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gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets.

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Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
This commit is contained in:
Wim Taymans 2008-11-20 18:41:34 +00:00
parent df25221378
commit da17b1b643
7 changed files with 246 additions and 60 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

33
ChangeLog
View file

@ -1,3 +1,36 @@
2008-11-20 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
2008-11-20 Stefan Kost <ensonic@users.sf.net>
patch by: Leandro Melo de Sales <leandroal@gmail.com>

43
gst/rtpmanager/gstrtpsession.c
View file

@ -860,12 +860,13 @@ gst_rtp_session_get_property (GObject * object, guint prop_id,
}
}
static guint64
get_current_ntp_ns_time (GstRtpSession * rtpsession)
static void
get_current_times (GstRtpSession * rtpsession,
GstClockTime * running_time, guint64 * ntpnstime)
{
guint64 ntpnstime;
guint64 ntpns;
GstClock *clock;
GstClockTime base_time, ntpnsbase;
GstClockTime base_time, ntpnsbase, rt;
GST_OBJECT_LOCK (rtpsession);
if ((clock = GST_ELEMENT_CLOCK (rtpsession))) {
@ -874,20 +875,21 @@ get_current_ntp_ns_time (GstRtpSession * rtpsession)
gst_object_ref (clock);
GST_OBJECT_UNLOCK (rtpsession);
/* get current NTP time */
ntpnstime = gst_clock_get_time (clock);
/* convert to running time */
ntpnstime -= base_time;
/* add NTP base offset */
ntpnstime += ntpnsbase;
/* get current clock time and convert to running time */
rt = gst_clock_get_time (clock) - base_time;
/* add NTP base offset to get NTP ns time */
ntpns = rt + ntpnsbase;
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (rtpsession);
ntpnstime = -1;
rt = -1;
ntpns = -1;
}
return ntpnstime;
if (running_time)
*running_time = rt;
if (ntpnstime)
*ntpnstime = ntpns;
}
static void
@ -935,7 +937,7 @@ rtcp_thread (GstRtpSession * rtpsession)
current_time = gst_clock_get_time (rtpsession->priv->sysclock);
/* get current NTP time */
ntpnstime = get_current_ntp_ns_time (rtpsession);
get_current_times (rtpsession, NULL, &ntpnstime);
/* we get unlocked because we need to perform reconsideration, don't perform
* the timeout but get a new reporting estimate. */
@ -1423,7 +1425,7 @@ gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstFlowReturn ret;
GstClockTime current_time;
GstClockTime current_time, running_time;
guint64 ntpnstime;
GstClockTime timestamp;
@ -1436,22 +1438,21 @@ gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer)
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment values */
ntpnstime =
running_time =
gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
timestamp);
/* add constant to convert running time to NTP time */
ntpnstime += priv->ntpnsbase;
ntpnstime = running_time + priv->ntpnsbase;
} else {
ntpnstime = get_current_ntp_ns_time (rtpsession);
get_current_times (rtpsession, &running_time, &ntpnstime);
}
current_time = gst_clock_get_time (priv->sysclock);
ret = rtp_session_process_rtp (priv->session, buffer, current_time,
ntpnstime);
running_time, ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
done:
gst_object_unref (rtpsession);

