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rtpbin: Clean up stream association state

Browse source 
Use fewer magic numbers and keep track of the different synchronization
mechanisms separately. Also keep track of more state to detect more
situations when resynchronization should happen.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
This commit is contained in:
Sebastian Dröge 2024-04-17 17:01:21 +03:00 committed by GStreamer Marge Bot
parent d8dabf142f
commit 3eb22af88b
1 changed files with 62 additions and 25 deletions
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87
subprojects/gst-plugins-good/gst/rtpmanager/gstrtpbin.c
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@ -443,6 +443,13 @@ static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
static GstElement *session_request_element (GstRtpBinSession * session,
guint signal);
typedef enum
{
GST_RTP_BIN_STREAM_SYNCED_NONE,
GST_RTP_BIN_STREAM_SYNCED_RTCP,
GST_RTP_BIN_STREAM_SYNCED_RTP_INFO
} GstRtpBinStreamSynced;
/* Manages the RTP stream for one SSRC.
*
* We pipe the stream (coming from the SSRC demuxer) into a jitterbuffer.
@ -475,15 +482,23 @@ struct _GstRtpBinStream
gulong demux_ptreq_sig;
gulong demux_ptchange_sig;
/* if we have calculated a valid rt_delta for this stream */
gboolean have_sync;
/* if we have synced this stream */
GstRtpBinStreamSynced have_sync;
/* mapping to local RTP and NTP time */
gint64 rt_delta;
gint64 rtp_delta;
gint64 rt_delta; /* based on RTCP */
gint64 rtp_delta; /* based on RTP-Info */
/* base RTP time */
gint64 base_rtptime;
guint64 base_time;
/* clock-base for RTP-Info */
guint64 clock_base;
/* Smoothing of ts-offset adjustments */
gint64 avg_ts_offset;
gboolean is_initialized;
/* base rtptime in gst time */
gint64 clock_base;
};
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
@ -1048,12 +1063,16 @@ gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
/* make use require a new SR packet for this stream before we attempt new
* lip-sync */
stream->have_sync = FALSE;
stream->rt_delta = 0;
stream->have_sync = GST_RTP_BIN_STREAM_SYNCED_NONE;
stream->rt_delta = G_MININT64;
stream->rtp_delta = G_MININT64;
stream->base_rtptime = -1;
stream->base_time = GST_CLOCK_TIME_NONE;
stream->clock_base = -1;
stream->avg_ts_offset = 0;
stream->is_initialized = FALSE;
stream->rtp_delta = 0;
stream->clock_base = -100 * GST_SECOND;
}
}
GST_RTP_BIN_UNLOCK (rtpbin);
@ -1616,8 +1635,6 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
gboolean all_sync = TRUE;
gint64 min = G_MAXINT64, rtp_min = G_MAXINT64;
stream->have_sync = TRUE;
/* recalc inter stream playout offset based on the selected mode */
if (rtp_info_sync) {
guint64 ext_base = -1;
@ -1652,7 +1669,7 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
/* if not set yet for the other stream then ignore it */
if (!ostream->have_sync) {
if (ostream->rtp_delta == G_MININT64) {
all_sync = FALSE;
continue;
}
@ -1662,8 +1679,11 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
if (stream != ostream && stream->clock_base >= 0 &&
(stream->clock_base != rtp_clock_base)) {
GST_DEBUG_OBJECT (bin, "reset upon clock base change");
ostream->clock_base = -100 * GST_SECOND;
ostream->rtp_delta = 0;
ostream->clock_base = -1;
ostream->rtp_delta = G_MININT64;
if (ostream->have_sync == GST_RTP_BIN_STREAM_SYNCED_RTP_INFO)
ostream->have_sync = GST_RTP_BIN_STREAM_SYNCED_NONE;
continue;
}
if (ostream->rtp_delta < rtp_min)
@ -1711,7 +1731,7 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
/* if not set yet for the other stream then ignore it */
if (!ostream->have_sync) {
if (ostream->rt_delta == G_MININT64) {
all_sync = FALSE;
continue;
}
@ -1739,8 +1759,15 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
/* arrange to re-sync for each stream upon significant change, e.g. post-seek
* or when the RTP base time changes because of a jitterbuffer reset */
all_sync = all_sync && stream->clock_base == rtp_clock_base;
stream->clock_base = rtp_clock_base;
all_sync = all_sync && stream->base_rtptime == base_rtptime &&
((rtp_info_sync && stream->clock_base == rtp_clock_base)
|| (!rtp_info_sync && stream->base_time == base_time));
stream->base_rtptime = base_rtptime;
if (rtp_info_sync) {
stream->clock_base = rtp_clock_base;
} else {
stream->base_time = base_time;
}
/* may need init performed above later on, but nothing more to do now */
if (client->nstreams <= 1)
@ -1779,8 +1806,10 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
/* 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)
if ((rtp_info_sync && ostream->rtp_delta == G_MININT64) ||
(!rtp_info_sync && ostream->rt_delta == G_MININT64)) {
continue;
}
/* calculate offset to our reference stream, this should always give a
* positive number. */
@ -1795,11 +1824,14 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream,
stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
bin->min_ts_offset, TRUE);
if (rtp_info_sync)
ostream->have_sync = GST_RTP_BIN_STREAM_SYNCED_RTP_INFO;
else
ostream->have_sync = GST_RTP_BIN_STREAM_SYNCED_RTCP;;
}
gst_rtp_bin_send_sync_event (stream);
return;
}
#define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
@ -2018,13 +2050,18 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
stream->buffer = gst_object_ref (buffer);
stream->demux = demux;
stream->have_sync = FALSE;
stream->rt_delta = 0;
stream->have_sync = GST_RTP_BIN_STREAM_SYNCED_NONE;
stream->rt_delta = G_MININT64;
stream->rtp_delta = G_MININT64;
stream->base_rtptime = -1;
stream->base_time = GST_CLOCK_TIME_NONE;
stream->clock_base = -1;
stream->avg_ts_offset = 0;
stream->is_initialized = FALSE;
stream->rtp_delta = 0;
stream->percent = 100;
stream->clock_base = -100 * GST_SECOND;
session->streams = g_slist_prepend (session->streams, stream);
jb_class = G_OBJECT_GET_CLASS (G_OBJECT (buffer));