gstreamer/gst/rtpmanager/rtpsource.c
Wim Taymans 56d5832287 gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Link to the right pads regardless of which one was created first in the
ssrc demuxer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsource.c: (calculate_jitter):
Improve debugging.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
(gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links):
* gst/rtpmanager/gstrtpssrcdemux.h:
Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
2009-08-11 02:30:30 +01:00

928 lines
26 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "rtpsource.h"
GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
#define GST_CAT_DEFAULT rtp_source_debug
#define RTP_MAX_PROBATION_LEN 32
/* signals and args */
enum
{
LAST_SIGNAL
};
enum
{
PROP_0
};
/* GObject vmethods */
static void rtp_source_finalize (GObject * object);
/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
static void
rtp_source_class_init (RTPSourceClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_source_finalize;
GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
}
static void
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->probation = RTP_DEFAULT_PROBATION;
src->payload = 0;
src->clock_rate = -1;
src->clock_base = -1;
src->packets = g_queue_new ();
src->seqnum_base = -1;
src->last_rtptime = -1;
src->stats.cycles = -1;
src->stats.jitter = 0;
src->stats.transit = -1;
src->stats.curr_sr = 0;
src->stats.curr_rr = 0;
}
static void
rtp_source_finalize (GObject * object)
{
RTPSource *src;
GstBuffer *buffer;
src = RTP_SOURCE_CAST (object);
while ((buffer = g_queue_pop_head (src->packets)))
gst_buffer_unref (buffer);
g_queue_free (src->packets);
G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
}
/**
* rtp_source_new:
* @ssrc: an SSRC
*
* Create a #RTPSource with @ssrc.
*
* Returns: a new #RTPSource. Use g_object_unref() after usage.
*/
RTPSource *
rtp_source_new (guint32 ssrc)
{
RTPSource *src;
src = g_object_new (RTP_TYPE_SOURCE, NULL);
src->ssrc = ssrc;
return src;
}
/**
* rtp_source_update_caps:
* @src: an #RTPSource
* @caps: a #GstCaps
*
* Parse @caps and store all relevant information in @source.
*/
void
rtp_source_update_caps (RTPSource * src, GstCaps * caps)
{
GstStructure *s;
guint val;
gint ival;
/* nothing changed, return */
if (src->caps == caps)
return;
s = gst_caps_get_structure (caps, 0);
if (gst_structure_get_int (s, "payload", &ival))
src->payload = ival;
GST_DEBUG ("got payload %d", src->payload);
gst_structure_get_int (s, "clock-rate", &src->clock_rate);
GST_DEBUG ("got clock-rate %d", src->clock_rate);
if (gst_structure_get_uint (s, "clock-base", &val))
src->clock_base = val;
GST_DEBUG ("got clock-base %" G_GINT64_FORMAT, src->clock_base);
if (gst_structure_get_uint (s, "seqnum-base", &val))
src->seqnum_base = val;
GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
gst_caps_replace (&src->caps, caps);
}
/**
* rtp_source_set_callbacks:
* @src: an #RTPSource
* @cb: callback functions
* @user_data: user data
*
* Set the callbacks for the source.
*/
void
rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
gpointer user_data)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->callbacks.push_rtp = cb->push_rtp;
src->callbacks.clock_rate = cb->clock_rate;
src->user_data = user_data;
}
/**
* rtp_source_set_as_csrc:
* @src: an #RTPSource
*
* Configure @src as a CSRC, this will validate the RTpSource.
*/
void
rtp_source_set_as_csrc (RTPSource * src)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->validated = TRUE;
src->is_csrc = TRUE;
}
/**
* rtp_source_set_rtp_from:
* @src: an #RTPSource
* @address: the RTP address to set
*
* Set that @src is receiving RTP packets from @address. This is used for
* collistion checking.
*/
void
rtp_source_set_rtp_from (RTPSource * src, GstNetAddress * address)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->have_rtp_from = TRUE;
memcpy (&src->rtp_from, address, sizeof (GstNetAddress));
}
/**
* rtp_source_set_rtcp_from:
* @src: an #RTPSource
* @address: the RTCP address to set
*
* Set that @src is receiving RTCP packets from @address. This is used for
* collistion checking.
