mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-04-12 19:14:16 +00:00
session: Rearrange RTCP reporting a little
Make a function to generate an RTCP packet for a source, pass the source as a parameter. Move timeout of collisions to session cleanup phase.
This commit is contained in:
Wim Taymans
parent
a3bf374351
commit
2163355a47
1 changed files with 85 additions and 81 deletions
|
|
@ -2649,6 +2649,7 @@ typedef struct
|
|||
{
|
||||
GstRTCPBuffer rtcpbuf;
|
||||
RTPSession *sess;
|
||||
RTPSource *source;
|
||||
GstBuffer *rtcp;
|
||||
GstClockTime current_time;
|
||||
guint64 ntpnstime;
|
||||
|
|
@ -2659,16 +2660,19 @@ typedef struct
|
|||
gboolean has_sdes;
|
||||
gboolean is_early;
|
||||
gboolean may_suppress;
|
||||
gboolean notify;
|
||||
} ReportData;
|
||||
|
||||
static void
|
||||
session_start_rtcp (RTPSession * sess, ReportData * data)
|
||||
{
|
||||
GstRTCPPacket *packet = &data->packet;
|
||||
RTPSource *own = sess->source;
|
||||
RTPSource *own = data->source;
|
||||
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
||||
|
||||
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
|
||||
data->is_bye = FALSE;
|
||||
data->has_sdes = FALSE;
|
||||
|
||||
gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
|
||||
|
||||
|
|
@ -2703,19 +2707,11 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
|
|||
static void
|
||||
session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
|
||||
{
|
||||
RTPSession *sess = data->sess;
|
||||
GstRTCPPacket *packet = &data->packet;
|
||||
|
||||
/* create a new buffer if needed */
|
||||
if (data->rtcp == NULL) {
|
||||
session_start_rtcp (sess, data);
|
||||
} else if (data->is_early) {
|
||||
/* Put a single RR or SR in minimal compound packets */
|
||||
return;
|
||||
}
|
||||
if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
|
||||
/* only report about other sender sources */
|
||||
if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
|
||||
if (source != data->source && RTP_SOURCE_IS_SENDER (source)) {
|
||||
guint8 fractionlost;
|
||||
gint32 packetslost;
|
||||
guint32 exthighestseq, jitter;
|
||||
|
|
@ -2753,13 +2749,22 @@ session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
|
|||
GstClockTime interval, binterval;
|
||||
GstClockTime btime;
|
||||
|
||||
is_sender = RTP_SOURCE_IS_SENDER (source);
|
||||
is_active = RTP_SOURCE_IS_ACTIVE (source);
|
||||
/* check for outdated collisions */
|
||||
if (source->internal) {
|
||||
GST_DEBUG ("Timing out collisions");
|
||||
rtp_source_timeout (source, data->current_time,
|
||||
/* "a relatively long time" -- RFC 3550 section 8.2 */
|
||||
RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
|
||||
data->running_time - sess->rtcp_feedback_retention_window);
|
||||
}
|
||||
|
||||
/* nothing to do when without RTCP */
|
||||
/* nothing else to do when without RTCP */
|
||||
if (data->interval == GST_CLOCK_TIME_NONE)
|
||||
return;
|
||||
|
||||
is_sender = RTP_SOURCE_IS_SENDER (source);
|
||||
is_active = RTP_SOURCE_IS_ACTIVE (source);
|
||||
|
||||
/* our own rtcp interval may have been forced low by secondary configuration,
|
||||
* while sender side may still operate with higher interval,
|
||||
* so do not just take our interval to decide on timing out sender,
|
||||
|
|
@ -2856,9 +2861,9 @@ session_sdes (RTPSession * sess, ReportData * data)
|
|||
/* add SDES packet */
|
||||
gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
|
||||
|
||||
gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
|
||||
gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
|
||||
|
||||
sdes = rtp_source_get_sdes_struct (sess->source);
|
||||
sdes = rtp_source_get_sdes_struct (data->source);
|
||||
|
||||
