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author | Wim Taymans <wim.taymans@gmail.com> | 2008-09-13 01:37:50 +0000 |
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committer | Wim Taymans <wim.taymans@gmail.com> | 2008-09-13 01:37:50 +0000 |
commit | c1647d0369a523469dbd5d75de30b6f61319b8f0 (patch) | |
tree | 682910f0ad31bf4580842f7c0b6c30729b419e13 /gst/rtpmanager | |
parent | 007478f09c8c10a24b473d6ac8eef1929ce25ca0 (diff) | |
download | gst-plugins-bad-c1647d0369a523469dbd5d75de30b6f61319b8f0.tar.gz gst-plugins-bad-c1647d0369a523469dbd5d75de30b6f61319b8f0.tar.bz2 gst-plugins-bad-c1647d0369a523469dbd5d75de30b6f61319b8f0.zip |
gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...
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.
Diffstat (limited to 'gst/rtpmanager')
-rw-r--r-- | gst/rtpmanager/gstrtpbin.c | 41 |
1 files changed, 34 insertions, 7 deletions
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 7f402c36..bab52384 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -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); |