mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 10:11:08 +00:00
gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property), (gst_rtp_bin_get_property): Use lock to protect variable. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_class_init), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain), (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop): Reconstruct GST timestamp from RTP timestamps based on measured clock skew and sync offset. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init), (rtp_jitter_buffer_set_tail_changed), (rtp_jitter_buffer_set_clock_rate), (rtp_jitter_buffer_get_clock_rate), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek): * gst/rtpmanager/rtpjitterbuffer.h: Measure clock skew. Add callback to be notfied when a new packet was inserted at the tail. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (calculate_jitter), (rtp_source_send_rtp): * gst/rtpmanager/rtpsource.h: Remove clock skew detection, it's move to the jitterbuffer now.
This commit is contained in:
parent
0441ef80b0
commit
b2aa36cb0d
6 changed files with 341 additions and 139 deletions
|
@ -1183,7 +1183,9 @@ gst_rtp_bin_set_property (GObject * object, guint prop_id,
|
|||
|
||||
switch (prop_id) {
|
||||
case PROP_LATENCY:
|
||||
GST_RTP_BIN_LOCK (rtpbin);
|
||||
rtpbin->latency = g_value_get_uint (value);
|
||||
GST_RTP_BIN_UNLOCK (rtpbin);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
|
@ -1201,7 +1203,9 @@ gst_rtp_bin_get_property (GObject * object, guint prop_id,
|
|||
|
||||
switch (prop_id) {
|
||||
case PROP_LATENCY:
|
||||
GST_RTP_BIN_LOCK (rtpbin);
|
||||
g_value_set_uint (value, rtpbin->latency);
|
||||
GST_RTP_BIN_UNLOCK (rtpbin);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
|
|
|
@ -320,7 +320,7 @@ gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
|
|||
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
|
||||
|
||||
GST_DEBUG_CATEGORY_INIT
|
||||
(rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
|
||||
(rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
|
||||
}
|
||||
|
||||
static void
|
||||
|
@ -453,6 +453,8 @@ gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
|
|||
if (priv->clock_rate <= 0)
|
||||
goto wrong_rate;
|
||||
|
||||
rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
|
||||
|
||||
GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
|
||||
|
||||
/* gah, clock-base is uint. If we don't have a base, we will use the first
|
||||
|
@ -794,6 +796,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
|
|||
GstRtpJitterBufferPrivate *priv;
|
||||
guint16 seqnum;
|
||||
GstFlowReturn ret = GST_FLOW_OK;
|
||||
GstClockTime timestamp;
|
||||
|
||||
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
|
||||
|
||||
|
@ -811,10 +814,23 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
|
|||
gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
|
||||
if (priv->clock_rate == -1)
|
||||
goto not_negotiated;
|
||||
|
||||
rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
|
||||
}
|
||||
|
||||
/* take the timestamp of the buffer. This is the time when the packet was
|
||||
* received and is used to calculate jitter and clock skew. We will adjust
|
||||
* this timestamp with the smoothed value after processing it in the
|
||||
* jitterbuffer. */
|
||||
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
||||
/* bring to running time */
|
||||
timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
|
||||
timestamp);
|
||||
|
||||
seqnum = gst_rtp_buffer_get_seq (buffer);
|
||||
GST_DEBUG_OBJECT (jitterbuffer, "Received packet #%d", seqnum);
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
|
||||
GST_TIME_ARGS (timestamp));
|
||||
|
||||
JBUF_LOCK_CHECK (priv, out_flushing);
|
||||
/* don't accept more data on EOS */
|
||||
|
@ -852,7 +868,7 @@ gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
|
|||
/* now insert the packet into the queue in sorted order. This function returns
|
||||
* FALSE if a packet with the same seqnum was already in the queue, meaning we
|
||||
* have a duplicate. */
|
||||
if (!rtp_jitter_buffer_insert (priv->jbuf, buffer))
|
||||
if (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp))
|
||||
goto duplicate;
|
||||
|
||||
/* signal addition of new buffer */
|
||||
|
@ -926,6 +942,37 @@ duplicate:
|
|||
}
|
||||
}
|
||||
|
||||
static GstClockTime
|
||||
convert_rtptime_to_gsttime (GstRtpJitterBuffer * jitterbuffer,
|
||||
guint64 exttimestamp)
|
||||
{
|
||||
GstClockTime timestamp;
|
||||
GstRtpJitterBufferPrivate *priv;
|
||||
|
||||
priv = jitterbuffer->priv;
|
||||
|
||||
/* construct a timestamp from the RTP timestamp now. We don't apply this
|
||||
* timestamp to the outgoing buffer yet as the popped buffer might not be the
|
||||
* one we need to push out right now. */
|
||||
timestamp =
|
||||
gst_util_uint64_scale_int (exttimestamp, GST_SECOND, priv->clock_rate);
|
||||
|
||||
/* apply first observed timestamp */
|
||||
timestamp += priv->jbuf->base_time;
|
||||
|
||||
/* apply the current clock skew */
|
||||
timestamp += priv->jbuf->skew;
|
||||
|
||||
/* apply the timestamp offset */
|
||||
timestamp += priv->ts_offset;
|
||||
|
||||
/* add latency, this includes our own latency and the peer latency. */
|
||||
timestamp += (priv->latency_ms * GST_MSECOND);
|
||||
timestamp += priv->peer_latency;
|
||||
|
||||
return timestamp;
|
||||
}
|
||||
|
||||
/**
|
||||
* This funcion will push out buffers on the source pad.