37
gst/rtpmanager/rtpsession.c
View file

@ -329,6 +329,7 @@ rtp_session_init (RTPSession * sess)
/* create an active SSRC for this session manager */
sess->source = rtp_session_create_source (sess);
sess->source->validated = TRUE;
sess->source->internal = TRUE;
sess->stats.active_sources++;
/* default UDP header length */
@ -1329,11 +1330,11 @@ rtp_session_create_source (RTPSession * sess)
static void
update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
guint64 ntpnstime)
GstClockTime running_time, guint64 ntpnstime)
{
/* get time of arrival */
arrival->time = current_time;
arrival->timestamp = GST_BUFFER_TIMESTAMP (buffer);
arrival->running_time = running_time;
arrival->ntpnstime = ntpnstime;
/* get packet size including header overhead */
@ -1368,7 +1369,7 @@ update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
*/
GstFlowReturn
rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
GstClockTime current_time, guint64 ntpnstime)
GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
{
GstFlowReturn result;
guint32 ssrc;
@ -1385,7 +1386,8 @@ rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
RTP_SESSION_LOCK (sess);
/* update arrival stats */
update_arrival_stats (sess, &arrival, TRUE, buffer, current_time, ntpnstime);
update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
running_time, ntpnstime);
/* ignore more RTP packets when we left the session */
if (sess->source->received_bye)
@ -1629,6 +1631,8 @@ rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
j++;
}
source->validated = TRUE;
if (created)
on_new_ssrc (sess, source);
if (changed)
@ -1759,7 +1763,7 @@ rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
RTP_SESSION_LOCK (sess);
/* update arrival stats */
update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1);
update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, -1);
if (sess->sent_bye)
goto ignore;
@ -2097,8 +2101,8 @@ session_start_rtcp (RTPSession * sess, ReportData * data)
rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
&packet_count, &octet_count);
/* store stats */
rtp_source_process_sr (own, data->ntpnstime, ntptime, rtptime, packet_count,
octet_count);
rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
packet_count, octet_count);
/* fill in sender report info */
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
@ -2331,6 +2335,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
GstFlowReturn result = GST_FLOW_OK;
GList *item;
ReportData data;
RTPSource *own;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
@ -2344,6 +2349,8 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
data.is_bye = FALSE;
data.has_sdes = FALSE;
own = sess->source;
RTP_SESSION_LOCK (sess);
/* get a new interval, we need this for various cleanups etc */
data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
@ -2354,7 +2361,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
/* see if we need to generate SR or RR packets */
if (is_rtcp_time (sess, current_time, &data)) {
if (sess->source->received_bye) {
if (own->received_bye) {
/* generate BYE instead */
GST_DEBUG ("generating BYE message");
session_bye (sess, &data);
@ -2401,21 +2408,21 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
}
if (sess->change_ssrc) {
GST_DEBUG ("need to change our SSRC (%08x)", sess->source->ssrc);
GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc));
GINT_TO_POINTER (own->ssrc));
sess->source->ssrc = rtp_session_create_new_ssrc (sess);
rtp_source_reset (sess->source);
own->ssrc = rtp_session_create_new_ssrc (sess);
rtp_source_reset (own);
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc), sess->source);
GINT_TO_POINTER (own->ssrc), own);
g_free (sess->bye_reason);
sess->bye_reason = NULL;
sess->sent_bye = FALSE;
sess->change_ssrc = FALSE;
GST_DEBUG ("changed our SSRC to %08x", sess->source->ssrc);
GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
}
RTP_SESSION_UNLOCK (sess);
@ -2426,7 +2433,7 @@ rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
GST_DEBUG ("sending packet");
if (sess->callbacks.send_rtcp)
result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
result = sess->callbacks.send_rtcp (sess, own, data.rtcp,
sess->sent_bye, sess->send_rtcp_user_data);
else {
GST_DEBUG ("freeing packet");

3
gst/rtpmanager/rtpsession.h
View file

@ -282,7 +282,8 @@ RTPSource* rtp_session_create_source (RTPSession *sess);
/* processing packets from receivers */
GstFlowReturn rtp_session_process_rtp (RTPSession *sess, GstBuffer *buffer,
GstClockTime current_time, guint64 ntpnstime);
GstClockTime current_time,
GstClockTime running_time, guint64 ntpnstime);
GstFlowReturn rtp_session_process_rtcp (RTPSession *sess, GstBuffer *buffer,
GstClockTime current_time);