*/
void
rtp_source_set_rtcp_from (RTPSource * src, GstNetAddress * address)
{
g_return_if_fail (RTP_IS_SOURCE (src));
src->have_rtcp_from = TRUE;
memcpy (&src->rtcp_from, address, sizeof (GstNetAddress));
}
static GstFlowReturn
push_packet (RTPSource * src, GstBuffer * buffer)
{
GstFlowReturn ret = GST_FLOW_OK;
/* push queued packets first if any */
while (!g_queue_is_empty (src->packets)) {
GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
GST_DEBUG ("pushing queued packet");
if (src->callbacks.push_rtp)
src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
}
GST_DEBUG ("pushing new packet");
/* push packet */
if (src->callbacks.push_rtp)
ret = src->callbacks.push_rtp (src, buffer, src->user_data);
else
gst_buffer_unref (buffer);
return ret;
}
static gint
get_clock_rate (RTPSource * src, guint8 payload)
{
if (src->clock_rate == -1) {
gint clock_rate = -1;
if (src->callbacks.clock_rate)
clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
GST_DEBUG ("new payload %d, got clock-rate %d", payload, clock_rate);
src->clock_rate = clock_rate;
}
src->payload = payload;
return src->clock_rate;
}
/* Jitter is the variation in the delay of received packets in a flow. It is
* measured by comparing the interval when RTP packets were sent to the interval
* at which they were received. For instance, if packet #1 and packet #2 leave
* 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
* milliseconds. */
static void
calculate_jitter (RTPSource * src, GstBuffer * buffer,
RTPArrivalStats * arrival)
{
guint64 ntpnstime;
guint32 rtparrival, transit, rtptime;
gint32 diff;
gint clock_rate;
guint8 pt;
/* get arrival time */
if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE)
goto no_time;
pt = gst_rtp_buffer_get_payload_type (buffer);
GST_DEBUG ("SSRC %08x got payload %d", src->ssrc, pt);
/* get clockrate */
if ((clock_rate = get_clock_rate (src, pt)) == -1)
goto no_clock_rate;
rtptime = gst_rtp_buffer_get_timestamp (buffer);
/* no clock-base, take first rtptime as base */
if (src->clock_base == -1) {
GST_DEBUG ("using clock-base of %" G_GUINT32_FORMAT, rtptime);
src->clock_base = rtptime;
}
/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
* care about the absolute value, just the difference. */
rtparrival = gst_util_uint64_scale_int (ntpnstime, clock_rate, GST_SECOND);
/* transit time is difference with RTP timestamp */
transit = rtparrival - rtptime;
/* get ABS diff with previous transit time */
if (src->stats.transit != -1) {
if (transit > src->stats.transit)
diff = transit - src->stats.transit;
else
diff = src->stats.transit - transit;
} else
diff = 0;
src->stats.transit = transit;
/* update jitter, the value we store is scaled up so we can keep precision. */
src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
src->stats.prev_rtptime = src->stats.last_rtptime;
src->stats.last_rtptime = rtparrival;
GST_DEBUG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
return;
/* ERRORS */
no_time:
{
GST_WARNING ("cannot get current time");
return;
}
no_clock_rate:
{
GST_WARNING ("cannot get clock-rate for pt %d", pt);
return;
}
}
static void
init_seq (RTPSource * src, guint16 seq)
{
src->stats.base_seq = seq;
src->stats.max_seq = seq;
src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
src->stats.cycles = 0;
src->stats.packets_received = 0;
src->stats.octets_received = 0;
src->stats.bytes_received = 0;
src->stats.prev_received = 0;
src->stats.prev_expected = 0;
GST_DEBUG ("base_seq %d", seq);
}
/**
* rtp_source_process_rtp:
* @src: an #RTPSource
* @buffer: an RTP buffer
*
* Let @src handle the incomming RTP @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
RTPArrivalStats * arrival)
{
GstFlowReturn result = GST_FLOW_OK;
guint16 seqnr, udelta;
RTPSourceStats *stats;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
stats = &src->stats;
seqnr = gst_rtp_buffer_get_seq (buffer);
rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
if (stats->cycles == -1) {
GST_DEBUG ("received first buffer");
/* first time we heard of this source */
init_seq (src, seqnr);
src->stats.max_seq = seqnr - 1;
src->probation = RTP_DEFAULT_PROBATION;
}
udelta = seqnr - stats->max_seq;
/* if we are still on probation, check seqnum */
if (src->probation) {
guint16 expected;
expected = src->stats.max_seq + 1;
/* when in probation, we require consecutive seqnums */
if (seqnr == expected) {
/* expected packet */
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
src->probation--;
src->stats.max_seq = seqnr;
if (src->probation == 0) {
GST_DEBUG ("probation done!");
init_seq (src, seqnr);
} else {
GstBuffer *q;
GST_DEBUG ("probation %d: queue buffer", src->probation);
/* when still in probation, keep packets in a list. */
g_queue_push_tail (src->packets, buffer);
/* remove packets from queue if there are too many */
while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
q = g_queue_pop_head (src->packets);
gst_object_unref (q);