|
||||
*
|
||||
|
@ -942,9 +989,7 @@ gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
|
|||
guint16 seqnum;
|
||||
guint32 rtp_time;
|
||||
GstClockTime timestamp;
|
||||
gint64 running_time;
|
||||
guint64 exttimestamp;
|
||||
gint ts_offset_rtp;
|
||||
|
||||
priv = jitterbuffer->priv;
|
||||
|
||||
|
@ -968,19 +1013,29 @@ again:
|
|||
|
||||
/* pop a buffer, we must have a buffer now */
|
||||
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
|
||||
|
||||
seqnum = gst_rtp_buffer_get_seq (outbuf);
|
||||
|
||||
/* get the max deadline to wait for the missing packets, this is the time
|
||||
* of the currently popped packet */
|
||||
/* construct extended RTP timestamp from packet */
|
||||
rtp_time = gst_rtp_buffer_get_timestamp (outbuf);
|
||||
exttimestamp = gst_rtp_buffer_ext_timestamp (&priv->exttimestamp, rtp_time);
|
||||
|
||||
/* if no clock_base was given, take first ts as base */
|
||||
if (priv->clock_base == -1) {
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"no clock base, using exttimestamp %" G_GUINT64_FORMAT, exttimestamp);
|
||||
priv->clock_base = exttimestamp;
|
||||
}
|
||||
/* subtract the base clock time so that we start counting from 0 */
|
||||
exttimestamp -= priv->clock_base;
|
||||
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"Popped buffer #%d, rtptime %u, exttime %" G_GUINT64_FORMAT
|
||||
", now %d left", seqnum, rtp_time, exttimestamp,
|
||||
rtp_jitter_buffer_num_packets (priv->jbuf));
|
||||
|
||||
/* convert the RTP timestamp to a gstreamer timestamp. */
|
||||
timestamp = convert_rtptime_to_gsttime (jitterbuffer, exttimestamp);
|
||||
|
||||
/* If we don't know what the next seqnum should be (== -1) we have to wait
|
||||
* because it might be possible that we are not receiving this buffer in-order,
|
||||
* a buffer with a lower seqnum could arrive later and we want to push that
|
||||
|
@ -991,7 +1046,7 @@ again:
|
|||
* packet expires. */
|
||||
if (priv->next_seqnum == -1 || priv->next_seqnum != seqnum) {
|
||||
GstClockID id;
|
||||
GstClockTimeDiff jitter;
|
||||
GstClockTime sync_time;
|
||||
GstClockReturn ret;
|
||||
GstClock *clock;
|
||||
|
||||
|
@ -1007,34 +1062,6 @@ again:
|
|||
GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
|
||||
}
|
||||
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"exttimestamp %" G_GUINT64_FORMAT ", base %" G_GINT64_FORMAT,
|
||||
exttimestamp, priv->clock_base);
|
||||
|
||||
/* if no clock_base was given, take first ts as base */
|
||||
if (priv->clock_base == -1) {
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"no clock base, using exttimestamp %" G_GUINT64_FORMAT, exttimestamp);
|
||||
priv->clock_base = exttimestamp;
|
||||
}
|
||||
|
||||
/* take rtp timestamp offset into account, this should not wrap around since
|
||||
* we are dealing with the extended timestamp here. */
|
||||
exttimestamp -= priv->clock_base;
|
||||
|
||||
/* bring timestamp to gst time */
|
||||
timestamp =
|
||||
gst_util_uint64_scale_int (exttimestamp, GST_SECOND, priv->clock_rate);
|
||||
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"exttimestamp %" G_GUINT64_FORMAT ", clock-rate %u, timestamp %"
|
||||
GST_TIME_FORMAT, exttimestamp, priv->clock_rate,
|
||||
GST_TIME_ARGS (timestamp));
|
||||
|
||||
/* bring to running time */
|
||||
running_time = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
|
||||
timestamp);
|
||||
|
||||
GST_OBJECT_LOCK (jitterbuffer);
|
||||
clock = GST_ELEMENT_CLOCK (jitterbuffer);
|
||||
if (!clock) {
|
||||
|
@ -1043,25 +1070,21 @@ again:
|
|||
goto push_buffer;
|
||||
}
|
||||
|
||||
/* add latency, this includes our own latency and the peer latency. */
|
||||
running_time += (priv->latency_ms * GST_MSECOND);
|
||||
running_time += priv->peer_latency;
|
||||
|
||||
GST_DEBUG_OBJECT (jitterbuffer, "sync to running_time %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (running_time));
|
||||
GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (timestamp));
|
||||
|
||||
/* prepare for sync against clock */
|
||||
running_time += GST_ELEMENT_CAST (jitterbuffer)->base_time;
|
||||
sync_time = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
|
||||
|
||||
/* create an entry for the clock */
|
||||
id = priv->clock_id = gst_clock_new_single_shot_id (clock, running_time);
|
||||
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
|
||||
priv->waiting_seqnum = seqnum;
|
||||
GST_OBJECT_UNLOCK (jitterbuffer);
|
||||
|
||||
/* release the lock so that the other end can push stuff or unlock */
|
||||
JBUF_UNLOCK (priv);
|
||||
|
||||
ret = gst_clock_id_wait (id, &jitter);
|
||||
ret = gst_clock_id_wait (id, NULL);
|
||||
|
||||
JBUF_LOCK (priv);
|
||||
/* and free the entry */
|
||||
|
@ -1080,8 +1103,9 @@ again:
|
|||
if (ret == GST_CLOCK_UNSCHEDULED) {
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"Wait got unscheduled, will retry to push with new buffer");
|
||||
/* reinsert popped buffer into queue */
|
||||
if (!rtp_jitter_buffer_insert (priv->jbuf, outbuf)) {
|
||||
/* reinsert popped buffer into queue, no need to recalculate skew, we do
|
||||
* that when inserting the buffer in the chain function */
|
||||
if (!rtp_jitter_buffer_insert (priv->jbuf, outbuf, -1)) {
|
||||
GST_DEBUG_OBJECT (jitterbuffer,
|
||||
"Duplicate packet #%d detected, dropping", seqnum);
|
||||
priv->num_duplicates++;
|
||||
|
@ -1089,6 +1113,9 @@ again:
|
|||
}
|
||||
goto again;
|
||||
}
|
||||
/* After waiting, we might have a better estimate of skew, generate a new
|
||||
* timestamp before pushing out the buffer */
|
||||
timestamp = convert_rtptime_to_gsttime (jitterbuffer, exttimestamp);
|
||||
}
|
||||
push_buffer:
|
||||
/* check if we are pushing something unexpected */
|
||||
|
@ -1105,37 +1132,13 @@ push_buffer:
|
|||
/* update stats */
|
||||
priv->num_late += dropped;
|
||||
|
||||
/* set DISCONT flag */
|
||||
/* set DISCONT flag when we missed a packet. */
|
||||
outbuf = gst_buffer_make_metadata_writable (outbuf);
|
||||
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
||||
}
|
||||
|
||||
/* apply the timestamp offset */
|
||||
if (priv->ts_offset > 0)
|
||||
ts_offset_rtp =
|
||||
gst_util_uint64_scale_int (priv->ts_offset, priv->clock_rate,
|
||||
GST_SECOND);
