static void
kms_rtp_synchronizer_process_rtcp_packet (KmsRtpSynchronizer * self,
GstRTCPPacket * packet, GstClockTime current_time)
{
GstRTCPType type;
guint32 ssrc, rtp_time;
guint64 ntp_time, ntp_ns_time;
type = gst_rtcp_packet_get_type (packet);
GST_DEBUG_OBJECT (self, "Received RTCP buffer of type: %d", type);
if (type != GST_RTCP_TYPE_SR) {
return;
}
gst_rtcp_packet_sr_get_sender_info (packet, &ssrc, &ntp_time, &rtp_time,
NULL, NULL);
/* convert ntp_time to nanoseconds */
ntp_ns_time =
gst_util_uint64_scale (ntp_time, GST_SECOND,
(G_GINT64_CONSTANT (1) << 32));
KMS_RTP_SYNCHRONIZER_LOCK (self);
GST_DEBUG_OBJECT (self,
"Received RTCP SR packet SSRC: %u, rtp_time: %u, ntp_time: %"
G_GUINT64_FORMAT ", ntp_ns_time: %" GST_TIME_FORMAT, ssrc, rtp_time,
ntp_time, GST_TIME_ARGS (ntp_ns_time));
if (!self->priv->base_initiated) {
kms_rtp_sync_context_get_time_matching (self->priv->context, ntp_ns_time,
current_time, &self->priv->base_ntp_ns_time,
&self->priv->base_sync_time);
self->priv->base_initiated = TRUE;
}
self->priv->last_sr_ext_ts =
gst_rtp_buffer_ext_timestamp (&self->priv->ext_ts, rtp_time);
self->priv->last_sr_ntp_ns_time = ntp_ns_time;
KMS_RTP_SYNCHRONIZER_UNLOCK (self);
}
/* 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.
*
* We use a 2 second window or up to 512 data points, which is statistically big
* enough to catch spikes (FIXME, detect spikes).
* We also use a rather large weighting factor (125) to smoothly adapt. During
* startup, when filling the window, we use a parabolic weighting factor, the
* more the window is filled, the faster we move to the detected possible skew.
*
* Returns: @time adjusted with the clock skew.
*/
static GstClockTime
calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time,
guint32 clock_rate, GstClockTime max_delay)
{
guint64 ext_rtptime;
guint64 send_diff, recv_diff;
gint64 delta;
gint64 old;
gint pos, i;
GstClockTime gstrtptime, out_time;
guint64 slope;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
/* keep track of the last extended rtptime */
jbuf->last_rtptime = ext_rtptime;
if (jbuf->clock_rate != clock_rate) {
if (jbuf->clock_rate == -1) {
GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
} else {
GST_WARNING ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
}
jbuf->base_time = -1;
jbuf->base_rtptime = -1;
jbuf->clock_rate = clock_rate;
jbuf->prev_out_time = -1;
jbuf->prev_send_diff = -1;
}
/* first time, lock on to time and gstrtptime */
if (G_UNLIKELY (jbuf->base_time == -1)) {
jbuf->base_time = time;
jbuf->prev_out_time = -1;
GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
}
if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
jbuf->base_rtptime = gstrtptime;
jbuf->base_extrtp = ext_rtptime;
jbuf->prev_send_diff = -1;
GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
GST_TIME_ARGS (gstrtptime));
}
if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
send_diff = gstrtptime - jbuf->base_rtptime;
else {
/* elapsed time at sender, timestamps can go backwards and thus be smaller
* than our base time, take a new base time in that case. */
GST_WARNING ("backward timestamps at server, taking new base time");
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
send_diff = 0;
}
GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
GST_TIME_ARGS (send_diff));
/* we don't have an arrival timestamp so we can't do skew detection. we
* should still apply a timestamp based on RTP timestamp and base_time */