/* add all fields in the structure, the order is not important. */
|
||||
n_fields = gst_structure_n_fields (sdes);
|
||||
|
|
@ -2912,18 +2917,13 @@ session_sdes (RTPSession * sess, ReportData * data)
|
|||
|
||||
/* schedule a BYE packet */
|
||||
static void
|
||||
session_bye (RTPSession * sess, ReportData * data)
|
||||
make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
|
||||
{
|
||||
GstRTCPPacket *packet = &data->packet;
|
||||
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
||||
RTPSource *source = sess->source;
|
||||
|
||||
/* open packet */
|
||||
session_start_rtcp (sess, data);
|
||||
|
||||
/* add SDES */
|
||||
session_sdes (sess, data);
|
||||
|
||||
/* add a BYE packet */
|
||||
gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
|
||||
gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
|
||||
|
|
@ -2932,6 +2932,7 @@ session_bye (RTPSession * sess, ReportData * data)
|
|||
|
||||
/* we have a BYE packet now */
|
||||
data->is_bye = TRUE;
|
||||
source->sent_bye = TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
|
|
@ -3027,6 +3028,51 @@ remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
|
|||
return source->closing;
|
||||
}
|
||||
|
||||
static void
|
||||
generate_rtcp (RTPSource * source, ReportData * data)
|
||||
{
|
||||
RTPSession *sess = data->sess;
|
||||
|
||||
/* only generate RTCP for active internal sources */
|
||||
if (!source->internal || source->sent_bye)
|
||||
return;
|
||||
|
||||
data->source = source;
|
||||
|
||||
/* open packet */
|
||||
session_start_rtcp (sess, data);
|
||||
|
||||
if (source->marked_bye) {
|
||||
/* send BYE */
|
||||
make_source_bye (sess, source, data);
|
||||
} else if (!data->is_early) {
|
||||
/* loop over all known sources and add report blocks. If we are ealy, we
|
||||
* just make a minimal RTCP packet and skip this step */
|
||||
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
||||
(GHFunc) session_report_blocks, data);
|
||||
}
|
||||
if (!data->has_sdes)
|
||||
session_sdes (sess, data);
|
||||
|
||||
gst_rtcp_buffer_unmap (&data->rtcpbuf);
|
||||
|
||||
if (sess->change_ssrc) {
|
||||
GST_DEBUG ("need to change our SSRC (%08x)", source->ssrc);
|
||||
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
|
||||
GINT_TO_POINTER (source->ssrc));
|
||||
|
||||
source->ssrc = rtp_session_create_new_ssrc (sess);
|
||||
rtp_source_reset (source);
|
||||
|
||||
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
|
||||
GINT_TO_POINTER (source->ssrc), source);
|
||||
|
||||
sess->change_ssrc = FALSE;
|
||||
data->notify = TRUE;
|
||||
GST_DEBUG ("changed our SSRC to %08x", source->ssrc);
|
||||
}
|
||||
}
|
||||
|
||||
/**
|
||||
* rtp_session_on_timeout:
|
||||
* @sess: an #RTPSession
|
||||
|
|
@ -3053,7 +3099,6 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
|
|||
ReportData data = { GST_RTCP_BUFFER_INIT };
|
||||
RTPSource *own;
|
||||
GHashTable *table_copy;
|
||||
gboolean notify = FALSE;
|
||||
|
||||
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
|
||||
|
||||
|
|
@ -3062,13 +3107,11 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
|
|||
GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
|
||||
|
||||
data.sess = sess;
|
||||
data.rtcp = NULL;
|
||||
data.current_time = current_time;
|
||||
data.ntpnstime = ntpnstime;
|
||||
data.is_bye = FALSE;
|
||||
data.has_sdes = FALSE;
|
||||
data.may_suppress = FALSE;
|
||||
data.running_time = running_time;
|
||||
data.may_suppress = FALSE;
|
||||
data.notify = FALSE;
|
||||
|
||||
own = sess->source;
|
||||
|
||||
|
|
@ -3093,88 +3136,49 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
|
|||
(GHRFunc) remove_closing_sources, NULL);
|
||||
|
||||
/* see if we need to generate SR or RR packets */
|
