/**
* 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, GstClockTime max_delay,
gboolean * tail)
{
GList *list;
guint32 rtptime;
guint16 seqnum;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (buf != NULL, FALSE);
seqnum = gst_rtp_buffer_get_seq (buf);
/* loop the list to skip strictly smaller seqnum buffers */
for (list = jbuf->packets->head; list; list = g_list_next (list)) {
guint16 qseq;
gint gap;
qseq = gst_rtp_buffer_get_seq (GST_BUFFER_CAST (list->data));
/* 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;
}
/* do skew calculation by measuring the difference between rtptime and the
* receive time, this function will retimestamp @buf with the skew corrected
* running time. */
rtptime = gst_rtp_buffer_get_timestamp (buf);
time = calculate_skew (jbuf, rtptime, time, clock_rate, max_delay);
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);
/* 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 gint
_compare_fec_map_info (gconstpointer a, gconstpointer b, gpointer userdata)
{
guint16 aseq =
gst_rtp_buffer_get_seq (&RTP_FEC_MAP_INFO_NTH (userdata, a)->rtp);
guint16 bseq =
gst_rtp_buffer_get_seq (&RTP_FEC_MAP_INFO_NTH (userdata, b)->rtp);
return gst_rtp_buffer_compare_seqnum (bseq, aseq);
}
static GstFlowReturn
gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
{
GstBaseRTPDepayload *filter;
GstBaseRTPDepayloadPrivate *priv;
GstBaseRTPDepayloadClass *bclass;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
GstClockTime timestamp;
guint16 seqnum;
guint32 rtptime;
gboolean reset_seq, discont;
gint gap;
filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
priv = filter->priv;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
/* we must validate, it's possible that this element is plugged right after a
* network receiver and we don't want to operate on invalid data */
if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
goto invalid_buffer;
priv->discont = GST_BUFFER_IS_DISCONT (in);
timestamp = GST_BUFFER_TIMESTAMP (in);
/* convert to running_time and save the timestamp, this is the timestamp
* we put on outgoing buffers. */
timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
timestamp);
priv->timestamp = timestamp;
priv->duration = GST_BUFFER_DURATION (in);
seqnum = gst_rtp_buffer_get_seq (in);
rtptime = gst_rtp_buffer_get_timestamp (in);
reset_seq = TRUE;
discont = FALSE;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
GST_TIME_ARGS (timestamp));
/* Check seqnum. This is a very simple check that makes sure that the seqnums
* are striclty increasing, dropping anything that is out of the ordinary. We
* can only do this when the next_seqnum is known. */
if (G_LIKELY (priv->next_seqnum != -1)) {
gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
/* if we have no gap, all is fine */
if (G_UNLIKELY (gap != 0)) {
GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
priv->next_seqnum, gap);
if (gap < 0) {
/* seqnum > next_seqnum, we are missing some packets, this is always a
* DISCONT. */
GST_LOG_OBJECT (filter, "%d missing packets", gap);
discont = TRUE;
} else {
/* seqnum < next_seqnum, we have seen this packet before or the sender
* could be restarted. If the packet is not too old, we throw it away as
* a duplicate, otherwise we mark discont and continue. 100 misordered
* packets is a good threshold. See also RFC 4737. */
if (gap < 100)
goto dropping;
GST_LOG_OBJECT (filter,
"%d > 100, packet too old, sender likely restarted", gap);
discont = TRUE;
}
}
}
priv->next_seqnum = (seqnum + 1) & 0xffff;
if (G_UNLIKELY (discont && !priv->discont)) {
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
/* we detected a seqnum discont but the buffer was not flagged with a discont,
* set the discont flag so that the subclass can throw away old data. */
priv->discont = TRUE;
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
}
bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
if (G_UNLIKELY (bclass->process == NULL))
goto no_process;
/* let's send it out to processing */
out_buf = bclass->process (filter, in);
if (out_buf) {
/* we pass rtptime as backward compatibility, in reality, the incomming
* buffer timestamp is always applied to the outgoing packet. */
ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
}
gst_buffer_unref (in);
return ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, (NULL),
("Not RTP format was negotiated"));
gst_buffer_unref (in);
return GST_FLOW_NOT_NEGOTIATED;
}
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (in);
return GST_FLOW_OK;
}
dropping:
{
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
gst_buffer_unref (in);
return GST_FLOW_OK;
}
no_process:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
("The subclass does not have a process method"));
gst_buffer_unref (in);
return GST_FLOW_ERROR;
}
}
/**
* 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 GstFlowReturn
gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