if (time == -1 || jbuf->base_time == -1)
goto no_skew;
/* elapsed time at receiver, includes the jitter */
recv_diff = time - jbuf->base_time;
/* measure the diff */
delta = ((gint64) recv_diff) - ((gint64) send_diff);
/* measure the slope, this gives a rought estimate between the sender speed
* and the receiver speed. This should be approximately 8, higher values
* indicate a burst (especially when the connection starts) */
if (recv_diff > 0)
slope = (send_diff * 8) / recv_diff;
else
slope = 8;
GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
/* if the difference between the sender timeline and the receiver timeline
* changed too quickly we have to resync because the server likely restarted
* its timestamps. */
if (ABS (delta - jbuf->skew) > GST_SECOND) {
GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
GST_TIME_ARGS (delta - jbuf->skew));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
send_diff = 0;
delta = 0;
}
pos = jbuf->window_pos;
if (G_UNLIKELY (jbuf->window_filling)) {
/* we are filling the window */
GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
jbuf->window[pos++] = delta;
/* calc the min delta we observed */
if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
jbuf->window_min = delta;
if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
jbuf->window_size = pos;
/* window filled */
GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
/* the skew is now the min */
jbuf->skew = jbuf->window_min;
jbuf->window_filling = FALSE;
} else {
gint perc_time, perc_window, perc;
/* figure out how much we filled the window, this depends on the amount of
* time we have or the max number of points we keep. */
perc_time = send_diff * 100 / MAX_TIME;
perc_window = pos * 100 / MAX_WINDOW;
perc = MAX (perc_time, perc_window);
/* make a parabolic function, the closer we get to the MAX, the more value
* we give to the scaling factor of the new value */
perc = perc * perc;
/* quickly go to the min value when we are filling up, slowly when we are
* just starting because we're not sure it's a good value yet. */
jbuf->skew =
(perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
jbuf->window_size = pos + 1;
}
} 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 (G_UNLIKELY (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 (G_UNLIKELY (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 < jbuf->window_size; 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 + (124 * jbuf->skew)) / 125;
GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
delta, jbuf->window_min);
}
/* wrap around in the window */
if (G_UNLIKELY (pos >= jbuf->window_size))
pos = 0;
jbuf->window_pos = pos;
no_skew:
/* the output time is defined as the base timestamp plus the RTP time
* adjusted for the clock skew .*/
if (jbuf->base_time != -1) {
out_time = jbuf->base_time + send_diff;
/* skew can be negative and we don't want to make invalid timestamps */
if (jbuf->skew < 0 && out_time < -jbuf->skew) {
out_time = 0;
} else {
out_time += jbuf->skew;
}
/* check if timestamps are not going backwards, we can only check this if we
* have a previous out time and a previous send_diff */
if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
/* now check for backwards timestamps */
if (G_UNLIKELY (
/* if the server timestamps went up and the out_time backwards */
(send_diff > jbuf->prev_send_diff
&& out_time < jbuf->prev_out_time) ||
/* if the server timestamps went backwards and the out_time forwards */