||||
if (is_rtcp_time (sess, current_time, &data)) {
|
||||
if (own->marked_bye) {
|
||||
/* generate BYE instead */
|
||||
GST_DEBUG ("generating BYE message");
|
||||
session_bye (sess, &data);
|
||||
own->sent_bye = TRUE;
|
||||
} else {
|
||||
/* loop over all known sources and do something */
|
||||
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
||||
(GHFunc) session_report_blocks, &data);
|
||||
}
|
||||
}
|
||||
if (!is_rtcp_time (sess, current_time, &data))
|
||||
goto done;
|
||||
|
||||
if (data.rtcp) {
|
||||
/* we keep track of the last report time in order to timeout inactive
|
||||
* receivers or senders */
|
||||
if (!data.is_early && !data.may_suppress)
|
||||
sess->last_rtcp_send_time = data.current_time;
|
||||
sess->first_rtcp = FALSE;
|
||||
sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
|
||||
generate_rtcp (own, &data);
|
||||
|
||||
/* add SDES for this source when not already added */
|
||||
if (!data.has_sdes)
|
||||
session_sdes (sess, &data);
|
||||
}
|
||||
|
||||
/* check for outdated collisions */
|
||||
GST_DEBUG ("Timing out collisions");
|
||||
rtp_source_timeout (sess->source, current_time,
|
||||
/* "a relatively long time" -- RFC 3550 section 8.2 */
|
||||
RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
|
||||
running_time - sess->rtcp_feedback_retention_window);
|
||||
|
||||
if (sess->change_ssrc) {
|
||||
GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
|
||||
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
|
||||
GINT_TO_POINTER (own->ssrc));
|
||||
|
||||
own->ssrc = rtp_session_create_new_ssrc (sess);
|
||||
rtp_source_reset (own);
|
||||
|
||||
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
|
||||
GINT_TO_POINTER (own->ssrc), own);
|
||||
|
||||
sess->change_ssrc = FALSE;
|
||||
notify = TRUE;
|
||||
GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
|
||||
}
|
||||
|
||||
sess->allow_early = TRUE;
|
||||
/* we keep track of the last report time in order to timeout inactive
|
||||
* receivers or senders */
|
||||
if (!data.is_early && !data.may_suppress)
|
||||
sess->last_rtcp_send_time = data.current_time;
|
||||
sess->first_rtcp = FALSE;
|
||||
sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
|
||||
|
||||
done:
|
||||
RTP_SESSION_UNLOCK (sess);
|
||||
|
||||
if (notify)
|
||||
if (data.notify)
|
||||
g_object_notify (G_OBJECT (sess), "internal-ssrc");
|
||||
|
||||
/* push out the RTCP packet */
|
||||
if (data.rtcp) {
|
||||
gboolean do_not_suppress;
|
||||
|
||||
gst_rtcp_buffer_unmap (&data.rtcpbuf);
|
||||
GstBuffer *buffer = data.rtcp;
|
||||
|
||||
/* Give the user a change to add its own packet */
|
||||
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
|
||||
data.rtcp, data.is_early, &do_not_suppress);
|
||||
buffer, data.is_early, &do_not_suppress);
|
||||
|
||||
if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
|
||||
guint packet_size;
|
||||
|
||||
packet_size = gst_buffer_get_size (data.rtcp) + sess->header_len;
|
||||
packet_size = gst_buffer_get_size (buffer) + sess->header_len;
|
||||
|
||||
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
|
||||
GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
|
||||
sess->stats.avg_rtcp_packet_size, packet_size);
|
||||
result =
|
||||
sess->callbacks.send_rtcp (sess, own, data.rtcp, own->sent_bye,
|
||||
sess->callbacks.send_rtcp (sess, own, buffer, data.is_bye,
|
||||
sess->send_rtcp_user_data);
|
||||
} else {
|
||||
GST_DEBUG ("freeing packet callback: %p"
|
||||
" do_not_suppress: %d may_suppress: %d",
|
||||
sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
|
||||
gst_buffer_unref (data.rtcp);
|
||||
gst_buffer_unref (buffer);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue