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srtsrc: Don't calculate a delay if the srctime is 0

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A zero srctime is a missing srctime. Apparently this can happen when
["the connection is not between SRT peers or if Timestamp-Based Packet
Delivery mode (TSBPDMODE) is not enabled"][1] so it may not apply to us,
but it's best to be defensive.

[1]: https://github.com/Haivision/srt/blob/v1.4.2/docs/API.md#sending-and-receiving

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1674>
This commit is contained in:
Jan Alexander Steffens (heftig) 2020-10-12 14:12:24 +02:00 committed by GStreamer Merge Bot
parent 6b2fcb52e5
commit ec11ad9d55
1 changed files with 9 additions and 2 deletions
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11
ext/srt/gstsrtsrc.c
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@ -125,6 +125,7 @@ gst_srt_src_fill (GstPushSrc * src, GstBuffer * outbuf)
GstClockTime base_time;
GstClockTime capture_time;
GstClockTime delay;
int64_t srt_time;
SRT_MSGCTRL mctrl;
if (g_cancellable_is_cancelled (self->cancellable)) {
@ -154,10 +155,10 @@ gst_srt_src_fill (GstPushSrc * src, GstBuffer * outbuf)
capture_time = gst_clock_get_time (clock);
#if SRT_VERSION_VALUE >= 0x10402
/* Use SRT clock value if available (SRT > 1.4.2) */
delay = (srt_time_now () - mctrl.srctime) * GST_USECOND;
srt_time = srt_time_now ();
#else
/* Else use the unix epoch monotonic clock */
delay = (g_get_real_time () - mctrl.srctime) * GST_USECOND;
srt_time = g_get_real_time ();
#endif
gst_object_unref (clock);
@ -191,6 +192,12 @@ gst_srt_src_fill (GstPushSrc * src, GstBuffer * outbuf)
/* pktseq is a 31bit field */
self->next_pktseq = (mctrl.pktseq + 1) % G_MAXINT32;
/* 0 means we do not have a srctime */
if (mctrl.srctime != 0)
delay = (srt_time - mctrl.srctime) * GST_USECOND;
else
delay = 0;
/* Subtract the base_time (since the pipeline started) ... */
if (capture_time > base_time)
capture_time -= base_time;