(send_diff < jbuf->prev_send_diff
&& out_time > jbuf->prev_out_time) ||
/* if the server timestamps did not change */
send_diff == jbuf->prev_send_diff)) {
GST_DEBUG ("backwards timestamps, using previous time");
out_time = jbuf->prev_out_time;
}
}
if (time != -1 && out_time + max_delay < time) {
/* if we are going to produce a timestamp that is later than the input
* timestamp, we need to reset the jitterbuffer. Likely the server paused
* temporarily */
GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
max_delay, GST_TIME_ARGS (time));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
out_time = time;
send_diff = 0;
}
} else
out_time = -1;
jbuf->prev_out_time = out_time;
jbuf->prev_send_diff = send_diff;
GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
jbuf->skew, GST_TIME_ARGS (out_time));
return out_time;
}
gboolean
kms_rtp_synchronizer_process_rtp_buffer_mapped (KmsRtpSynchronizer * self,
GstRTPBuffer * rtp_buffer, GError ** error)
{
GstBuffer *buffer = rtp_buffer->buffer;
guint64 pts_orig, ext_ts, last_sr_ext_ts, last_sr_ntp_ns_time;
guint64 diff_ntp_ns_time;
guint8 pt;
guint32 ssrc, ts;
gint32 clock_rate;
gboolean ret = TRUE;
ssrc = gst_rtp_buffer_get_ssrc (rtp_buffer);
KMS_RTP_SYNCHRONIZER_LOCK (self);
if (self->priv->ssrc == 0) {
self->priv->ssrc = ssrc;
} else if (ssrc != self->priv->ssrc) {
gchar *msg = g_strdup_printf ("Invalid SSRC (%u), not matching with %u",
ssrc, self->priv->ssrc);
GST_ERROR_OBJECT (self, "%s", msg);
g_set_error_literal (error, KMS_RTP_SYNC_ERROR, KMS_RTP_SYNC_INVALID_DATA,
msg);
g_free (msg);
KMS_RTP_SYNCHRONIZER_UNLOCK (self);
return FALSE;
}
pt = gst_rtp_buffer_get_payload_type (rtp_buffer);
if (pt != self->priv->pt || self->priv->clock_rate <= 0) {
gchar *msg =
g_strdup_printf ("Invalid clock-rate %d for PT %u, not changing PTS",
self->priv->clock_rate, pt);
GST_ERROR_OBJECT (self, "%s", msg);
g_set_error_literal (error, KMS_RTP_SYNC_ERROR, KMS_RTP_SYNC_INVALID_DATA,
msg);
g_free (msg);
KMS_RTP_SYNCHRONIZER_UNLOCK (self);
return FALSE;
}
pts_orig = GST_BUFFER_PTS (buffer);
ts = gst_rtp_buffer_get_timestamp (rtp_buffer);
gst_rtp_buffer_ext_timestamp (&self->priv->ext_ts, ts);
if (self->priv->feeded_sorted) {
if (self->priv->fs_last_ext_ts != -1
&& self->priv->ext_ts < self->priv->fs_last_ext_ts) {
guint16 seq = gst_rtp_buffer_get_seq (rtp_buffer);
gchar *msg =
g_strdup_printf
("Received an unsorted RTP buffer when expecting sorted (ssrc: %"
G_GUINT32_FORMAT ", seq: %" G_GUINT16_FORMAT ", ts: %"
G_GUINT32_FORMAT ", ext_ts: %" G_GUINT64_FORMAT
"). Moving to unsorted mode",
ssrc, seq, ts, self->priv->ext_ts);
GST_ERROR_OBJECT (self, "%s", msg);
g_set_error_literal (error, KMS_RTP_SYNC_ERROR, KMS_RTP_SYNC_INVALID_DATA,
msg);
g_free (msg);
self->priv->feeded_sorted = FALSE;
ret = FALSE;
} else if (self->priv->ext_ts == self->priv->fs_last_ext_ts) {
GST_BUFFER_PTS (buffer) = self->priv->fs_last_pts;
goto end;
}
}
if (!self->priv->base_initiated) {
GST_DEBUG_OBJECT (self,
"Do not sync data for SSRC %u and PT %u, interpolating PTS", ssrc, pt);
if (!self->priv->base_interpolate_initiated) {
self->priv->base_interpolate_ext_ts = self->priv->ext_ts;
self->priv->base_interpolate_time = GST_BUFFER_PTS (buffer);
self->priv->base_interpolate_initiated = TRUE;
} else {
buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_PTS (buffer) = self->priv->base_interpolate_time;
kms_rtp_synchronizer_rtp_diff (self, rtp_buffer, self->priv->clock_rate,
