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add valid statistics for the RTCP receiver report.

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Basically taken verbatim from RFC 1889.
Patch by Ryan Martell % rdm4 A martellventures P com %
Original thread:
Date: Oct 31, 2006 12:43 AM
Subject: [Ffmpeg-devel] [PATCH] RTCP valid receiver statistics....

Originally committed as revision 6879 to svn://svn.ffmpeg.org/ffmpeg/trunk
pull/126/head
Ryan Martell 18 years ago committed by Guillaume Poirier
parent a21711022e
commit 4a6cc06123
2 changed files with 162 additions and 9 deletions
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  1. 152
      libavformat/rtp.c
  2. 19
      libavformat/rtp_internal.h

152
libavformat/rtp.c
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@ -258,6 +258,98 @@ static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int l
return 0;
}
#define RTP_SEQ_MOD (1<<16)
/**
* called on parse open packet
*/
static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
{
memset(s, 0, sizeof(RTPStatistics));
s->max_seq= base_sequence;
s->probation= 1;
}
/**
* called whenever there is a large jump in sequence numbers, or when they get out of probation...
*/
static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
{
s->max_seq= seq;
s->cycles= 0;
s->base_seq= seq -1;
s->bad_seq= RTP_SEQ_MOD + 1;
s->received= 0;
s->expected_prior= 0;
s->received_prior= 0;
s->jitter= 0;
s->transit= 0;
}
/**
* returns 1 if we should handle this packet.
*/
static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
{
uint16_t udelta= seq - s->max_seq;
const int MAX_DROPOUT= 3000;
const int MAX_MISORDER = 100;
const int MIN_SEQUENTIAL = 2;
/* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
if(s->probation)
{
if(seq==s->max_seq + 1) {
s->probation--;
s->max_seq= seq;
if(s->probation==0) {
rtp_init_sequence(s, seq);
s->received++;
return 1;
}
} else {
s->probation= MIN_SEQUENTIAL - 1;
s->max_seq = seq;
}
} else if (udelta < MAX_DROPOUT) {
// in order, with permissible gap
if(seq < s->max_seq) {
//sequence number wrapped; count antother 64k cycles
s->cycles += RTP_SEQ_MOD;
}
s->max_seq= seq;
} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
// sequence made a large jump...
if(seq==s->bad_seq) {
// two sequential packets-- assume that the other side restarted without telling us; just resync.
rtp_init_sequence(s, seq);
} else {
s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
return 0;
}
} else {
// duplicate or reordered packet...
}
s->received++;
return 1;
}
#if 0
/**
* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
* never change. I left this in in case someone else can see a way. (rdm)
*/
static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
{
uint32_t transit= arrival_timestamp - sent_timestamp;
int d;
s->transit= transit;
d= FFABS(transit - s->transit);
s->jitter += d - ((s->jitter + 8)>>4);
}
#endif
/**
* some rtp servers assume client is dead if they don't hear from them...
* so we send a Receiver Report to the provided ByteIO context
@ -269,10 +361,20 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
uint8_t *buf;
int len;
int rtcp_bytes;
RTPStatistics *stats= &s->statistics;
uint32_t lost;
uint32_t extended_max;
uint32_t expected_interval;
uint32_t received_interval;
uint32_t lost_interval;
uint32_t expected;
uint32_t fraction;
uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
if (!s->rtp_ctx || (count < 1))
return -1;
/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
s->octet_count += count;
rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
@ -292,11 +394,36 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
put_be32(&pb, s->ssrc); // our own SSRC
put_be32(&pb, s->ssrc); // XXX: should be the server's here!
// some placeholders we should really fill...
put_be32(&pb, ((0 << 24) | (0 & 0x0ffffff))); /* 0% lost, total 0 lost */
put_be32(&pb, (0 << 16) | s->seq);
put_be32(&pb, 0x68); /* jitter */
put_be32(&pb, -1); /* last SR timestamp */
put_be32(&pb, 1); /* delay since last SR */
// RFC 1889/p64
extended_max= stats->cycles + stats->max_seq;
expected= extended_max - stats->base_seq + 1;
lost= expected - stats->received;
lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
expected_interval= expected - stats->expected_prior;
stats->expected_prior= expected;
received_interval= stats->received - stats->received_prior;
stats->received_prior= stats->received;
lost_interval= expected_interval - received_interval;
if (expected_interval==0 || lost_interval<=0) fraction= 0;
else fraction = (lost_interval<<8)/expected_interval;
fraction= (fraction<<24) | lost;
put_be32(&pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
put_be32(&pb, extended_max); /* max sequence received */
put_be32(&pb, stats->jitter>>4); /* jitter */
if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
{
put_be32(&pb, 0); /* last SR timestamp */
put_be32(&pb, 0); /* delay since last SR */
} else {
uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
put_be32(&pb, middle_32_bits); /* last SR timestamp */
put_be32(&pb, delay_since_last); /* delay since last SR */
}
// CNAME
put_byte(&pb, (RTP_VERSION << 6) + 1); /* 1 report block */
@ -315,10 +442,14 @@ int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
put_flush_packet(&pb);
len = url_close_dyn_buf(&pb, &buf);
if ((len > 0) && buf) {
int result;
#if defined(DEBUG)
printf("sending %d bytes of RR\n", len);
#endif
url_write(s->rtp_ctx, buf, len);
result= url_write(s->rtp_ctx, buf, len);
#if defined(DEBUG)
printf("result from url_write: %d\n", result);
#endif
av_free(buf);
}
return 0;
@ -343,6 +474,7 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r
s->ic = s1;
s->st = st;
s->rtp_payload_data = rtp_payload_data;
rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
if (!strcmp(AVRtpPayloadTypes[payload_type].enc_name, "MP2T")) {
s->ts = mpegts_parse_open(s->ic);
if (s->ts == NULL) {
@ -514,12 +646,14 @@ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
return -1;
st = s->st;
#if defined(DEBUG) || 1
if (seq != ((s->seq + 1) & 0xffff)) {
// only do something with this if all the rtp checks pass...
if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
{
av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
return -1;
}
#endif
s->seq = seq;
len -= 12;
buf += 12;

19
libavformat/rtp_internal.h
Unescape View File

@ -23,6 +23,21 @@
#ifndef RTP_INTERNAL_H
#define RTP_INTERNAL_H
// these statistics are used for rtcp receiver reports...
typedef struct {
uint16_t max_seq; ///< highest sequence number seen
uint32_t cycles; ///< shifted count of sequence number cycles
uint32_t base_seq; ///< base sequence number
uint32_t bad_seq; ///< last bad sequence number + 1
int probation; ///< sequence packets till source is valid
int received; ///< packets received
int expected_prior; ///< packets expected in last interval
int received_prior; ///< packets received in last interval
uint32_t transit; ///< relative transit time for previous packet
uint32_t jitter; ///< estimated jitter.
} RTPStatistics;
typedef int (*DynamicPayloadPacketHandlerProc) (struct RTPDemuxContext * s,
AVPacket * pkt,
uint32_t *timestamp,
@ -64,6 +79,8 @@ struct RTPDemuxContext {
URLContext *rtp_ctx;
char hostname[256];
RTPStatistics statistics; ///< Statistics for this stream (used by RTCP receiver reports)
/* rtcp sender statistics receive */
int64_t last_rtcp_ntp_time; // TODO: move into statistics
int64_t first_rtcp_ntp_time; // TODO: move into statistics
@ -87,5 +104,7 @@ struct RTPDemuxContext {
};
extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers.
#endif /* RTP_INTERNAL_H */

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