174
gst/rtpmanager/rtpsource.c
View file

@ -60,6 +60,7 @@ enum
PROP_SDES_LOCATION,
PROP_SDES_TOOL,
PROP_SDES_NOTE,
PROP_STATS,
PROP_LAST
};
@ -89,55 +90,73 @@ rtp_source_class_init (RTPSourceClass * klass)
g_object_class_install_property (gobject_class, PROP_SSRC,
g_param_spec_uint ("ssrc", "SSRC",
"The SSRC of this source", 0, G_MAXUINT,
DEFAULT_SSRC, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY));
DEFAULT_SSRC,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_CSRC,
g_param_spec_boolean ("is-csrc", "Is CSRC",
"If this SSRC is acting as a contributing source",
DEFAULT_IS_CSRC, G_PARAM_READABLE));
DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_VALIDATED,
g_param_spec_boolean ("is-validated", "Is Validated",
"If this SSRC is validated", DEFAULT_IS_VALIDATED, G_PARAM_READABLE));
"If this SSRC is validated", DEFAULT_IS_VALIDATED,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_IS_SENDER,
g_param_spec_boolean ("is-sender", "Is Sender",
"If this SSRC is a sender", DEFAULT_IS_SENDER, G_PARAM_READABLE));
"If this SSRC is a sender", DEFAULT_IS_SENDER,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
g_param_spec_string ("sdes-cname", "SDES CNAME",
"The CNAME to put in SDES messages of this source",
DEFAULT_SDES_CNAME, G_PARAM_READWRITE));
DEFAULT_SDES_CNAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_NAME,
g_param_spec_string ("sdes-name", "SDES NAME",
"The NAME to put in SDES messages of this source",
DEFAULT_SDES_NAME, G_PARAM_READWRITE));
DEFAULT_SDES_NAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
g_param_spec_string ("sdes-email", "SDES EMAIL",
"The EMAIL to put in SDES messages of this source",
DEFAULT_SDES_EMAIL, G_PARAM_READWRITE));
DEFAULT_SDES_EMAIL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
g_param_spec_string ("sdes-phone", "SDES PHONE",
"The PHONE to put in SDES messages of this source",
DEFAULT_SDES_PHONE, G_PARAM_READWRITE));
DEFAULT_SDES_PHONE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
g_param_spec_string ("sdes-location", "SDES LOCATION",
"The LOCATION to put in SDES messages of this source",
DEFAULT_SDES_LOCATION, G_PARAM_READWRITE));
DEFAULT_SDES_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
g_param_spec_string ("sdes-tool", "SDES TOOL",
"The TOOL to put in SDES messages of this source",
DEFAULT_SDES_TOOL, G_PARAM_READWRITE));
DEFAULT_SDES_TOOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
g_param_spec_string ("sdes-note", "SDES NOTE",
"The NOTE to put in SDES messages of this source",
DEFAULT_SDES_NOTE, G_PARAM_READWRITE));
DEFAULT_SDES_NOTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* RTPSource::stats
*
* The statistics of the source. This property returns a GstStructure with
* name application/x-rtp-source-stats with the following fields:
*
*
*
*
*/
g_object_class_install_property (gobject_class, PROP_STATS,
g_param_spec_boxed ("stats", "Stats",
"The stats of this source", GST_TYPE_STRUCTURE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
}
@ -166,6 +185,7 @@ rtp_source_init (RTPSource * src)
/* sources are initialy on probation until we receive enough valid RTP
* packets or a valid RTCP packet */
src->validated = FALSE;
src->internal = FALSE;
src->probation = RTP_DEFAULT_PROBATION;
src->payload = 0;
@ -200,6 +220,84 @@ rtp_source_finalize (GObject * object)
G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
}
static GstStructure *
rtp_source_create_stats (RTPSource * src)
{
GstStructure *s;
gboolean is_sender = src->is_sender;
gboolean internal = src->internal;
/* common data for all types of sources */
s = gst_structure_new ("application/x-rtp-source-stats",
"ssrc", G_TYPE_UINT, (guint) src->ssrc,
"internal", G_TYPE_BOOLEAN, internal,
"validated", G_TYPE_BOOLEAN, src->validated,
"received-bye", G_TYPE_BOOLEAN, src->received_bye,
"is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
"is-sender", G_TYPE_BOOLEAN, is_sender, NULL);
if (internal) {
/* our internal source */
if (is_sender) {
/* if we are sending, report about how much we sent, other sources will
* have a RB with info on reception. */
gst_structure_set (s,
"octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
"packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
"bitrate", G_TYPE_UINT64, src->bitrate, NULL);
} else {
/* if we are not sending we have nothing more to report */
}
} else {
gboolean have_rb;
guint8 fractionlost = 0;
gint32 packetslost = 0;
guint32 exthighestseq = 0;
guint32 jitter = 0;
guint32 lsr = 0;
guint32 dlsr = 0;
guint32 round_trip = 0;
/* other sources */
if (is_sender) {
gboolean have_sr;
GstClockTime time = 0;
guint64 ntptime = 0;
guint32 rtptime = 0;
guint32 packet_count = 0;
guint32 octet_count = 0;
/* this source is sending to us, get the last SR. */