self->priv->base_interpolate_ext_ts);
}
} else {
gboolean wrapped_down, wrapped_up;
wrapped_down = wrapped_up = FALSE;
buffer = gst_buffer_make_writable (buffer);
GST_BUFFER_PTS (buffer) = self->priv->base_sync_time;
if (self->priv->last_sr_ntp_ns_time > self->priv->base_ntp_ns_time) {
diff_ntp_ns_time =
self->priv->last_sr_ntp_ns_time - self->priv->base_ntp_ns_time;
wrapped_up = diff_ntp_ns_time > (G_MAXUINT64 - GST_BUFFER_PTS (buffer));
GST_BUFFER_PTS (buffer) += diff_ntp_ns_time;
} else if (self->priv->last_sr_ntp_ns_time < self->priv->base_ntp_ns_time) {
diff_ntp_ns_time =
self->priv->base_ntp_ns_time - self->priv->last_sr_ntp_ns_time;
wrapped_down = GST_BUFFER_PTS (buffer) < diff_ntp_ns_time;
GST_BUFFER_PTS (buffer) -= diff_ntp_ns_time;
}
/* if equals do nothing */
kms_rtp_synchronizer_rtp_diff_full (self, rtp_buffer,
self->priv->clock_rate, self->priv->last_sr_ext_ts, wrapped_down,
wrapped_up);
}
if (self->priv->feeded_sorted) {
if (GST_BUFFER_PTS (buffer) < self->priv->fs_last_pts) {
guint16 seq = gst_rtp_buffer_get_seq (rtp_buffer);
GST_WARNING_OBJECT (self,
"Non monotonic PTS assignment in sorted mode (ssrc: %"
G_GUINT32_FORMAT ", seq: %" G_GUINT16_FORMAT ", ts: %"
G_GUINT32_FORMAT ", ext_ts: %" G_GUINT64_FORMAT
"). Forcing monotonic", ssrc, seq, ts, self->priv->ext_ts);
GST_BUFFER_PTS (buffer) = self->priv->fs_last_pts;
}
self->priv->fs_last_ext_ts = self->priv->ext_ts;
self->priv->fs_last_pts = GST_BUFFER_PTS (buffer);
}
end:
clock_rate = self->priv->clock_rate;
ext_ts = self->priv->ext_ts;
last_sr_ext_ts = self->priv->last_sr_ext_ts;
last_sr_ntp_ns_time = self->priv->last_sr_ntp_ns_time;
KMS_RTP_SYNCHRONIZER_UNLOCK (self);
kms_rtp_sync_context_write_stats (self->priv->context, ssrc, clock_rate,
pts_orig, GST_BUFFER_PTS (buffer), GST_BUFFER_DTS (buffer), ext_ts,
last_sr_ntp_ns_time, last_sr_ext_ts);
return ret;
}
/**
* rtp_jitter_buffer_insert:
* @jbuf: an #RTPJitterBuffer
* @item: an #RTPJitterBufferItem to insert
* @tail: TRUE when the tail element changed.
* @percent: the buffering percent after insertion
*
* Inserts @item into the packet queue of @jbuf. The sequence number of the
* packet will be used to sort the packets. This function takes ownerhip of
* @buf when the function returns %TRUE.
*
* Returns: %FALSE if a packet with the same number already existed.
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
gboolean * tail, gint * percent)
{
GList *list = NULL;
guint32 rtptime;
guint16 seqnum;
GstClockTime dts;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (item != NULL, FALSE);
/* no seqnum, simply append then */
if (item->seqnum == -1) {
goto append;
}
seqnum = item->seqnum;
/* loop the list to skip strictly smaller seqnum buffers */
for (list = jbuf->packets->head; list; list = g_list_next (list)) {
guint16 qseq;
gint gap;
RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
if (qitem->seqnum == -1)
continue;
qseq = qitem->seqnum;
/* compare the new seqnum to the one in the buffer */
gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
/* we hit a packet with the same seqnum, notify a duplicate */
if (G_UNLIKELY (gap == 0))
goto duplicate;
/* seqnum < qseq, we can stop looking */
if (G_LIKELY (gap > 0))
break;
}
dts = item->dts;
if (item->rtptime == -1)
goto append;
rtptime = item->rtptime;
/* rtp time jumps are checked for during skew calculation, but bypassed
* in other mode, so mind those here and reset jb if needed.