guint16 seqnum;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
guint64 latency_ts;
gboolean tail;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
if (!gst_rtp_buffer_validate (buffer))
goto invalid_buffer;
priv = jitterbuffer->priv;
if (priv->last_pt != gst_rtp_buffer_get_payload_type (buffer)) {
GstCaps *caps;
priv->last_pt = gst_rtp_buffer_get_payload_type (buffer);
/* reset clock-rate so that we get a new one */
priv->clock_rate = -1;
/* Try to get the clock-rate from the caps first if we can. If there are no
* caps we must fire the signal to get the clock-rate. */
if ((caps = GST_BUFFER_CAPS (buffer))) {
gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
}
}
if (priv->clock_rate == -1) {
guint8 pt;
/* no clock rate given on the caps, try to get one with the signal */
pt = gst_rtp_buffer_get_payload_type (buffer);
gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer, pt);
if (priv->clock_rate == -1)
goto not_negotiated;
}
/* 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 at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
JBUF_LOCK_CHECK (priv, out_flushing);
/* don't accept more data on EOS */
if (priv->eos)
goto have_eos;
/* let's check if this buffer is too late, we can only accept packets with
* bigger seqnum than the one we last pushed. */
if (priv->last_popped_seqnum != -1) {
gint gap;
gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
if (gap <= 0) {
/* priv->last_popped_seqnum >= seqnum, this packet is too late or the
* sender might have been restarted with different seqnum. */
if (gap < -100) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
} else {
goto too_late;
}
} else {
/* priv->last_popped_seqnum < seqnum, this is a new packet */
if (gap > 3000) {
GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
gap);
priv->last_popped_seqnum = -1;
priv->next_seqnum = -1;
}
}
}
/* let's drop oldest packet if the queue is already full and drop-on-latency
* is set. We can only do this when there actually is a latency. When no
* latency is set, we just pump it in the queue and let the other end push it
* out as fast as possible. */
if (priv->latency_ms && priv->drop_on_latency) {
latency_ts =
gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
if (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts) {
GstBuffer *old_buf;
GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
seqnum);
old_buf = rtp_jitter_buffer_pop (priv->jbuf);
gst_buffer_unref (old_buf);
}
}
/* 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, timestamp,
priv->clock_rate, &tail))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
if (priv->waiting)
JBUF_SIGNAL (priv);
/* let's unschedule and unblock any waiting buffers. We only want to do this
* when the tail buffer changed */
if (priv->clock_id && tail) {
GST_DEBUG_OBJECT (jitterbuffer,
"Unscheduling waiting buffer, new tail buffer");
gst_clock_id_unschedule (priv->clock_id);
}
GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets",
seqnum, rtp_jitter_buffer_num_packets (priv->jbuf));
finished:
JBUF_UNLOCK (priv);
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
not_negotiated:
{
GST_WARNING_OBJECT (jitterbuffer, "No clock-rate in caps!");
gst_buffer_unref (buffer);
gst_object_unref (jitterbuffer);
return GST_FLOW_OK;
}
out_flushing:
{
ret = priv->srcresult;
GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
gst_buffer_unref (buffer);
goto finished;
}
have_eos:
{
ret = GST_FLOW_UNEXPECTED;
GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
gst_buffer_unref (buffer);
goto finished;
}
too_late:
{
GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
" popped, dropping", seqnum, priv->last_popped_seqnum);
priv->num_late++;
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
seqnum);
priv->num_duplicates++;
gst_buffer_unref (buffer);
goto finished;
}
}
/**
* This funcion will push out buffers on the source pad.
*
* For each pushed buffer, the seqnum is recorded, if the next buffer B has a
* different seqnum (missing packets before B), this function will wait for the
* missing packet to arrive up to the timestamp of buffer B.
*/
static void
gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn result;
guint16 seqnum;
guint32 next_seqnum;
GstClockTime timestamp, out_time;
gboolean discont = FALSE;
gint gap;
priv = jitterbuffer->priv;
JBUF_LOCK_CHECK (priv, flushing);
again:
GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
while (TRUE) {
/* always wait if we are blocked */
if (!priv->blocked) {
/* if we have a packet, we can exit the loop and grab it */
if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
break;
/* no packets but we are EOS, do eos logic */
if (priv->eos)
goto do_eos;
}
/* underrun, wait for packets or flushing now */
priv->waiting = TRUE;
JBUF_WAIT_CHECK (priv, flushing);
priv->waiting = FALSE;
}
/* peek a buffer, we're just looking at the timestamp and the sequence number.