have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
&packet_count, &octet_count);
gst_structure_set (s,
"octets-received", G_TYPE_UINT64, src->stats.octets_received,
"packets-received", G_TYPE_UINT64, src->stats.packets_received,
"have-sr", G_TYPE_BOOLEAN, have_sr,
"sr-ntptime", G_TYPE_UINT64, ntptime,
"sr-rtptime", G_TYPE_UINT, (guint) rtptime,
"sr-octet-count", G_TYPE_UINT, (guint) octet_count,
"sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
}
/* we might be sending to this SSRC so we report about how it is
* receiving our data */
have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
&exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
gst_structure_set (s,
"have-rb", G_TYPE_BOOLEAN, have_rb,
"rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
"rb-packetslost", G_TYPE_INT, (gint) packetslost,
"rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
"rb-jitter", G_TYPE_UINT, (guint) jitter,
"rb-lsr", G_TYPE_UINT, (guint) lsr,
"rb-dlsr", G_TYPE_UINT, (guint) dlsr,
"rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
}
return s;
}
static void
rtp_source_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
@ -295,6 +393,9 @@ rtp_source_get_property (GObject * object, guint prop_id,
g_value_take_string (value, rtp_source_get_sdes_string (src,
GST_RTCP_SDES_NOTE));
break;
case PROP_STATS:
g_value_take_boxed (value, rtp_source_create_stats (src));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -992,6 +1093,7 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
guint32 rtptime;
guint64 ext_rtptime;
guint64 ntp_diff, rtp_diff;
guint64 elapsed;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
@ -1006,6 +1108,34 @@ rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
/* update stats for the SR */
src->stats.packets_sent++;
src->stats.octets_sent += len;
src->bytes_sent += len;
if (src->prev_ntpnstime) {
elapsed = ntpnstime - src->prev_ntpnstime;
if (elapsed > (G_GINT64_CONSTANT (1) << 31)) {
guint64 rate;
rate =
gst_util_uint64_scale (src->bytes_sent, elapsed,
(G_GINT64_CONSTANT (1) << 29));
GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
", rate %" G_GUINT64_FORMAT, elapsed, src->bytes_sent, rate);
if (src->bitrate == 0)
src->bitrate = rate;
else
src->bitrate = ((src->bitrate * 3) + rate) / 4;
src->prev_ntpnstime = ntpnstime;
src->bytes_sent = 0;
}
} else {
GST_LOG ("Reset bitrate measurement");
src->prev_ntpnstime = ntpnstime;
src->bitrate = 0;
}
rtptime = gst_rtp_buffer_get_timestamp (buffer);
ext_rtptime = src->last_rtptime;
@ -1140,10 +1270,13 @@ rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost,
curr->lsr = lsr;
curr->dlsr = dlsr;
/* calculate round trip */
ntp = (gst_rtcp_unix_to_ntp (time) >> 16) & 0xffffffff;
A = ntp - dlsr;
A -= lsr;
/* calculate round trip, round the time up */
ntp = ((gst_rtcp_unix_to_ntp (time) + 0xffff) >> 16) & 0xffffffff;
A = dlsr + lsr;
if (A > 0 && ntp > A)
A = ntp - A;
else
A = 0;
curr->round_trip = A;
GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
@ -1229,7 +1362,7 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
/**
* rtp_source_get_new_rb:
* @src: an #RTPSource
* @ntpnstime: the current time in nanoseconds since 1970
* @time: the current time of the system clock
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
@ -1242,7 +1375,7 @@ rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
* Returns: %TRUE on success.
*/
gboolean
rtp_source_get_new_rb (RTPSource * src, guint64 ntpnstime,
rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
guint32 * jitter, guint32 * lsr, guint32 * dlsr)
{
@ -1288,7 +1421,7 @@ rtp_source_get_new_rb (RTPSource * src, guint64 ntpnstime,
/* LSR is middle 32 bits of the last ntptime */
LSR = (ntptime >> 16) & 0xffffffff;
diff = ntpnstime - sr_time;
diff = time - sr_time;
GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
/* DLSR, delay since last SR is expressed in 1/65536 second units */
DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
@ -1365,6 +1498,7 @@ rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
* @jitter: the interarrival jitter
* @lsr: the last SR packet from this source
* @dlsr: the delay since last SR packet
* @round_trip: the round trip time
*
* Get the values of the last RB report set with rtp_source_process_rb().
*
@ -1373,7 +1507,7 @@ rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
gboolean
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
guint32 * lsr, guint32 * dlsr)
guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
{
RTPReceiverReport *curr;
@ -1395,6 +1529,8 @@ rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
*lsr = curr->lsr;
if (dlsr)
*dlsr = curr->dlsr;
if (round_trip)
*round_trip = curr->round_trip;
return TRUE;
}