* Only reset if valid input time, which is likely for UDP input
* where we expect this might happen due to async thread effects
* (in seek and state change cycles), but not so much for TCP input */
if (GST_CLOCK_TIME_IS_VALID (dts) &&
jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
GstClockTime ext_rtptime = jbuf->ext_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
/* reset even if we don't have valid incoming time;
* still better than producing possibly very bogus output timestamp */
GST_WARNING ("rtp delta too big, reset skew");
rtp_jitter_buffer_reset_skew (jbuf);
}
}
switch (jbuf->mode) {
case RTP_JITTER_BUFFER_MODE_NONE:
case RTP_JITTER_BUFFER_MODE_BUFFER:
/* send 0 as the first timestamp and -1 for the other ones. This will
* interpollate them from the RTP timestamps with a 0 origin. In buffering
* mode we will adjust the outgoing timestamps according to the amount of
* time we spent buffering. */
if (jbuf->base_time == -1)
dts = 0;
else
dts = -1;
break;
case RTP_JITTER_BUFFER_MODE_SYNCED:
/* synchronized clocks, take first timestamp as base, use RTP timestamps
* to interpolate */
if (jbuf->base_time != -1)
dts = -1;
break;
case RTP_JITTER_BUFFER_MODE_SLAVE:
default:
break;
}
/* do skew calculation by measuring the difference between rtptime and the
* receive dts, this function will return the skew corrected rtptime. */
item->pts = calculate_skew (jbuf, rtptime, dts);
append:
queue_do_insert (jbuf, list, (GList *) item);
/* buffering mode, update buffer stats */
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
update_buffer_level (jbuf, percent);
else if (percent)
*percent = -1;
/* tail was changed when we did not find a previous packet, we set the return
* flag when requested. */
if (G_LIKELY (tail))
*tail = (list == NULL);
return TRUE;
/* ERRORS */
duplicate:
{
GST_WARNING ("duplicate packet %d found", (gint) seqnum);
return FALSE;
}
}
static GstEvent *
create_segment_event (GstRTPBaseDepayload * filter, guint rtptime,
GstClockTime position)
{
GstEvent *event;
GstClockTime start, stop, running_time;
GstRTPBaseDepayloadPrivate *priv;
GstSegment segment;
priv = filter->priv;
/* We don't need the object lock around - the segment
* can't change here while we're holding the STREAM_LOCK
*/
/* determining the start of the segment */
start = filter->segment.start;
if (priv->clock_base != -1 && position != -1) {
GstClockTime exttime, gap;
exttime = priv->clock_base;
gst_rtp_buffer_ext_timestamp (&exttime, rtptime);
gap = gst_util_uint64_scale_int (exttime - priv->clock_base,
filter->clock_rate, GST_SECOND);
/* account for lost packets */
if (position > gap) {
GST_DEBUG_OBJECT (filter,
"Found gap of %" GST_TIME_FORMAT ", adjusting start: %"
GST_TIME_FORMAT " = %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (gap), GST_TIME_ARGS (position - gap),
GST_TIME_ARGS (position), GST_TIME_ARGS (gap));
start = position - gap;
}
}
/* determining the stop of the segment */
stop = filter->segment.stop;
if (priv->npt_stop != -1)
stop = start + (priv->npt_stop - priv->npt_start);
if (position == -1)
position = start;
running_time = gst_segment_to_running_time (&filter->segment,
GST_FORMAT_TIME, start);
gst_segment_init (&segment, GST_FORMAT_TIME);
segment.rate = priv->play_speed;
segment.applied_rate = priv->play_scale;
segment.start = start;
segment.stop = stop;
segment.time = priv->npt_start;
segment.position = position;
segment.base = running_time;
GST_DEBUG_OBJECT (filter, "Creating segment event %" GST_SEGMENT_FORMAT,
&segment);
event = gst_event_new_segment (&segment);
return event;
}
/**
* rtp_jitter_buffer_insert:
* @jbuf: an #RTPJitterBuffer
* @buf: a buffer
* @time: a running_time when this buffer was received in nanoseconds
* @clock_rate: the clock-rate of the payload of @buf
* @max_delay: the maximum lateness of @buf
* @tail: TRUE when the tail element changed.