|
||||
else if (priv->ts_offset < 0)
|
||||
ts_offset_rtp =
|
||||
-gst_util_uint64_scale_int (-priv->ts_offset, priv->clock_rate,
|
||||
GST_SECOND);
|
||||
else
|
||||
ts_offset_rtp = 0;
|
||||
|
||||
if (ts_offset_rtp != 0) {
|
||||
guint32 timestamp;
|
||||
|
||||
/* if the offset changed, mark with discont */
|
||||
if (priv->ts_offset != priv->prev_ts_offset) {
|
||||
GST_DEBUG_OBJECT (jitterbuffer, "changing offset to %d", ts_offset_rtp);
|
||||
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
|
||||
priv->prev_ts_offset = priv->ts_offset;
|
||||
}
|
||||
|
||||
timestamp = gst_rtp_buffer_get_timestamp (outbuf);
|
||||
timestamp += ts_offset_rtp;
|
||||
gst_rtp_buffer_set_timestamp (outbuf, timestamp);
|
||||
}
|
||||
/* apply timestamp to buffer now */
|
||||
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
|
||||
|
||||
/* now we are ready to push the buffer. Save the seqnum and release the lock
|
||||
* so the other end can push stuff in the queue again. */
|
||||
|
|
|
@ -61,7 +61,20 @@ rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
|
|||
static void
|
||||
rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
|
||||
{
|
||||
gint i;
|
||||
|
||||
jbuf->packets = g_queue_new ();
|
||||
jbuf->base_time = -1;
|
||||
jbuf->base_rtptime = -1;
|
||||
jbuf->ext_rtptime = -1;
|
||||
|
||||
for (i = 0; i < 100; i++) {
|
||||
jbuf->window[i] = 0;
|
||||
}
|
||||
jbuf->window_pos = 0;
|
||||
jbuf->window_filling = TRUE;
|
||||
jbuf->window_min = 0;
|
||||
jbuf->skew = 0;
|
||||
}
|
||||
|
||||
static void
|
||||
|
@ -94,6 +107,168 @@ rtp_jitter_buffer_new (void)
|
|||
return jbuf;
|
||||
}
|
||||
|
||||
void
|
||||
rtp_jitter_buffer_set_tail_changed (RTPJitterBuffer * jbuf, RTPTailChanged func,
|
||||
gpointer user_data)
|
||||
{
|
||||
g_return_if_fail (jbuf != NULL);
|
||||
|
||||
jbuf->tail_changed = func;
|
||||
jbuf->user_data = user_data;
|
||||
}
|
||||
|
||||
void
|
||||
rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, gint clock_rate)
|
||||
{
|
||||
g_return_if_fail (jbuf != NULL);
|
||||
|
||||
jbuf->clock_rate = clock_rate;
|
||||
}
|
||||
|
||||
gint
|
||||
rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
|
||||
{
|
||||
g_return_val_if_fail (jbuf != NULL, 0);
|
||||
|
||||
return jbuf->clock_rate;
|
||||
}
|
||||
|
||||
|
||||
/* For the clock skew we use a windowed low point averaging algorithm as can be
|
||||
* found in http://www.grame.fr/pub/TR-050601.pdf. The idea is that the jitter is
|
||||
* composed of:
|
||||
*
|
||||
* J = N + n
|
||||
*
|
||||
* N : a constant network delay.
|
||||
* n : random added noise. The noise is concentrated around 0
|
||||
*
|
||||
* In the receiver we can track the elapsed time at the sender with:
|
||||
*
|
||||
* send_diff(i) = (Tsi - Ts0);
|
||||
*
|
||||
* Tsi : The time at the sender at packet i
|
||||
* Ts0 : The time at the sender at the first packet
|
||||
*
|
||||
* This is the difference between the RTP timestamp in the first received packet
|
||||
* and the current packet.
|
||||
*
|
||||
* At the receiver we have to deal with the jitter introduced by the network.
|
||||
*
|
||||
* recv_diff(i) = (Tri - Tr0)
|
||||
*
|
||||
* Tri : The time at the receiver at packet i
|
||||
* Tr0 : The time at the receiver at the first packet
|
||||
*
|
||||
* Both of these values contain a jitter Ji, a jitter for packet i, so we can
|
||||
* write:
|
||||
*
|
||||
* recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
|
||||
*
|
||||
* Cri : The time of the clock at the receiver for packet i
|
||||
* D + ni : The jitter when receiving packet i
|
||||
*
|
||||
* We see that the network delay is irrelevant here as we can elliminate D:
|
||||
*
|
||||
* recv_diff(i) = (Cri + ni) - (Cr0 + n0))
|
||||
*
|
||||
* The drift is now expressed as:
|
||||
*
|
||||
* Drift(i) = recv_diff(i) - send_diff(i);
|
||||
*
|
||||
* We now keep the W latest values of Drift and find the minimum (this is the
|
||||
* one with the lowest network jitter and thus the one which is least affected
|
||||
* by it). We average this lowest value to smooth out the resulting network skew.
|
||||
*
|
||||
* Both the window and the weighting used for averaging influence the accuracy
|
||||
* of the drift estimation. Finding the correct parameters turns out to be a
|
||||
* compromise between accuracy and inertia.
|
||||
*/
|
||||
static void
|
||||
calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
|
||||
{
|
||||
guint64 ext_rtptime;
|
||||
guint64 send_diff, recv_diff;
|
||||
gint64 delta;
|
||||
gint64 old;
|
||||
gint pos, i;
|
||||
GstClockTime gstrtptime;
|
||||
|
||||
ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
|
||||
|
||||
gstrtptime =
|
||||
gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
|
||||
|
||||
/* first time, lock on to time and gstrtptime */
|
||||
if (jbuf->base_time == -1)
|
||||
jbuf->base_time = time;
|
||||
if (jbuf->base_rtptime == -1)
|
||||
jbuf->base_rtptime = gstrtptime;
|
||||
|
||||
/* elapsed time at sender */
|
||||
send_diff = gstrtptime - jbuf->base_rtptime;
|
||||
/* elapsed time at receiver, includes the jitter */
|
||||
recv_diff = time - jbuf->base_time;
|
||||
|
||||
/* measure the diff */
|
||||
delta = ((gint64) recv_diff) - ((gint64) send_diff);
|
||||
|
||||
pos = jbuf->window_pos;
|
||||
|
||||
if (jbuf->window_filling) {
|
||||
/* we are filling the window */
|
||||
GST_DEBUG ("filling %d %" G_GINT64_FORMAT, pos, delta);
|
||||
jbuf->window[pos++] = delta;
|
||||
/* calc the min delta we observed */
|
||||
if (pos == 1 || delta < jbuf->window_min)
|
||||
jbuf->window_min = delta;
|
||||
|
||||
if (pos >= 100) {
|
||||
/* window filled, fill window with min */
|
||||
GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
|
||||
for (i = 0; i < 100; i++)
|
||||
jbuf->window[i] = jbuf->window_min;
|
||||
|
||||
/* the skew is initially the min */
|
||||
jbuf->skew = jbuf->window_min;
|
||||
jbuf->window_filling = FALSE;
|
||||
}
|
||||
} else {
|
||||
/* pick old value and store new value. We keep the previous value in order
|
||||
* to quickly check if the min of the window changed */
|
||||
old = jbuf->window[pos];
|
||||
jbuf->window[pos++] = delta;
|
||||
|
||||
if (delta <= jbuf->window_min) {
|
||||
/* if the new value we inserted is smaller or equal to the current min,
|
||||
* it becomes the new min */
|
||||
jbuf->window_min = delta;
|
||||
} else if (old == jbuf->window_min) {
|
||||
gint64 min = G_MAXINT64;
|
||||
|
||||
/* if we removed the old min, we have to find a new min */
|
||||
for (i = 0; i < 100; i++) {
|
||||
/* we found another value equal to the old min, we can stop searching now */
|
||||
if (jbuf->window[i] == old) {
|
||||
min = old;
|
||||
break;
|
||||
}
|
||||
if (jbuf->window[i] < min)
|
||||
min = jbuf->window[i];
|
||||
}
|
||||
jbuf->window_min = min;
|
||||
}
|
||||
/* average the min values */
|
||||
jbuf->skew = (jbuf->window_min + (15 * jbuf->skew)) / 16;
|
||||
GST_DEBUG ("new min: %" G_GINT64_FORMAT ", skew %" G_GINT64_FORMAT,
|
||||
jbuf->window_min, jbuf->skew);
|
||||
}
|
||||
/* wrap around in the window */
|
||||
if (pos >= 100)
|
||||
pos = 0;
|
||||
jbuf->window_pos = pos;
|
||||
}
|
||||
|
||||
static gint
|
||||
compare_seqnum (GstBuffer * a, GstBuffer * b, RTPJitterBuffer * jbuf)
|
||||
{
|
||||
|
@ -115,6 +290,7 @@ compare_seqnum (GstBuffer * a, GstBuffer * b, RTPJitterBuffer * jbuf)
|
|||
* rtp_jitter_buffer_insert:
|
||||
* @jbuf: an #RTPJitterBuffer
|
||||
* @buf: a buffer
|
||||
* @time: a timestamp when this buffer was received in nanoseconds
|
||||
*
|
||||
* Inserts @buf into the packet queue of @jbuf. The sequence number of the
|
||||
* packet will be used to sort the packets. This function takes ownerhip of
|
||||
|
@ -123,10 +299,12 @@ compare_seqnum (GstBuffer * a, GstBuffer * b, RTPJitterBuffer * jbuf)
|
|||
* Returns: %FALSE if a packet with the same number already existed.
|
||||
*/
|
||||
gboolean
|
||||
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf)
|
||||
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
|
||||
GstClockTime time)
|
||||
{
|
||||
GList *list;
|
||||
gint func_ret = 1;
|
||||
guint32 rtptime;
|
||||
|
||||
g_return_val_if_fail (jbuf != NULL, FALSE);
|
||||
g_return_val_if_fail (buf != NULL, FALSE);
|
||||
|
@ -142,11 +320,23 @@ rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf)
|
|||
if (func_ret == 0)
|
||||
return FALSE;
|
||||
|
||||
/* do skew calculation by measuring the difference between rtptime and the
|
||||
* receive time */
|
||||
if (time != -1) {
|
||||
rtptime = gst_rtp_buffer_get_timestamp (buf);
|
||||
calculate_skew (jbuf, rtptime, time);
|
||||
}
|
||||
|
||||
if (list)
|
||||
g_queue_insert_before (jbuf->packets, list, buf);
|
||||
else
|
||||
else {
|
||||
g_queue_push_tail (jbuf->packets, buf);
|
||||
|
||||
/* tail buffer changed, signal callback */
|
||||
if (jbuf->tail_changed)
|
||||
jbuf->tail_changed (jbuf, jbuf->user_data);
|
||||
}
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
|
@ -170,6 +360,28 @@ rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf)
|
|||
return buf;
|
||||
}
|
||||
|
||||
/**
|
||||
* rtp_jitter_buffer_peek:
|
||||
* @jbuf: an #RTPJitterBuffer
|
||||
*
|
||||
* Peek the oldest buffer from the packet queue of @jbuf. Register a callback
|
||||
* with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
|
||||
* was inserted in the queue.
|
||||
*
|
||||
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
|
||||
*/
|
||||
GstBuffer *
|
||||
rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
|
||||
{
|
||||
GstBuffer *buf;
|
||||
|
||||
g_return_val_if_fail (jbuf != NULL, FALSE);
|
||||
|
||||
buf = g_queue_peek_tail (jbuf->packets);
|
||||
|
||||
return buf;
|
||||
}
|
||||
|
||||
/**
|
||||
* rtp_jitter_buffer_flush:
|
||||
* @jbuf: an #RTPJitterBuffer
|
||||
|
|
|
@ -34,15 +34,39 @@ typedef struct _RTPJitterBufferClass RTPJitterBufferClass;
|
|||
#define RTP_IS_JITTER_BUFFER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),RTP_TYPE_JITTER_BUFFER))
|
||||
#define RTP_JITTER_BUFFER_CAST(src) ((RTPJitterBuffer *)(src))
|
||||
|
||||
/**
|
||||
* RTPTailChanged:
|
||||
* @jbuf: an #RTPJitterBuffer
|
||||
* @user_data: user data specified when registering
|
||||
*
|
||||
* This callback will be called when the tail buffer of @jbuf changed.
|
||||
*/
|
||||
typedef void (*RTPTailChanged) (RTPJitterBuffer *jbuf, gpointer user_data);
|
||||
|
||||
/**
|
||||
* RTPJitterBuffer:
|
||||
*
|
||||
* A JitterBuffer in the #RTPSession
|
||||
*/
|
||||
struct _RTPJitterBuffer {
|
||||
GObject object;
|
||||
GObject object;
|
||||
|
||||
GQueue *packets;
|
||||
GQueue *packets;
|
||||
|
||||
gint clock_rate;
|
||||
|
||||
/* for calculating skew */
|
||||
GstClockTime base_time;
|
||||
GstClockTime base_rtptime;
|
||||
guint64 ext_rtptime;
|
||||
gint64 window[100];
|
||||
guint window_pos;
|
||||
gboolean window_filling;
|
||||
gint64 window_min;
|
||||
gint64 skew;
|
||||
|
||||
RTPTailChanged tail_changed;
|
||||
gpointer user_data;
|
||||
};
|
||||
|
||||
struct _RTPJitterBufferClass {
|
||||
|
@ -52,14 +76,20 @@ struct _RTPJitterBufferClass {
|
|||
GType rtp_jitter_buffer_get_type (void);
|
||||
|
||||
/* managing lifetime */
|
||||
RTPJitterBuffer* rtp_jitter_buffer_new (void);
|
||||
RTPJitterBuffer* rtp_jitter_buffer_new (void);
|
||||
|
||||
gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf, GstBuffer *buf);
|
||||
GstBuffer * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf);
|
||||
void rtp_jitter_buffer_set_tail_changed (RTPJitterBuffer *jbuf, RTPTailChanged func,
|
||||
gpointer user_data);
|
||||
|
||||
void rtp_jitter_buffer_flush (RTPJitterBuffer *jbuf);
|
||||
void rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer *jbuf, gint clock_rate);
|
||||
gint rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer *jbuf);
|
||||
|
||||
guint rtp_jitter_buffer_num_packets (RTPJitterBuffer *jbuf);
|
||||
guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer *jbuf);
|
||||
gboolean rtp_jitter_buffer_insert (RTPJitterBuffer *jbuf, GstBuffer *buf, GstClockTime time);
|
||||
GstBuffer * rtp_jitter_buffer_pop (RTPJitterBuffer *jbuf);
|
||||
|
||||
void rtp_jitter_buffer_flush (RTPJitterBuffer *jbuf);
|
||||
|
||||
guint rtp_jitter_buffer_num_packets (RTPJitterBuffer *jbuf);
|
||||
guint32 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer *jbuf);
|
||||
|
||||
#endif /* __RTP_JITTER_BUFFER_H__ */
|
||||
|
|
|
@ -69,9 +69,6 @@ rtp_source_init (RTPSource * src)
|
|||
src->payload = 0;
|
||||
src->clock_rate = -1;
|
||||
src->clock_base = -1;
|
||||
src->skew_base_ntpnstime = -1;
|
||||
src->ext_rtptime = -1;
|
||||
src->prev_ext_rtptime = -1;
|
||||
src->packets = g_queue_new ();
|
||||
src->seqnum_base = -1;
|
||||
src->last_rtptime = -1;
|
||||
|
@ -266,18 +263,20 @@ get_clock_rate (RTPSource * src, guint8 payload)
|
|||
return src->clock_rate;
|
||||
}
|
||||
|
||||
/* Jitter is the variation in the delay of received packets in a flow. It is
|
||||
* measured by comparing the interval when RTP packets were sent to the interval
|
||||
* at which they were received. For instance, if packet #1 and packet #2 leave
|
||||
* 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
|
||||
* milliseconds. */
|
||||
static void
|
||||
calculate_jitter (RTPSource * src, GstBuffer * buffer,
|
||||
RTPArrivalStats * arrival)
|
||||
{
|
||||
guint64 ntpnstime;
|
||||
guint32 rtparrival, transit, rtptime;
|
||||
guint64 ext_rtptime;
|
||||
gint32 diff;
|
||||
gint clock_rate;
|
||||
guint8 pt;
|
||||
guint64 rtpdiff, ntpdiff;
|
||||
gint64 skew;
|
||||
|
||||
/* get arrival time */
|
||||
if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE)
|
||||
|
@ -291,50 +290,12 @@ calculate_jitter (RTPSource * src, GstBuffer * buffer,
|
|||
|
||||
rtptime = gst_rtp_buffer_get_timestamp (buffer);
|
||||
|
||||
/* convert to extended timestamp right away */
|
||||
ext_rtptime = gst_rtp_buffer_ext_timestamp (&src->ext_rtptime, rtptime);
|
||||
|
||||
/* no clock-base, take first rtptime as base */
|
||||
if (src->clock_base == -1) {
|
||||
GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
|
||||
src->clock_base = rtptime;
|
||||
}
|
||||
|
||||
if (src->skew_base_ntpnstime == -1) {
|
||||
/* lock on first observed NTP and RTP time, they should increment in-sync or
|
||||
* we have a clock skew. */
|
||||
GST_DEBUG ("using base_ntpnstime of %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (ntpnstime));
|
||||
src->skew_base_ntpnstime = ntpnstime;
|
||||
src->skew_base_rtptime = rtptime;
|
||||
src->prev_ext_rtptime = ext_rtptime;
|
||||
src->avg_skew = 0;
|
||||
} else if (src->prev_ext_rtptime < ext_rtptime) {
|
||||
/* get elapsed rtptime but only when the previous rtptime was stricly smaller
|
||||
* than the new one. */
|
||||
rtpdiff = ext_rtptime - src->skew_base_rtptime;
|
||||
/* get NTP diff and convert to RTP time, this is always positive */
|
||||
ntpdiff = ntpnstime - src->skew_base_ntpnstime;
|
||||
ntpdiff = gst_util_uint64_scale_int (ntpdiff, clock_rate, GST_SECOND);
|
||||
|
||||
/* see how the NTP and RTP relate any deviation from 0 means that they drift
|
||||
* out of sync and we must compensate. */
|
||||
skew = ntpdiff - rtpdiff;
|
||||
/* average out the skew to get a smooth value. */
|
||||
src->avg_skew = (63 * src->avg_skew + skew) / 64;
|
||||
|
||||
GST_DEBUG ("new skew %" G_GINT64_FORMAT ", avg %" G_GINT64_FORMAT, skew,
|
||||
src->avg_skew);
|
||||
/* store previous extended timestamp */
|
||||
src->prev_ext_rtptime = ext_rtptime;
|
||||
}
|
||||
if (src->avg_skew != 0) {
|
||||
/* patch the buffer RTP timestamp with the skew */
|
||||
GST_DEBUG ("skew timestamp RTP %" G_GUINT32_FORMAT " -> %" G_GINT64_FORMAT,
|
||||
rtptime, rtptime + src->avg_skew);
|
||||
gst_rtp_buffer_set_timestamp (buffer, rtptime + src->avg_skew);
|
||||
}
|
||||
|
||||
/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
|
||||
* care about the absolute value, just the difference. */
|
||||
rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
|
||||
|
@ -603,7 +564,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
|
|||
/* the SSRC of the packet is not correct, make a writable buffer and
|
||||
* update the SSRC. This could involve a complete copy of the packet when
|
||||
* it is not writable. Usually the payloader will use caps negotiation to
|
||||
* get the correct SSRC. */
|
||||
* get the correct SSRC from the session manager before pushing anything. */
|
||||
buffer = gst_buffer_make_writable (buffer);
|
||||
|
||||
GST_WARNING ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
|
||||
|
@ -614,7 +575,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
|
|||
src->stats.packets_sent);
|
||||
result = src->callbacks.push_rtp (src, buffer, src->user_data);
|
||||
} else {
|
||||
GST_DEBUG ("no callback installed");
|
||||
GST_WARNING ("no callback installed, dropping packet");
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
|
||||
|
|
|
@ -137,16 +137,8 @@ struct _RTPSource {
|
|||
GstCaps *caps;
|
||||
gint clock_rate;
|
||||
gint32 seqnum_base;
|
||||
|
||||
gint64 clock_base;
|
||||
|
||||
/* to calculate the clock skew */
|
||||
guint64 skew_base_ntpnstime;
|
||||
guint64 skew_base_rtptime;
|
||||
gint64 avg_skew;
|
||||
guint64 ext_rtptime;
|
||||
guint64 prev_ext_rtptime;
|
||||
|
||||
GstClockTime bye_time;
|
||||
GstClockTime last_activity;
|
||||
GstClockTime last_rtp_activity;
|
||||
|
|
Loading…
Reference in a new issue