mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-06 14:45:54 +00:00
Code Issues 1 Projects Releases Wiki Activity

audiortppay: add some debugging

Browse source
This commit is contained in:
Wim Taymans 2009-09-03 18:53:19 +02:00
parent c1db9ebb20
commit 7a7663476f
1 changed files with 10 additions and 2 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

12
gst-libs/gst/rtp/gstbasertpaudiopayload.c
View file

@ -668,11 +668,19 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
* output buffers containing the same RTP timestamp gap as the gap
* between the GST timestamps. */
if (timestamp > priv->last_timestamp) {
GstClockTime diff;
guint64 bytes;
/* we're only going to apply a positive gap, otherwise we let the marker
* bit do its thing. simply convert to bytes and add the the current
* offset */
priv->offset +=
priv->time_to_bytes (payload, timestamp - priv->last_timestamp);
diff = timestamp - priv->last_timestamp;
bytes = priv->time_to_bytes (payload, diff);
priv->offset += bytes;
GST_DEBUG_OBJECT (payload,
"elapsed time %" GST_TIME_FORMAT ", bytes %" G_GUINT64_FORMAT
", new offset %" G_GUINT64_FORMAT, GST_TIME_ARGS (diff), bytes,
priv->offset);
}
}
}