static gboolean
gst_rdt_manager_src_activate_mode (GstPad * pad, GstObject * parent,
GstPadMode mode, gboolean active)
{
gboolean result;
GstRDTManager *rdtmanager;
GstRDTManagerSession *session;
session = gst_pad_get_element_private (pad);
rdtmanager = session->dec;
switch (mode) {
case GST_PAD_MODE_PUSH:
if (active) {
/* allow data processing */
JBUF_LOCK (session);
GST_DEBUG_OBJECT (rdtmanager, "Enabling pop on queue");
/* Mark as non flushing */
session->srcresult = GST_FLOW_OK;
gst_segment_init (&session->segment, GST_FORMAT_TIME);
session->last_popped_seqnum = -1;
session->last_out_time = -1;
session->next_seqnum = -1;
session->eos = FALSE;
JBUF_UNLOCK (session);
/* start pushing out buffers */
GST_DEBUG_OBJECT (rdtmanager, "Starting task on srcpad");
result =
gst_pad_start_task (pad, (GstTaskFunction) gst_rdt_manager_loop,
pad, NULL);
} else {
/* make sure all data processing stops ASAP */
JBUF_LOCK (session);
/* mark ourselves as flushing */
session->srcresult = GST_FLOW_FLUSHING;
GST_DEBUG_OBJECT (rdtmanager, "Disabling pop on queue");
/* this unblocks any waiting pops on the src pad task */
JBUF_SIGNAL (session);
/* unlock clock, we just unschedule, the entry will be released by
* the locking streaming thread. */
if (session->clock_id)
gst_clock_id_unschedule (session->clock_id);
JBUF_UNLOCK (session);
/* NOTE this will hardlock if the state change is called from the src pad
* task thread because we will _join() the thread. */
GST_DEBUG_OBJECT (rdtmanager, "Stopping task on srcpad");
result = gst_pad_stop_task (pad);
}
break;
default:
result = FALSE;
break;
}
return result;
}
static GstFlowReturn
gst_rdt_manager_handle_data_packet (GstRDTManagerSession * session,
GstClockTime timestamp, GstRDTPacket * packet)
{
GstRDTManager *rdtmanager;
guint16 seqnum;
gboolean tail;
GstFlowReturn res;
GstBuffer *buffer;
rdtmanager = session->dec;
res = GST_FLOW_OK;
seqnum = 0;
GST_DEBUG_OBJECT (rdtmanager,
"Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
GST_TIME_ARGS (timestamp));
buffer = gst_rdt_packet_to_buffer (packet);
JBUF_LOCK_CHECK (session, out_flushing);
/* insert the packet into the queue now, FIXME, use seqnum */
if (!rdt_jitter_buffer_insert (session->jbuf, buffer, timestamp,
session->clock_rate, &tail))
goto duplicate;
/* signal addition of new buffer when the _loop is waiting. */
if (session->waiting)
JBUF_SIGNAL (session);
finished:
JBUF_UNLOCK (session);
return res;
/* ERRORS */
out_flushing:
{
res = session->srcresult;
GST_DEBUG_OBJECT (rdtmanager, "flushing %s", gst_flow_get_name (res));
gst_buffer_unref (buffer);
goto finished;
}
duplicate:
{
GST_WARNING_OBJECT (rdtmanager, "Duplicate packet #%d detected, dropping",
seqnum);
session->num_duplicates++;
gst_buffer_unref (buffer);
goto finished;
}
}
static void
gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
{
GstRtpJitterBufferPrivate *priv;
priv = jitterbuffer->priv;
JBUF_LOCK (priv);
/* mark ourselves as flushing */
priv->srcresult = GST_FLOW_WRONG_STATE;
GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
/* this unblocks any waiting pops on the src pad task */
JBUF_SIGNAL (priv);
/* unlock clock, we just unschedule, the entry will be released by the
* locking streaming thread. */
if (priv->clock_id)
gst_clock_id_unschedule (priv->clock_id);
JBUF_UNLOCK (priv);
}
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;
}
}
static gboolean
gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
{
gboolean ret = TRUE;
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
priv = jitterbuffer->priv;
GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* we need time for now */
if (format != GST_FORMAT_TIME)
goto newseg_wrong_format;
GST_DEBUG_OBJECT (jitterbuffer,
"newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
GST_TIME_ARGS (time));
/* now configure the values, we need these to time the release of the
* buffers on the srcpad. */
gst_segment_set_newsegment_full (&priv->segment, update,
rate, arate, format, start, stop, time);
/* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
ret = gst_pad_push_event (priv->srcpad, event);
break;
}
case GST_EVENT_FLUSH_START:
gst_rtp_jitter_buffer_flush_start (jitterbuffer);
ret = gst_pad_push_event (priv->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
ret = gst_pad_push_event (priv->srcpad, event);
ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE);
break;
case GST_EVENT_EOS:
{
/* push EOS in queue. We always push it at the head */
JBUF_LOCK (priv);
/* check for flushing, we need to discard the event and return FALSE when
* we are flushing */
ret = priv->srcresult == GST_FLOW_OK;
if (ret && !priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "queuing EOS");
priv->eos = TRUE;
JBUF_SIGNAL (priv);
} else if (priv->eos) {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
} else {
GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
gst_flow_get_name (priv->srcresult));
}
JBUF_UNLOCK (priv);
gst_event_unref (event);
break;
}
default:
ret = gst_pad_push_event (priv->srcpad, event);
break;
}
done:
gst_object_unref (jitterbuffer);
return ret;
/* ERRORS */
newseg_wrong_format:
{
GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
ret = FALSE;
goto done;
}
}
static GstStateChangeReturn
gst_rtp_jitter_buffer_change_state (GstElement * element,
GstStateChange transition)
{
GstRtpJitterBuffer *jitterbuffer;
GstRtpJitterBufferPrivate *priv;
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
jitterbuffer = GST_RTP_JITTER_BUFFER (element);
priv = jitterbuffer->priv;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
JBUF_LOCK (priv);
/* reset negotiated values */
priv->clock_rate = -1;
priv->clock_base = -1;
priv->peer_latency = 0;
priv->last_pt = -1;
/* block until we go to PLAYING */
priv->blocked = TRUE;
/* reset skew detection initialy */
rtp_jitter_buffer_reset_skew (priv->jbuf);
JBUF_UNLOCK (priv);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
JBUF_LOCK (priv);
/* unblock to allow streaming in PLAYING */
priv->blocked = FALSE;
JBUF_SIGNAL (priv);
JBUF_UNLOCK (priv);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
/* we are a live element because we sync to the clock, which we can only
* do in the PLAYING state */
if (ret != GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
JBUF_LOCK (priv);
/* block to stop streaming when PAUSED */
priv->blocked = TRUE;
JBUF_UNLOCK (priv);
if (ret != GST_STATE_CHANGE_FAILURE)
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
