mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 04:01:08 +00:00
gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji...
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found): Do not try to keep track of the clock-rate ourselves but simply get the value from the jitterbuffer. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add some debug info. Pass the clock-rate to the jitterbuffer. Also pass the clock-rate along with the rtp timestamp when getting the sync parameters. * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain): Fix some debug. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Keep track of clock-rate changes and return the clock-rate together with the rtp timestamps used for sync. Don't try to construct timestamps when we have no base_time. * gst/rtpmanager/rtpsource.c: (get_clock_rate): Request a new clock-rate when the payload type changes. Reset the jitter calculation when the clock-rate changes.
This commit is contained in:
parent
1656fad93e
commit
5ab3e10594
7 changed files with 48 additions and 62 deletions
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@ -327,7 +327,6 @@ struct _GstRtpBinStream
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guint64 clock_base_time;
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gint clock_rate;
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gint64 ts_offset;
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gint last_pt;
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};
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#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
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@ -856,29 +855,8 @@ gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
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if (stream->clock_base == -1)
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goto no_clock_base;
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if (stream->clock_rate <= 0) {
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gint pt = -1;
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GstCaps *caps = NULL;
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GstStructure *s = NULL;
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GST_RTP_SESSION_LOCK (stream->session);
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pt = stream->last_pt;
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GST_RTP_SESSION_UNLOCK (stream->session);
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if (pt < 0)
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goto no_clock_rate;
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caps = get_pt_map (stream->session, pt);
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if (!caps)
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goto no_clock_rate;
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s = gst_caps_get_structure (caps, 0);
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gst_structure_get_int (s, "clock-rate", &stream->clock_rate);
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gst_caps_unref (caps);
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if (stream->clock_rate <= 0)
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goto no_clock_rate;
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}
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if (stream->clock_rate <= 0)
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goto no_clock_rate;
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/* take the extended rtptime we found in the SR packet and map it to the
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* local rtptime. The local rtp time is used to construct timestamps on the
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@ -1021,6 +999,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
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gboolean more;
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guint64 clock_base;
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guint64 clock_base_time;
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guint clock_rate;
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stream = gst_pad_get_element_private (pad);
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bin = stream->bin;
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@ -1035,7 +1014,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
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* constructs gstreamer timestamps from rtp timestamps and so it know exactly
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* what the current situation is. */
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gst_rtp_jitter_buffer_get_sync (GST_RTP_JITTER_BUFFER (stream->buffer),
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&clock_base, &clock_base_time);
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&clock_base, &clock_base_time, &clock_rate);
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/* clock base changes when there is a huge gap in the timestamps or seqnum.
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* When this happens we don't want to calculate the extended timestamp based
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@ -1098,6 +1077,7 @@ gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
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if (type == GST_RTCP_SDES_CNAME) {
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stream->clock_base = clock_base;
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stream->clock_base_time = clock_base_time;
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stream->clock_rate = clock_rate;
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/* associate the stream to CNAME */
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gst_rtp_bin_associate (bin, stream, len, data);
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}
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@ -1150,7 +1130,7 @@ create_stream (GstRtpBinSession * session, guint32 ssrc)
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stream->buffer = buffer;
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stream->demux = demux;
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stream->last_extrtptime = -1;
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stream->last_pt = -1;
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stream->clock_rate = -1;
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stream->have_sync = FALSE;
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session->streams = g_slist_prepend (session->streams, stream);
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@ -1961,15 +1941,6 @@ caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
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GST_RTP_SESSION_UNLOCK (session);
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}
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/* Stores the last payload type received on a particular stream */
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static void
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payload_type_change (GstElement * element, guint pt, GstRtpBinStream * stream)
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{
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GST_RTP_SESSION_LOCK (stream->session);
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stream->last_pt = pt;
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GST_RTP_SESSION_UNLOCK (stream->session);
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}
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/* a new pad (SSRC) was created in @session */
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static void
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new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
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@ -2004,14 +1975,6 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
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s = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (s, "clock-rate", &stream->clock_rate)) {
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stream->clock_rate = -1;
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GST_WARNING_OBJECT (rtpbin,
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"Caps have no clock rate %s from pad %s:%s",
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gst_caps_to_string (caps), GST_DEBUG_PAD_NAME (pad));
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}
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stream->last_clock_base = -1;
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if (gst_structure_get_uint (s, "clock-base", &val))
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stream->clock_base = val;
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@ -2048,11 +2011,6 @@ new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
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* depayloaders. */
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stream->demux_ptreq_sig = g_signal_connect (stream->demux,
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"request-pt-map", (GCallback) pt_map_requested, session);
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/* connect to the payload-type-change signal so that we can know which
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* PT is the current PT so that the jitterbuffer can be matched to the right
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* offset. */
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stream->demux_pt_change_sig = g_signal_connect (stream->demux,
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"payload-type-change", (GCallback) payload_type_change, stream);
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GST_RTP_SESSION_UNLOCK (session);
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GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
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@ -851,6 +851,9 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
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if (G_UNLIKELY (priv->last_pt != pt)) {
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GstCaps *caps;
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GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
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pt);
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priv->last_pt = pt;
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/* reset clock-rate so that we get a new one */
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priv->clock_rate = -1;
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@ -865,7 +868,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
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/* no clock rate given on the caps, try to get one with the signal */
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gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
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if (G_UNLIKELY (priv->clock_rate == -1))
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goto not_negotiated;
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goto no_clock_rate;
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}
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/* take the timestamp of the buffer. This is the time when the packet was
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@ -997,9 +1000,10 @@ invalid_buffer:
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gst_object_unref (jitterbuffer);
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return GST_FLOW_OK;
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}
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not_negotiated:
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no_clock_rate:
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{
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GST_WARNING_OBJECT (jitterbuffer, "No clock-rate in caps!");
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GST_WARNING_OBJECT (jitterbuffer,
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"No clock-rate in caps!, dropping buffer");
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gst_buffer_unref (buffer);
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gst_object_unref (jitterbuffer);
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return GST_FLOW_OK;
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@ -1484,7 +1488,7 @@ gst_rtp_jitter_buffer_get_property (GObject * object,
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void
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gst_rtp_jitter_buffer_get_sync (GstRtpJitterBuffer * buffer, guint64 * rtptime,
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guint64 * timestamp)
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guint64 * timestamp, guint32 * clock_rate)
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{
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GstRtpJitterBufferPrivate *priv;
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@ -1493,6 +1497,6 @@ gst_rtp_jitter_buffer_get_sync (GstRtpJitterBuffer * buffer, guint64 * rtptime,
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priv = buffer->priv;
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JBUF_LOCK (priv);
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rtp_jitter_buffer_get_sync (priv->jbuf, rtptime, timestamp);
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rtp_jitter_buffer_get_sync (priv->jbuf, rtptime, timestamp, clock_rate);
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JBUF_UNLOCK (priv);
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}
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@ -80,7 +80,8 @@ struct _GstRtpJitterBufferClass
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GType gst_rtp_jitter_buffer_get_type (void);
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void gst_rtp_jitter_buffer_get_sync (GstRtpJitterBuffer *buffer,
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guint64 *rtptime, guint64 *timestamp);
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guint64 *rtptime, guint64 *timestamp,
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guint32 *clock_rate);
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G_END_DECLS
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@ -353,7 +353,7 @@ gst_rtp_pt_demux_chain (GstPad * pad, GstBuffer * buf)
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gst_pad_set_active (srcpad, TRUE);
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gst_element_add_pad (element, srcpad);
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GST_DEBUG ("emitting new-payload_type for pt %d", pt);
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GST_DEBUG ("emitting new-payload-type for pt %d", pt);
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g_signal_emit (G_OBJECT (rtpdemux),
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gst_rtp_pt_demux_signals[SIGNAL_NEW_PAYLOAD_TYPE], 0, pt, srcpad);
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}
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@ -105,6 +105,7 @@ rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
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jbuf->base_time = -1;
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jbuf->base_rtptime = -1;
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jbuf->base_extrtp = -1;
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jbuf->clock_rate = -1;
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jbuf->ext_rtptime = -1;
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jbuf->window_pos = 0;
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jbuf->window_filling = TRUE;
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@ -187,6 +188,13 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
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gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
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if (jbuf->clock_rate != -1 && jbuf->clock_rate != clock_rate) {
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GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
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G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
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jbuf->base_time = -1;
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jbuf->base_rtptime = -1;
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}
