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rtpsession: Only warn once if configured latency needs to be known but isn't yet

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Otherwise we would warn about this once for every single packet until
the LATENCY event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5854>
This commit is contained in:
Sebastian Dröge 2023-12-22 12:56:33 +02:00 committed by GStreamer Marge Bot
parent 04ae39f25b
commit c292da7044
1 changed files with 12 additions and 2 deletions
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14
subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c
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@ -276,6 +276,9 @@ struct _GstRtpSessionPrivate
GHashTable *ptmap; GHashTable *ptmap;
GstClockTime send_latency; GstClockTime send_latency;
/* Set if we warned once already that no latency is configured yet but we
* need it to calculate correct send running time of the packets */
gboolean warned_latency_once;
gboolean use_pipeline_clock; gboolean use_pipeline_clock;
GstRtpNtpTimeSource ntp_time_source; GstRtpNtpTimeSource ntp_time_source;
@ -1367,6 +1370,7 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
GST_RTP_SESSION_LOCK (rtpsession); GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->wait_send = TRUE; rtpsession->priv->wait_send = TRUE;
rtpsession->priv->send_latency = GST_CLOCK_TIME_NONE; rtpsession->priv->send_latency = GST_CLOCK_TIME_NONE;
rtpsession->priv->warned_latency_once = FALSE;
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
break; break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING: case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
@ -2475,8 +2479,14 @@ gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
if (priv->send_latency != GST_CLOCK_TIME_NONE) { if (priv->send_latency != GST_CLOCK_TIME_NONE) {
running_time += priv->send_latency; running_time += priv->send_latency;
} else { } else {
GST_WARNING_OBJECT (rtpsession, if (!priv->warned_latency_once) {
"Can't determine running time for this packet without knowing configured latency"); priv->warned_latency_once = TRUE;
GST_WARNING_OBJECT (rtpsession,
"Can't determine running time for this packet without knowing configured latency");
} else {
GST_LOG_OBJECT (rtpsession,
"Can't determine running time for this packet without knowing configured latency");
}
running_time = -1; running_time = -1;
} }
} }