/**
* @brief Resume (or start) an RTP session
*
* @param session_gen The session to resume or start
* @param range_gen Pointer tp @ref RTSP_Range to start with
*
* @todo This function should probably take care of starting eventual
* libev events when the scheduler is replaced.
*
* This function is used by the PLAY method of RTSP to start or resume
* a session; since a newly-created session starts as paused, this is
* the only method available.
*
* The use of a pointer to double rather than a double itself is to
* make it possible to pass this function straight to foreach
* functions from glib.
*
* @internal This function should only be called from g_slist_foreach.
*/
static void rtp_session_resume(gpointer session_gen, gpointer range_gen) {
RTP_session *session = (RTP_session*)session_gen;
RTSP_Range *range = (RTSP_Range*)range_gen;
fnc_log(FNC_LOG_VERBOSE, "Resuming session %p\n", session);
session->range = range;
session->start_seq = 1 + session->seq_no;
session->start_rtptime = g_random_int();
session->send_time = 0.0;
session->last_packet_send_time = time(NULL);
/* Create the new thread pool for the read requests */
session->fill_pool = g_thread_pool_new(rtp_session_fill_cb, session,
1, true, NULL);
/* Prefetch frames */
rtp_session_fill(session);
ev_periodic_set(&session->transport.rtp_writer,
range->playback_time - 0.05,
0, NULL);
ev_periodic_start(session->srv->loop, &session->transport.rtp_writer);
ev_io_start(session->srv->loop, &session->transport.rtcp_reader);
}
static __inline__ void
rtp_session_resume_stored(RTP_session *rtp_s)
{
GEvent *periodic;
if (rtp_s->playing)
return;
if (rtp_s->track && rtp_s->track->properties.media_type != MP_video)
{//只创建一路audio是一个bug.
rtp_s->track->parent->audio_rtp = rtp_s;
return;
}
if (!rtp_s->fill_pool)
{
rtp_s->fill_pool = g_thread_pool_new(rtp_session_fill_cb,
rtp_s, 1, TRUE, NULL);
rtp_session_fill(rtp_s);
}
if (!rtp_s->transport.rtp_writer)
{
periodic = g_event_new(sizeof(RTP_writer), -1, 0);
g_event_set_timeout(periodic, 1);
g_event_set_callback(periodic, rtp_timer_cb, rtp_s, on_rtp_periodic_destroy);
rtp_s->transport.rtp_writer = periodic;
((RTP_writer*)periodic)->rtp_s = rtp_s;
rtp_session_ref(rtp_s);
g_scheduler_add(periodic, LOOP_WEIGHT_VIDEO);
}
rtp_s->playing = TRUE;
}
/**
* Send pending RTP packets to a session.
*
* @param loop eventloop
* @param w contains the session the RTP session for which to send the packets
* @todo implement a saner ratecontrol
*/
static void rtp_write_cb(struct ev_loop *loop, ev_periodic *w,
ATTR_UNUSED int revents)
{
RTP_session *session = w->data;
Resource *resource = session->track->parent;
MParserBuffer *buffer = NULL;
ev_tstamp next_time = w->offset;
#ifdef HAVE_METADATA
if (session->metadata)
cpd_send(session, now);
#endif
/* If there is no buffer, it means that either the producer
* has been stopped (as we reached the end of stream) or that
* there is no data for the consumer to read. If that's the
* case we just give control back to the main loop for now.
*/
if ( bq_consumer_stopped(session->consumer) ) {
/* If the producer has been stopped, we send the
* finishing packets and go away.
*/
fnc_log(FNC_LOG_INFO, "[rtp] Stream Finished");
rtcp_send_sr(session, BYE);
return;
}
/* Check whether we have enough extra frames to send. If we have
* no extra frames we have a problem, since we're going to send
* one packet at least.
*/
if (resource->eor)
fnc_log(FNC_LOG_INFO,
"[%s] end of resource %d packets to be fetched",
session->track->properties.encoding_name,
bq_consumer_unseen(session->consumer));
/* Get the current buffer, if there is enough data */
if ( !(buffer = bq_consumer_get(session->consumer)) ) {
/* We wait a bit of time to get the data but before it is
* expired.
*/
if (resource->eor) {
fnc_log(FNC_LOG_INFO, "[rtp] Stream Finished");
rtcp_send_sr(session, BYE);
return;
}
next_time += 0.01; // assumed to be enough
} else {
MParserBuffer *next;
double delivery = buffer->delivery;
double timestamp = buffer->timestamp;
double duration = buffer->duration;
gboolean marker = buffer->marker;
rtp_packet_send(session, buffer);
if (session->pkt_count % 29 == 1)
rtcp_send_sr(session, SDES);
if (bq_consumer_move(session->consumer)) {
next = bq_consumer_get(session->consumer);
if(delivery != next->delivery) {
if (session->track->properties.media_source == MS_live)
next_time += next->delivery - delivery;
else
next_time = session->range->playback_time -
session->range->begin_time +
next->delivery;
}
} else {
if (marker)
next_time += duration;
}
fnc_log(FNC_LOG_VERBOSE,
"[%s] Now: %5.4f, cur %5.4f[%5.4f][%5.4f], next %5.4f %s\n",
session->track->properties.encoding_name,
ev_now(loop) - session->range->playback_time,
delivery,
timestamp,
duration,
next_time - session->range->playback_time,
marker? "M" : " ");
}
ev_periodic_set(w, next_time, 0, NULL);
ev_periodic_again(loop, w);
rtp_session_fill(session);
}
static gboolean
rtp_timer_cb(GEvent *ev, gint revents, void *user_data)
{
RTP_session *session = (RTP_session*)user_data;
Resource *resource = session->track->parent;
MParserBuffer *buffer = NULL;
ev_tstamp now, sleep_secs;
gint to_send = MAX_SND_BLOCKSIZE;
now = g_event_time_now_sync(ev);
sleep_secs = .01;
#ifdef HAVE_METADATA
if (session->metadata)
cpd_send(session, now);
#endif
if (!session->audio_rtp && resource->audio_rtp)
{//Fixme, race against rtp_session_kill(resource->audio_rtp)
session->audio_rtp = rtp_session_ref(resource->audio_rtp);
}
for (;;)
{
if (to_send <= 0)
{
break;
}
if (bq_consumer_stopped(session->consumer))
{
fnc_log(FNC_LOG_INFO, "[rtp] Stream Finished");
rtcp_send_sr(session, BYE);
return FALSE;
}
if (rtp_session_would_block(session))
{
break;
}
if (!(buffer = bq_consumer_get(session->consumer)))
{
if (resource->eor)
{
fnc_log(FNC_LOG_INFO, "[rtp] Stream Finished");
rtp_packet_send_eof(session, (time_t)now);
rtcp_send_sr(session, BYE);
sleep_secs = 1;
break;
}
sleep_secs = .005;
break;
}
if (buffer->mtype != MP_video)
{
if (session->audio_rtp)
{
rtp_packet_send_rtp(session->audio_rtp, buffer, (time_t)now);
}
}
else
{
rtp_packet_send_rtp(session, buffer, (time_t)now);
}
to_send -= buffer->data_size;
if (!bq_consumer_move(session->consumer))
{
break;
}
}
g_event_mod_timer_sync(ev, sleep_secs);
rtp_session_fill(session);
return TRUE;
}