* If all is fine, we'll pop and push it. If the sequence number is wrong we
* wait on the timestamp. In the chain function we will unlock the wait when a
* new buffer is available. The peeked buffer is valid for as long as we hold
* the jitterbuffer lock. */
outbuf = rtp_jitter_buffer_peek (priv->jbuf);
/* get the seqnum and the next expected seqnum */
seqnum = gst_rtp_buffer_get_seq (outbuf);
next_seqnum = priv->next_seqnum;
/* get the timestamp, this is already corrected for clock skew by the
* jitterbuffer */
timestamp = GST_BUFFER_TIMESTAMP (outbuf);
GST_DEBUG_OBJECT (jitterbuffer,
"Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
rtp_jitter_buffer_num_packets (priv->jbuf));
/* apply our timestamp offset to the incomming buffer, this will be our output
* timestamp. */
out_time = apply_offset (jitterbuffer, timestamp);
/* get the gap between this and the previous packet. If we don't know the
* previous packet seqnum assume no gap. */
if (next_seqnum != -1) {
gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
/* if we have a packet that we already pushed or considered dropped, pop it
* off and get the next packet */
if (gap < 0) {
GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
seqnum, next_seqnum);
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
gst_buffer_unref (outbuf);
goto again;
}
} else {
GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
gap = -1;
}
/* 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
* earlier buffer before this buffer then.
* If we know the expected seqnum, we can compare it to the current seqnum to
* determine if we have missing a packet. If we have a missing packet (which
* must be before this packet) we can wait for it until the deadline for this
* packet expires. */
if (gap != 0 && out_time != -1) {
GstClockID id;
GstClockTime sync_time;
GstClockReturn ret;
GstClock *clock;
GstClockTime duration = GST_CLOCK_TIME_NONE;
if (gap > 0) {
/* we have a gap */
GST_WARNING_OBJECT (jitterbuffer,
"Sequence number GAP detected: expected %d instead of %d (%d missing)",
next_seqnum, seqnum, gap);
if (priv->last_out_time != -1) {
GST_DEBUG_OBJECT (jitterbuffer,
"out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
/* interpolate between the current time and the last time based on
* number of packets we are missing, this is the estimated duration
* for the missing packet based on equidistant packet spacing. Also make
* sure we never go negative. */
if (out_time > priv->last_out_time)
duration = (out_time - priv->last_out_time) / (gap + 1);
else
goto lost;
GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
/* add this duration to the timestamp of the last packet we pushed */
out_time = (priv->last_out_time + duration);
}
} else {
/* we don't know what the next_seqnum should be, wait for the last
* possible moment to push this buffer, maybe we get an earlier seqnum
* while we wait */
GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
}
GST_OBJECT_LOCK (jitterbuffer);
clock = GST_ELEMENT_CLOCK (jitterbuffer);
if (!clock) {
GST_OBJECT_UNLOCK (jitterbuffer);
/* let's just push if there is no clock */
goto push_buffer;
}
GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (out_time));
/* prepare for sync against clock */
sync_time = out_time + GST_ELEMENT_CAST (jitterbuffer)->base_time;
/* add latency, this includes our own latency and the peer latency. */
sync_time += (priv->latency_ms * GST_MSECOND);
sync_time += priv->peer_latency;
/* create an entry for the clock */
id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
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, NULL);
JBUF_LOCK (priv);
/* and free the entry */
gst_clock_id_unref (id);
priv->clock_id = NULL;
/* at this point, the clock could have been unlocked by a timeout, a new
* tail element was added to the queue or because we are shutting down. Check
* for shutdown first. */
if (priv->srcresult != GST_FLOW_OK)
goto flushing;
/* if we got unscheduled and we are not flushing, it's because a new tail
* element became available in the queue. Grab it and try to push or sync. */
if (ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (jitterbuffer,
"Wait got unscheduled, will retry to push with new buffer");
goto again;
}
lost:
