9765c4..f463c12d 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,16 @@ +2008-07-03 Peter Kjellerstedt + + * 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. + 2008-07-03 Peter Kjellerstedt * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init), 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"); -- cgit v1.2.1