{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_CLEAR_PT_MAP,
+ SIGNAL_RESET_SYNC,
SIGNAL_GET_INTERNAL_SESSION,
SIGNAL_ON_NEW_SSRC,
gulong demux_ptreq_sig;
gulong demux_pt_change_sig;
- /* data for the RTCP sync signal */
+ /* if we have calculated a valid unix_delta for this stream */
gboolean have_sync;
- guint64 last_unix;
- guint64 last_extrtptime;
-
/* mapping to local RTP and NTP time */
- guint64 local_rtp;
- guint64 local_unix;
gint64 unix_delta;
-
- /* for lip-sync */
- guint64 last_clock_base;
- guint64 clock_base;
- guint64 clock_base_time;
- gint clock_rate;
- gint64 ts_offset;
};
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
/* the streams */
guint nstreams;
GSList *streams;
-
- gint64 min_delta;
};
/* find a session with the given id. Must be called with RTP_BIN_LOCK */
result = g_new0 (GstRtpBinClient, 1);
result->cname = g_strndup ((gchar *) data, len);
result->cname_len = len;
- result->min_delta = G_MAXINT64;
bin->clients = g_slist_prepend (bin->clients, result);
GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
result->cname);
g_free (client);
}
+static void
+gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
+{
+ GSList *clients, *streams;
+
+ GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
+
+ GST_RTP_BIN_LOCK (rtpbin);
+ for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
+ GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
+
+ /* reset sync on all streams for this client */
+ for (streams = client->streams; streams; streams = g_slist_next (streams)) {
+ GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
+
+ /* make use require a new SR packet for this stream before we attempt new
+ * lip-sync */
+ stream->have_sync = FALSE;
+ stream->unix_delta = 0;
+ }
+ }
+ GST_RTP_BIN_UNLOCK (rtpbin);
+}
+
/* associate a stream to the given CNAME. This will make sure all streams for
* that CNAME are synchronized together.
* Must be called with GST_RTP_BIN_LOCK */
static void
gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
- guint8 * data)
+ guint8 * data, guint64 last_unix, guint64 last_extrtptime,
+ guint64 clock_base, guint64 clock_base_time, guint clock_rate)
{
GstRtpBinClient *client;
gboolean created;
GSList *walk;
+ guint64 local_unix;
+ guint64 local_rtp;
/* first find or create the CNAME */
client = get_client (bin, len, data, &created);
/* 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;
+ local_rtp = last_extrtptime - clock_base;
GST_DEBUG_OBJECT (bin,
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
- ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", stream->clock_base,
- stream->last_extrtptime, stream->local_rtp, stream->clock_rate);
+ ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", clock_base,
+ last_extrtptime, local_rtp, clock_rate);
/* 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;
+ local_unix = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
+ local_unix += clock_base_time;
+
/* 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;
+ stream->unix_delta = last_unix - local_unix;
+ stream->have_sync = TRUE;
GST_DEBUG_OBJECT (bin,
"local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT
- ", delta %" G_GINT64_FORMAT, stream->local_unix, stream->last_unix,
- stream->unix_delta);
+ ", delta %" G_GINT64_FORMAT, local_unix, last_unix, stream->unix_delta);
/* recalc inter stream playout offset, but only if there is more than one
* stream. */
/* calculate offsets for each stream */
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
- gint64 prev_ts_offset;
+ gint64 ts_offset, 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
/* calculate offset to our reference stream, this should always give a
* positive number. */
- ostream->ts_offset = ostream->unix_delta - min;
+ ts_offset = ostream->unix_delta - min;
g_object_get (ostream->buffer, "ts-offset", &prev_ts_offset, NULL);
/* delta changed, see how much */
- if (prev_ts_offset != ostream->ts_offset) {
+ if (prev_ts_offset != ts_offset) {
gint64 diff;
- if (prev_ts_offset > ostream->ts_offset)
- diff = prev_ts_offset - ostream->ts_offset;
+ if (prev_ts_offset > ts_offset)
+ diff = prev_ts_offset - ts_offset;
else
- diff = ostream->ts_offset - prev_ts_offset;
+ diff = ts_offset - prev_ts_offset;
GST_DEBUG_OBJECT (bin,
"ts-offset %" G_GUINT64_FORMAT ", prev %" G_GUINT64_FORMAT
- ", diff: %" G_GINT64_FORMAT, ostream->ts_offset, prev_ts_offset,
- diff);
+ ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
/* only change diff when it changed more than 4 milliseconds. This
* compensates for rounding errors in NTP to RTP timestamp
* conversions */
if (diff > 4 * GST_MSECOND && diff < (3 * GST_SECOND)) {
- g_object_set (ostream->buffer, "ts-offset", ostream->ts_offset, NULL);
+ g_object_set (ostream->buffer, "ts-offset", ts_offset, NULL);
}
}
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
- ostream->ssrc, ostream->ts_offset);
+ ostream->ssrc, ts_offset);
}
}
return;
if (type == GST_RTCP_SDES_CNAME) {
GST_RTP_BIN_LOCK (bin);
- /* store values in the stream */
- stream->have_sync = TRUE;
- stream->last_unix = gst_rtcp_ntp_to_unix (ntptime);
- stream->last_extrtptime = extrtptime;
- stream->clock_base = clock_base;
- stream->clock_base_time = clock_base_time;
- stream->clock_rate = clock_rate;
/* associate the stream to CNAME */
- gst_rtp_bin_associate (bin, stream, len, data);
+ gst_rtp_bin_associate (bin, stream, len, data,
+ gst_rtcp_ntp_to_unix (ntptime), extrtptime,
+ clock_base, clock_base_time, clock_rate);
GST_RTP_BIN_UNLOCK (bin);
}
}
stream->session = session;
stream->buffer = buffer;
stream->demux = demux;
- stream->last_extrtptime = -1;
- stream->clock_rate = -1;
stream->have_sync = FALSE;
+ stream->unix_delta = 0;
session->streams = g_slist_prepend (session->streams, stream);
/* provide clock_rate to the jitterbuffer when needed */
clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
0, G_TYPE_NONE);
/**
+ * GstRtpBin::reset-sync:
+ * @rtpbin: the object which received the signal
+ *
+ * Reset all currently configured lip-sync parameters and require new SR
+ * packets for all streams before lip-sync is attempted again.
+ */
+ gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
+ g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
+ G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
+ reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
+ 0, G_TYPE_NONE);
+
+ /**
* GstRtpBin::get-internal-session:
* @rtpbin: the object which received the signal
* @id: the session id
gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
+ klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
klass->get_internal_session =
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
GstRtpBinStream *stream;
GstPad *sinkpad, *srcpad;
gchar *padname;
- GstCaps *caps;
rtpbin = session->bin;
if (!stream)
goto no_stream;
- /* get the caps of the pad, we need the clock-rate and base_time if any. */
- if ((caps = gst_pad_get_caps (pad))) {
- const GstStructure *s;
- guint val;
-
- GST_DEBUG_OBJECT (rtpbin, "pad has caps %" GST_PTR_FORMAT, caps);
-
- s = gst_caps_get_structure (caps, 0);
-
- stream->last_clock_base = -1;
- if (gst_structure_get_uint (s, "clock-base", &val))
- stream->clock_base = val;
- else
- stream->clock_base = -1;
-
- gst_caps_unref (caps);
- }
-
/* get pad and link */
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
padname = g_strdup_printf ("src_%d", ssrc);
projects / platform / upstream / gstreamer.git / commitdiff