10
gst/rtpmanager/rtpsource.h
View file

@ -113,6 +113,7 @@ struct _RTPSource {
gint probation;
gboolean validated;
gboolean internal;
gboolean is_csrc;
gboolean is_sender;
@ -139,6 +140,11 @@ struct _RTPSource {
GstClockTime last_rtptime;
GstClockTime last_ntpnstime;
/* for bitrate estimation */
guint64 bitrate;
GstClockTime prev_ntpnstime;
guint64 bytes_sent;
GQueue *packets;
RTPSourceCallbacks callbacks;
@ -198,7 +204,7 @@ void rtp_source_process_rb (RTPSource *src, GstClockTime tim
gint32 packetslost, guint32 exthighestseq, guint32 jitter,
guint32 lsr, guint32 dlsr);
gboolean rtp_source_get_new_sr (RTPSource *src, GstClockTime time, guint64 *ntptime,
gboolean rtp_source_get_new_sr (RTPSource *src, guint64 ntpnstime, guint64 *ntptime,
guint32 *rtptime, guint32 *packet_count,
guint32 *octet_count);
gboolean rtp_source_get_new_rb (RTPSource *src, GstClockTime time, guint8 *fractionlost,
@ -210,7 +216,7 @@ gboolean rtp_source_get_last_sr (RTPSource *src, GstClockTime *ti
guint32 *octet_count);
gboolean rtp_source_get_last_rb (RTPSource *src, guint8 *fractionlost, gint32 *packetslost,
guint32 *exthighestseq, guint32 *jitter,
guint32 *lsr, guint32 *dlsr);
guint32 *lsr, guint32 *dlsr, guint32 *round_trip);
void rtp_source_reset (RTPSource * src);

6
gst/rtpmanager/rtpstats.h
View file

@ -57,7 +57,9 @@ typedef struct {
/**
* RTPArrivalStats:
* @time: arrival time of a packet according to the system clock
* @timestamp: arrival time of a packet as buffer timestamp
* @running_time: arrival time of a packet as buffer running_time
* @ntpnstime: arrival time of a packet as NTP time in nanoseconds
* @have_address: if the @address field contains a valid address
* @address: address of the sender of the packet
* @bytes: bytes of the packet including lowlevel overhead
* @payload_len: bytes of the RTP payload
@ -66,7 +68,7 @@ typedef struct {
*/
typedef struct {
GstClockTime time;
GstClockTime timestamp;
GstClockTime running_time;
guint64 ntpnstime;
gboolean have_address;
GstNetAddress address;