*
* 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
* @buf when the function returns %TRUE.
* @buf should have writable metadata when calling this function.
*
* Returns: %FALSE if a packet with the same number already existed.
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
GstClockTime time, guint32 clock_rate, gboolean * tail, gint * percent)
{
GList *list;
guint32 rtptime;
guint16 seqnum;
GstRTPBuffer rtp = {NULL};
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (buf != NULL, FALSE);
gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
seqnum = gst_rtp_buffer_get_seq (&rtp);
/* loop the list to skip strictly smaller seqnum buffers */
for (list = jbuf->packets->head; list; list = g_list_next (list)) {
guint16 qseq;
gint gap;
GstRTPBuffer rtpb = {NULL};
gst_rtp_buffer_map (GST_BUFFER_CAST (list->data), GST_MAP_READ, &rtpb);
qseq = gst_rtp_buffer_get_seq (&rtpb);
gst_rtp_buffer_unmap (&rtpb);
/* compare the new seqnum to the one in the buffer */
gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
/* we hit a packet with the same seqnum, notify a duplicate */
if (G_UNLIKELY (gap == 0))
goto duplicate;
/* seqnum > qseq, we can stop looking */
if (G_LIKELY (gap < 0))
break;
}
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
/* rtp time jumps are checked for during skew calculation, but bypassed
* in other mode, so mind those here and reset jb if needed.
* Only reset if valid input time, which is likely for UDP input
* where we expect this might happen due to async thread effects
* (in seek and state change cycles), but not so much for TCP input */
if (GST_CLOCK_TIME_IS_VALID (time) &&
jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
GstClockTime ext_rtptime = jbuf->ext_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
if (ext_rtptime > jbuf->last_rtptime + 3 * clock_rate ||
ext_rtptime + 3 * clock_rate < jbuf->last_rtptime) {
/* reset even if we don't have valid incoming time;
* still better than producing possibly very bogus output timestamp */
GST_WARNING ("rtp delta too big, reset skew");
rtp_jitter_buffer_reset_skew (jbuf);
}
}
switch (jbuf->mode) {
case RTP_JITTER_BUFFER_MODE_NONE:
case RTP_JITTER_BUFFER_MODE_BUFFER:
/* send 0 as the first timestamp and -1 for the other ones. This will
* interpollate them from the RTP timestamps with a 0 origin. In buffering
* mode we will adjust the outgoing timestamps according to the amount of
* time we spent buffering. */
if (jbuf->base_time == -1)
time = 0;
else
time = -1;
break;
case RTP_JITTER_BUFFER_MODE_SLAVE:
default:
break;
}
/* do skew calculation by measuring the difference between rtptime and the
* receive time, this function will retimestamp @buf with the skew corrected
* running time. */
time = calculate_skew (jbuf, rtptime, time, clock_rate);
GST_BUFFER_TIMESTAMP (buf) = time;
/* It's more likely that the packet was inserted in the front of the buffer */
if (G_LIKELY (list))
g_queue_insert_before (jbuf->packets, list, buf);
else
g_queue_push_tail (jbuf->packets, buf);
/* buffering mode, update buffer stats */
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
update_buffer_level (jbuf, percent);
else
*percent = -1;
/* tail was changed when we did not find a previous packet, we set the return
* flag when requested. */
if (G_LIKELY (tail))
*tail = (list == NULL);
gst_rtp_buffer_unmap (&rtp);
return TRUE;
/* ERRORS */
duplicate:
{
gst_rtp_buffer_unmap (&rtp);
GST_WARNING ("duplicate packet %d found", (gint) seqnum);
return FALSE;
}
}
