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gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...

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Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
This commit is contained in:
Wim Taymans 2008-09-13 01:37:50 +00:00 committed by Tim-Philipp Müller
parent 85e26f6546
commit 5e98fa572f
1 changed files with 34 additions and 7 deletions
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41
gst/rtpmanager/gstrtpbin.c
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@ -832,7 +832,9 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
goto no_clock_rate;
}
/* map last RTP time to local timeline using our clock-base */
/* take the extended rtptime we found in the SR packet and map it to the
* local rtptime. The local rtp time is used to construct timestamps on the
* buffers. */
stream->local_rtp = stream->last_extrtptime - stream->clock_base;
GST_DEBUG_OBJECT (bin,
@ -840,12 +842,16 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", stream->clock_base,
stream->last_extrtptime, stream->local_rtp, stream->clock_rate);
/* calculate local NTP time in gstreamer timestamp */
/* calculate local NTP time in gstreamer timestamp, we essentially perform the
* same conversion that a jitterbuffer would use to convert an rtp timestamp
* into a corresponding gstreamer timestamp. */
stream->local_unix =
gst_util_uint64_scale_int (stream->local_rtp, GST_SECOND,
stream->clock_rate);
stream->local_unix += stream->clock_base_time;
/* calculate delta between server and receiver */
/* calculate delta between server and receiver. last_unix is created by
* converting the ntptime in the last SR packet to a gstreamer timestamp. This
* delta expresses the difference to our timeline and the server timeline. */
stream->unix_delta = stream->last_unix - stream->local_unix;
GST_DEBUG_OBJECT (bin,
@ -853,17 +859,28 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
", delta %" G_GINT64_FORMAT, stream->local_unix, stream->last_unix,
stream->unix_delta);
/* recalc inter stream playout offset, but only if there are more than one
/* recalc inter stream playout offset, but only if there is more than one
* stream. */
if (client->nstreams > 1) {
gint64 min;
/* calculate the min of all deltas */
/* calculate the min of all deltas, ignoring streams that did not yet have a
* valid unix_delta because we did not yet receive an SR packet for those
* streams.
* We calculate the mininum because we would like to only apply positive
* offsets to streams, delaying their playback instead of trying to speed up
* other streams (which might be imposible when we have to create negative
* latencies).
* The stream that has the smalest diff is selected as the reference stream,
* all other streams will have a positive offset to this difference. */
min = G_MAXINT64;
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
if (ostream->unix_delta && ostream->unix_delta < min)
if (!ostream->have_sync)
continue;
if (ostream->unix_delta < min)
min = ostream->unix_delta;
}
@ -875,6 +892,14 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
gint64 prev_ts_offset;
/* ignore streams for which we didn't receive an SR packet yet, we
* can't synchronize them yet. We can however sync other streams just
* fine. */
if (!ostream->have_sync)
continue;
/* calculate offset to our reference stream, this should always give a
* positive number. */
ostream->ts_offset = ostream->unix_delta - min;
g_object_get (ostream->buffer, "ts-offset", &prev_ts_offset, NULL);
@ -905,6 +930,7 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
}
}
GST_RTP_BIN_UNLOCK (bin);
return;
no_clock_base:
@ -958,7 +984,8 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
/* get the last relation between the rtp timestamps and the gstreamer
* timestamps. We get this info directly from the jitterbuffer which
* constructs gstreamer timestamps from rtp timestamps */
* constructs gstreamer timestamps from rtp timestamps and so it know exactly
* what the current situation is. */
gst_rtp_jitter_buffer_get_sync (GST_RTP_JITTER_BUFFER (stream->buffer),
&clock_base, &clock_base_time);