}
goto done;
}
} else {
GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected);
src->probation = RTP_DEFAULT_PROBATION;
src->stats.max_seq = seqnr;
goto done;
}
} else if (udelta < RTP_MAX_DROPOUT) {
/* in order, with permissible gap */
if (seqnr < stats->max_seq) {
/* sequence number wrapped - count another 64K cycle. */
stats->cycles += RTP_SEQ_MOD;
}
stats->max_seq = seqnr;
} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
/* the sequence number made a very large jump */
if (seqnr == stats->bad_seq) {
/* two sequential packets -- assume that the other side
* restarted without telling us so just re-sync
* (i.e., pretend this was the first packet). */
init_seq (src, seqnr);
} else {
/* unacceptable jump */
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
goto bad_sequence;
}
} else {
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
GST_WARNING ("duplicate or reordered packet");
}
src->stats.octets_received += arrival->payload_len;
src->stats.bytes_received += arrival->bytes;
src->stats.packets_received++;
/* the source that sent the packet must be a sender */
src->is_sender = TRUE;
src->validated = TRUE;
GST_DEBUG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
seqnr, src->stats.packets_received, src->stats.octets_received);
/* calculate jitter and perform skew correction */
calculate_jitter (src, buffer, arrival);
/* we're ready to push the RTP packet now */
result = push_packet (src, buffer);
done:
return result;
/* ERRORS */
bad_sequence:
{
GST_WARNING ("unacceptable seqnum received");
return GST_FLOW_OK;
}
}
/**
* rtp_source_process_bye:
* @src: an #RTPSource
* @reason: the reason for leaving
*
* Notify @src that a BYE packet has been received. This will make the source
* inactive.
*/
void
rtp_source_process_bye (RTPSource * src, const gchar * reason)
{
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
GST_STR_NULL (reason));
/* copy the reason and mark as received_bye */
g_free (src->bye_reason);
src->bye_reason = g_strdup (reason);
src->received_bye = TRUE;
}
/**
* rtp_source_send_rtp:
* @src: an #RTPSource
* @buffer: an RTP buffer
* @ntpnstime: the NTP time when this buffer was captured in nanoseconds
*
* Send an RTP @buffer originating from @src. This will make @src a sender.
* This function takes ownership of @buffer and modifies the SSRC in the RTP
* packet to that of @src when needed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_source_send_rtp (RTPSource * src, GstBuffer * buffer, guint64 ntpnstime)
{
GstFlowReturn result = GST_FLOW_OK;
guint len;
guint32 rtptime;
guint64 ext_rtptime;
guint64 ntp_diff, rtp_diff;
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
len = gst_rtp_buffer_get_payload_len (buffer);
rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer));
/* we are a sender now */
src->is_sender = TRUE;
/* update stats for the SR */
src->stats.packets_sent++;
src->stats.octets_sent += len;
rtptime = gst_rtp_buffer_get_timestamp (buffer);
ext_rtptime = src->last_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
GST_DEBUG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT,
src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime));
if (ext_rtptime > src->last_rtptime) {
rtp_diff = ext_rtptime - src->last_rtptime;
ntp_diff = ntpnstime - src->last_ntpnstime;
/* calc the diff so we can detect drift at the sender. This can also be used
* to guestimate the clock rate if the NTP time is locked to the RTP
* timestamps (as is the case when the capture device is providing the clock). */
GST_DEBUG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff NTP %"
GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff));
}
/* we keep track of the last received RTP timestamp and the corresponding
* NTP timestamp so that we can use this info when constructing SR reports */
src->last_rtptime = ext_rtptime;
src->last_ntpnstime = ntpnstime;
/* push packet */
if (src->callbacks.push_rtp) {
guint32 ssrc;
ssrc = gst_rtp_buffer_get_ssrc (buffer);
if (ssrc != src->ssrc) {
/* the SSRC of the packet is not correct, make a writable buffer and
* update the SSRC. This could involve a complete copy of the packet when
* it is not writable. Usually the payloader will use caps negotiation to
* get the correct SSRC from the session manager before pushing anything. */
buffer = gst_buffer_make_writable (buffer);
GST_WARNING ("updating SSRC from %08x to %08x, fix the payloader", ssrc,
src->ssrc);
gst_rtp_buffer_set_ssrc (buffer, src->ssrc);
}
GST_DEBUG ("pushing RTP packet %" G_GUINT64_FORMAT,
src->stats.packets_sent);
result = src->callbacks.push_rtp (src, buffer, src->user_data);
} else {
GST_WARNING ("no callback installed, dropping packet");
gst_buffer_unref (buffer);
}
return result;
}
/**
* rtp_source_process_sr:
* @src: an #RTPSource
* @time: time of packet arrival
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
*
* Update the sender report in @src.