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/* first time, lock on to time and gstrtptime */
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if (G_UNLIKELY (jbuf->base_time == -1)) {
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jbuf->base_time = time;
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}
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if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
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jbuf->base_rtptime = gstrtptime;
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jbuf->clock_rate = clock_rate;
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jbuf->base_extrtp = ext_rtptime;
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GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
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GST_TIME_ARGS (gstrtptime));
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GST_DEBUG ("backward timestamps at server, taking new base time");
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jbuf->base_time = time;
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jbuf->base_rtptime = gstrtptime;
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jbuf->clock_rate = clock_rate;
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jbuf->base_extrtp = ext_rtptime;
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send_diff = 0;
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}
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@ -218,7 +228,7 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
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/* we don't have an arrival timestamp so we can't do skew detection. we
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* should still apply a timestamp based on RTP timestamp and base_time */
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if (time == -1)
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if (time == -1 || jbuf->base_time == -1)
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goto no_skew;
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/* elapsed time at receiver, includes the jitter */
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@ -240,6 +250,7 @@ calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
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jbuf->base_time = time;
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jbuf->base_rtptime = gstrtptime;
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jbuf->base_extrtp = ext_rtptime;
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jbuf->clock_rate = clock_rate;
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send_diff = 0;
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delta = 0;
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}
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no_skew:
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/* the output time is defined as the base timestamp plus the RTP time
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* adjusted for the clock skew .*/
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out_time = jbuf->base_time + send_diff + jbuf->skew;
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if (jbuf->base_time != -1)
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out_time = jbuf->base_time + send_diff + jbuf->skew;
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else
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out_time = -1;
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GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
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jbuf->skew, GST_TIME_ARGS (out_time));
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* @jbuf: an #RTPJitterBuffer
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* @rtptime: result RTP time
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* @timestamp: result GStreamer timestamp
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* @clock_rate: clock-rate of @rtptime
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*
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* Returns the relation between the RTP timestamp and the GStreamer timestamp
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* used for constructing timestamps.
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*/
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void
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rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
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guint64 * timestamp)
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guint64 * timestamp, guint32 * clock_rate)
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{
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if (rtptime)
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*rtptime = jbuf->base_extrtp;
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if (timestamp)
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*timestamp = jbuf->base_time + jbuf->skew;
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if (clock_rate)
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*clock_rate = jbuf->clock_rate;
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}
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@ -56,6 +56,7 @@ struct _RTPJitterBuffer {
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/* for calculating skew */
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GstClockTime base_time;
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GstClockTime base_rtptime;
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guint32 clock_rate;
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GstClockTime base_extrtp;
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guint64 ext_rtptime;
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gint64 window[RTP_JITTER_BUFFER_MAX_WINDOW];
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@ -91,7 +92,7 @@ guint rtp_jitter_buffer_num_packets (RTPJitterBuffer *jbuf)
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guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer *jbuf);
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void rtp_jitter_buffer_get_sync (RTPJitterBuffer *jbuf, guint64 *rtptime,
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guint64 *timestamp);
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guint64 *timestamp, guint32 *clock_rate);
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#endif /* __RTP_JITTER_BUFFER_H__ */
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@ -721,18 +721,23 @@ push_packet (RTPSource * src, GstBuffer * buffer)
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static gint
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get_clock_rate (RTPSource * src, guint8 payload)
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{
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if (payload != src->payload) {
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GST_DEBUG ("new payload %d", payload);
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src->payload = payload;
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src->clock_rate = -1;
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src->stats.transit = -1;
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}
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if (src->clock_rate == -1) {
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gint clock_rate = -1;
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if (src->callbacks.clock_rate)
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clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
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GST_DEBUG ("new payload %d, got clock-rate %d", payload, clock_rate);
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GST_DEBUG ("got clock-rate %d", payload, clock_rate);
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src->clock_rate = clock_rate;
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}
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src->payload = payload;
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return src->clock_rate;
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}
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