mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-20 15:27:07 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpbin: Add some documentation to gst_rtp_bin_associate()

Browse source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
This commit is contained in:
Sebastian Dröge 2024-04-17 13:29:08 +03:00 committed by GStreamer Marge Bot
parent 70a435c0c4
commit 4b0e75a094
1 changed files with 16 additions and 5 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

21
subprojects/gst-plugins-good/gst/rtpmanager/gstrtpbin.c
View file

@ -1456,10 +1456,21 @@ gst_rtp_bin_send_sync_event (GstRtpBinStream * stream)
/* associate a stream to the given CNAME. This will make sure all streams for /* associate a stream to the given CNAME. This will make sure all streams for
* that CNAME are synchronized together. * that CNAME are synchronized together.
*
* @len: length of CNAME in @data
* @data: CNAME
* @ntpnstime: NTP time in nanoseconds that corresponds to RTP time @extrtptime.
* @extrtptime: Extended RTP time that corresponds to NTP time @ntpnstime.
* @base_rtptime: Extended RTP time that corresponds to the local time @base_time.
* @base_time: Local time that corresponds to RTP time @base_rtptime.
* @clock_rate: RTP clock rate for this stream.
* @rtp_clock_base: RTP time signalled in RTSP RTP-Info for this stream, which
* corresponds to the NPT start of the PLAY request.
*
* Must be called with GST_RTP_BIN_LOCK */ * Must be called with GST_RTP_BIN_LOCK */
static void static void
gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len, gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
const guint8 * data, guint64 ntpnstime, guint64 last_extrtptime, const guint8 * data, guint64 ntpnstime, guint64 extrtptime,
guint64 base_rtptime, guint64 base_time, guint clock_rate, guint64 base_rtptime, guint64 base_time, guint clock_rate,
gint64 rtp_clock_base) gint64 rtp_clock_base)
{ {
@ -1498,12 +1509,12 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
stream->ssrc, client, client->cname); stream->ssrc, client, client->cname);
} }
if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) { if (!GST_CLOCK_TIME_IS_VALID (extrtptime)) {
GST_DEBUG_OBJECT (bin, "invalidated sync data"); GST_DEBUG_OBJECT (bin, "invalidated sync data");
if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) { if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
/* we don't need that data, so carry on, /* we don't need that data, so carry on,
* but make some values look saner */ * but make some values look saner */
last_extrtptime = base_rtptime; extrtptime = base_rtptime;
} else { } else {
/* nothing we can do with this data in this case */ /* nothing we can do with this data in this case */
GST_DEBUG_OBJECT (bin, "bailing out"); GST_DEBUG_OBJECT (bin, "bailing out");
@ -1515,13 +1526,13 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
* local rtptime. The local rtp time is used to construct timestamps on the * local rtptime. The local rtp time is used to construct timestamps on the
* buffers so we will calculate what running_time corresponds to the RTP * buffers so we will calculate what running_time corresponds to the RTP
* timestamp in the SR packet. */ * timestamp in the SR packet. */
running_time_rtp = last_extrtptime - base_rtptime; running_time_rtp = extrtptime - base_rtptime;
GST_DEBUG_OBJECT (bin, GST_DEBUG_OBJECT (bin,
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, " ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
"clock-base %" G_GINT64_FORMAT, base_rtptime, "clock-base %" G_GINT64_FORMAT, base_rtptime,
last_extrtptime, running_time_rtp, clock_rate, rtp_clock_base); extrtptime, running_time_rtp, clock_rate, rtp_clock_base);
/* calculate local RTP time in gstreamer timestamp, we essentially perform the /* calculate local RTP time in gstreamer timestamp, we essentially perform the
* same conversion that a jitterbuffer would use to convert an rtp timestamp * same conversion that a jitterbuffer would use to convert an rtp timestamp