*/
void
rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
guint32 rtptime, guint32 packet_count, guint32 octet_count)
{
RTPSenderReport *curr;
gint curridx;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
(guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
packet_count, octet_count);
curridx = src->stats.curr_sr ^ 1;
curr = &src->stats.sr[curridx];
/* this is a sender now */
src->is_sender = TRUE;
/* update current */
curr->is_valid = TRUE;
curr->ntptime = ntptime;
curr->rtptime = rtptime;
curr->packet_count = packet_count;
curr->octet_count = octet_count;
curr->time = time;
/* make current */
src->stats.curr_sr = curridx;
}
/**
* rtp_source_process_rb:
* @src: an #RTPSource
* @time: the current time in nanoseconds since 1970
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter
* @lsr: the last SR packet from this source
* @dlsr: the delay since last SR packet
*
* Update the report block in @src.
*/
void
rtp_source_process_rb (RTPSource * src, GstClockTime time, guint8 fractionlost,
gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr,
guint32 dlsr)
{
RTPReceiverReport *curr;
gint curridx;
guint32 ntp, A;
g_return_if_fail (RTP_IS_SOURCE (src));
GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
curridx = src->stats.curr_rr ^ 1;
curr = &src->stats.rr[curridx];
/* update current */
curr->is_valid = TRUE;
curr->fractionlost = fractionlost;
curr->packetslost = packetslost;
curr->exthighestseq = exthighestseq;
curr->jitter = jitter;
curr->lsr = lsr;
curr->dlsr = dlsr;
/* calculate round trip */
ntp = (gst_rtcp_unix_to_ntp (time) >> 16) & 0xffffffff;
A = ntp - dlsr;
A -= lsr;
curr->round_trip = A;
GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
A >> 16, A & 0xffff);
/* make current */
src->stats.curr_rr = curridx;
}
/**
* rtp_source_get_new_sr:
* @src: an #RTPSource
* @time: the current time in nanoseconds since 1970
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
*
* Get new values to put into a new SR report from this source.
*
* Returns: %TRUE on success.
*/
gboolean
rtp_source_get_new_sr (RTPSource * src, GstClockTime ntpnstime,
guint64 * ntptime, guint32 * rtptime, guint32 * packet_count,
guint32 * octet_count)
{
guint64 t_rtp;
guint64 t_current_ntp;
GstClockTimeDiff diff;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
/* use the sync params to interpollate the date->time member to rtptime. We
* use the last sent timestamp and rtptime as reference points. We assume
* that the slope of the rtptime vs timestamp curve is 1, which is certainly
* sufficient for the frequency at which we report SR and the rate we send
* out RTP packets. */
t_rtp = src->last_rtptime;
GST_DEBUG ("last_ntpnstime %" GST_TIME_FORMAT ", last_rtptime %"
G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_ntpnstime), t_rtp);
if (src->clock_rate != -1) {
/* get the diff with the SR time */
diff = GST_CLOCK_DIFF (src->last_ntpnstime, ntpnstime);
/* now translate the diff to RTP time, handle positive and negative cases.