/* we now timed out, this means we lost a packet or finished synchronizing
* on the first buffer. */
if (gap > 0) {
GstEvent *event;
/* we had a gap and thus we lost a packet. Create an event for this. */
GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
priv->num_late++;
discont = TRUE;
if (priv->do_lost) {
/* create paket lost event */
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
gst_structure_new ("GstRTPPacketLost",
"seqnum", G_TYPE_UINT, (guint) next_seqnum,
"timestamp", G_TYPE_UINT64, out_time,
"duration", G_TYPE_UINT64, duration, NULL));
gst_pad_push_event (priv->srcpad, event);
}
/* update our expected next packet */
priv->last_popped_seqnum = next_seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (next_seqnum + 1) & 0xffff;
/* look for next packet */
goto again;
}
/* there was no known gap,just the first packet, exit the loop and push */
GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
/* get new timestamp, latency might have changed */
out_time = apply_offset (jitterbuffer, timestamp);
}
push_buffer:
/* when we get here we are ready to pop and push the buffer */
outbuf = rtp_jitter_buffer_pop (priv->jbuf);
if (discont || priv->discont) {
/* set DISCONT flag when we missed a packet. */
outbuf = gst_buffer_make_metadata_writable (outbuf);
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
priv->discont = FALSE;
}
/* apply timestamp with offset to buffer now */
GST_BUFFER_TIMESTAMP (outbuf) = out_time;
/* 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. */
priv->last_popped_seqnum = seqnum;
priv->last_out_time = out_time;
priv->next_seqnum = (seqnum + 1) & 0xffff;
JBUF_UNLOCK (priv);
/* push buffer */
GST_DEBUG_OBJECT (jitterbuffer,
"Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (out_time));
result = gst_pad_push (priv->srcpad, outbuf);
if (result != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
do_eos:
{
/* store result, we are flushing now */
GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
priv->srcresult = GST_FLOW_UNEXPECTED;
gst_pad_pause_task (priv->srcpad);
gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
JBUF_UNLOCK (priv);
return;
}
flushing:
{
GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
pause:
{
const gchar *reason = gst_flow_get_name (result);
GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
JBUF_LOCK (priv);
/* store result */
priv->srcresult = result;
/* we don't post errors or anything because upstream will do that for us
* when we pass the return value upstream. */
gst_pad_pause_task (priv->srcpad);
JBUF_UNLOCK (priv);
return;
}
}
/* takes ownership of the input buffer */
static GstFlowReturn
gst_rtp_base_depayload_handle_buffer (GstRTPBaseDepayload * filter,
GstRTPBaseDepayloadClass * bclass, GstBuffer * in)
{
GstBuffer *(*process_rtp_packet_func) (GstRTPBaseDepayload * base,
GstRTPBuffer * rtp_buffer);
GstBuffer *(*process_func) (GstRTPBaseDepayload * base, GstBuffer * in);
GstRTPBaseDepayloadPrivate *priv;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint32 ssrc;
guint16 seqnum;
guint32 rtptime;
gboolean discont, buf_discont;
gint gap;
GstRTPBuffer rtp = { NULL };
priv = filter->priv;
process_func = bclass->process;
process_rtp_packet_func = bclass->process_rtp_packet;
/* we must have a setcaps first */
if (G_UNLIKELY (!priv->negotiated))
goto not_negotiated;
if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
goto invalid_buffer;
buf_discont = GST_BUFFER_IS_DISCONT (in);
priv->pts = GST_BUFFER_PTS (in);
priv->dts = GST_BUFFER_DTS (in);
priv->duration = GST_BUFFER_DURATION (in);
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
seqnum = gst_rtp_buffer_get_seq (&rtp);
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
priv->last_seqnum = seqnum;
priv->last_rtptime = rtptime;
discont = buf_discont;
GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, pts %"
GST_TIME_FORMAT ", dts %" GST_TIME_FORMAT, buf_discont, seqnum, rtptime,
GST_TIME_ARGS (priv->pts), GST_TIME_ARGS (priv->dts));
/* Check seqnum. This is a very simple check that makes sure that the seqnums
* are strictly increasing, dropping anything that is out of the ordinary. We
* can only do this when the next_seqnum is known. */
if (G_LIKELY (priv->next_seqnum != -1)) {
if (ssrc != priv->last_ssrc) {
GST_LOG_OBJECT (filter,