* If there is no diff, we already set rtptime correctly above. */
if (diff > 0) {
GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
} else {
diff = -diff;
GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff));
t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
}
} else {
GST_WARNING ("no clock-rate, cannot interpollate rtp time");
}
/* convert the NTP time in nanoseconds to 32.32 fixed point */
t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
(guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
(guint32) t_rtp);
if (ntptime)
*ntptime = t_current_ntp;
if (rtptime)
*rtptime = t_rtp;
if (packet_count)
*packet_count = src->stats.packets_sent;
if (octet_count)
*octet_count = src->stats.octets_sent;
return TRUE;
}
/**
* rtp_source_get_new_rb:
* @src: an #RTPSource
* @time: the current time in nanoseconds since 1970
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter
* @lsr: the last SR packet from this source
* @dlsr: the delay since last SR packet
*
* Get the values of the last RB report set with rtp_source_process_rb().
*
* Returns: %TRUE on success.
*/
gboolean
rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
guint32 * jitter, guint32 * lsr, guint32 * dlsr)
{
RTPSourceStats *stats;
guint64 extended_max, expected;
guint64 expected_interval, received_interval, ntptime;
gint64 lost, lost_interval;
guint32 fraction, LSR, DLSR;
GstClockTime sr_time;
stats = &src->stats;
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
extended_max, expected, stats->packets_received, stats->base_seq);
lost = expected - stats->packets_received;
lost = CLAMP (lost, -0x800000, 0x7fffff);
expected_interval = expected - stats->prev_expected;
stats->prev_expected = expected;
received_interval = stats->packets_received - stats->prev_received;
stats->prev_received = stats->packets_received;
lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0)
fraction = 0;
else
fraction = (lost_interval << 8) / expected_interval;
GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
/* we scaled the jitter up for additional precision */
GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
extended_max, stats->jitter >> 4);
if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
GstClockTime diff;
/* LSR is middle 32 bits of the last ntptime */
LSR = (ntptime >> 16) & 0xffffffff;
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);
} else {
/* No valid SR received, LSR/DLSR are set to 0 then */
GST_DEBUG ("no valid SR received");
LSR = 0;
DLSR = 0;
}
GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
DLSR >> 16, DLSR & 0xffff);
if (fractionlost)
*fractionlost = fraction;
if (packetslost)
*packetslost = lost;
if (exthighestseq)
*exthighestseq = extended_max;
if (jitter)
*jitter = stats->jitter >> 4;
if (lsr)
*lsr = LSR;
if (dlsr)
*dlsr = DLSR;
return TRUE;
}
/**
* rtp_source_get_last_sr:
* @src: an #RTPSource
* @time: time of packet arrival
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
*
* Get the values of the last sender report as set with rtp_source_process_sr().
*
* Returns: %TRUE if there was a valid SR report.
*/
gboolean
rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
{
RTPSenderReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.sr[src->stats.curr_sr];
if (!curr->is_valid)
return FALSE;
if (ntptime)
*ntptime = curr->ntptime;
if (rtptime)
*rtptime = curr->rtptime;
if (packet_count)
*packet_count = curr->packet_count;
if (octet_count)
*octet_count = curr->octet_count;
if (time)
*time = curr->time;
return TRUE;
}
/**
* rtp_source_get_last_rb:
* @src: an #RTPSource
* @fractionlost: fraction lost since last SR/RR
* @packetslost: the cumululative number of packets lost
* @exthighestseq: the extended last sequence number received
* @jitter: the interarrival jitter
* @lsr: the last SR packet from this source
* @dlsr: the delay since last SR packet
*
* Get the values of the last RB report set with rtp_source_process_rb().
*
* Returns: %TRUE if there was a valid SB report.
*/
gboolean
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
guint32 * lsr, guint32 * dlsr)
{
RTPReceiverReport *curr;
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
curr = &src->stats.rr[src->stats.curr_rr];
if (!curr->is_valid)
return FALSE;
if (fractionlost)
*fractionlost = curr->fractionlost;
if (packetslost)
*packetslost = curr->packetslost;
if (exthighestseq)
*exthighestseq = curr->exthighestseq;
if (jitter)
*jitter = curr->jitter;
if (lsr)
*lsr = curr->lsr;
if (dlsr)
*dlsr = curr->dlsr;
return TRUE;
}