"New ssrc %u (current ssrc %u), sender restarted",
ssrc, priv->last_ssrc);
discont = TRUE;
} else {
gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
/* if we have no gap, all is fine */
if (G_UNLIKELY (gap != 0)) {
GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
priv->next_seqnum, gap);
if (gap < 0) {
/* seqnum > next_seqnum, we are missing some packets, this is always a
* DISCONT. */
GST_LOG_OBJECT (filter, "%d missing packets", gap);
discont = TRUE;
} else {
/* seqnum < next_seqnum, we have seen this packet before, have a
* reordered packet or the sender could be restarted. If the packet
* is not too old, we throw it away as a duplicate. Otherwise we
* mark discont and continue assuming the sender has restarted. See
* also RFC 4737. */
GST_WARNING ("gap %d <= priv->max_reorder %d -> dropping %d",
gap, priv->max_reorder, gap <= priv->max_reorder);
if (gap <= priv->max_reorder)
goto dropping;
GST_LOG_OBJECT (filter,
"%d > %d, packet too old, sender likely restarted", gap,
priv->max_reorder);
discont = TRUE;
}
}
}
}
priv->next_seqnum = (seqnum + 1) & 0xffff;
priv->last_ssrc = ssrc;
if (G_UNLIKELY (discont)) {
priv->discont = TRUE;
if (!buf_discont) {
gpointer old_inbuf = in;
/* we detected a seqnum discont but the buffer was not flagged with a discont,
* set the discont flag so that the subclass can throw away old data. */
GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
in = gst_buffer_make_writable (in);
GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
/* depayloaders will check flag on rtpbuffer->buffer, so if the input
* buffer was not writable already we need to remap to make our
* newly-flagged buffer current on the rtpbuffer */
if (in != old_inbuf) {
gst_rtp_buffer_unmap (&rtp);
if (G_UNLIKELY (!gst_rtp_buffer_map (in, GST_MAP_READ, &rtp)))
goto invalid_buffer;
}
}
}
/* prepare segment event if needed */
if (filter->need_newsegment) {
priv->segment_event = create_segment_event (filter, rtptime,
GST_BUFFER_PTS (in));
filter->need_newsegment = FALSE;
}
priv->input_buffer = in;
if (process_rtp_packet_func != NULL) {
out_buf = process_rtp_packet_func (filter, &rtp);
gst_rtp_buffer_unmap (&rtp);
} else if (process_func != NULL) {
gst_rtp_buffer_unmap (&rtp);
out_buf = process_func (filter, in);
} else {
goto no_process;
}
/* let's send it out to processing */
if (out_buf) {
ret = gst_rtp_base_depayload_push (filter, out_buf);
}
gst_buffer_unref (in);
priv->input_buffer = NULL;
return ret;
/* ERRORS */
not_negotiated:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
("No RTP format was negotiated."),
("Input buffers need to have RTP caps set on them. This is usually "
"achieved by setting the 'caps' property of the upstream source "
"element (often udpsrc or appsrc), or by putting a capsfilter "
"element before the depayloader and setting the 'caps' property "
"on that. Also see http://cgit.freedesktop.org/gstreamer/"
"gst-plugins-good/tree/gst/rtp/README"));
gst_buffer_unref (in);
return GST_FLOW_NOT_NEGOTIATED;
}
invalid_buffer:
{
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
("Received invalid RTP payload, dropping"));
gst_buffer_unref (in);
return GST_FLOW_OK;
}
dropping:
{
gst_rtp_buffer_unmap (&rtp);
GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
gst_buffer_unref (in);
return GST_FLOW_OK;
}
no_process:
{
gst_rtp_buffer_unmap (&rtp);
/* this is not fatal but should be filtered earlier */
GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
("The subclass does not have a process or process_rtp_packet method"));
gst_buffer_unref (in);
return GST_FLOW_ERROR;
}
}
/**
* 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;
}
}
/**
* 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;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (buf != NULL, FALSE);
seqnum = gst_rtp_buffer_get_seq (buf);
/* loop the list to skip strictly smaller seqnum buffers */
for (list = jbuf->packets->head; list; list = g_list_next (list)) {
guint16 qseq;
gint gap;
qseq = gst_rtp_buffer_get_seq (GST_BUFFER_CAST (list->data));
/* 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 (buf);
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);
return TRUE;
/* ERRORS */
duplicate:
{
GST_WARNING ("duplicate packet %d found", (gint) seqnum);
return FALSE;
}
}
