diff options
author | Ryan Martell <rdm4@martellventures.com> | 2006-11-03 10:55:57 +0300 |
---|---|---|
committer | Guillaume Poirier <gpoirier@mplayerhq.hu> | 2006-11-03 10:55:57 +0300 |
commit | 4a6cc06123d969fe3214ff874bc87c1aec529143 (patch) | |
tree | bbd9c958081d74727cb643e6f092e810dca2f8b0 /libavformat/rtp.c | |
parent | a21711022e893919654a9cdbd26c4bd50b5b3f41 (diff) |
add valid statistics for the RTCP receiver report.
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
Diffstat (limited to 'libavformat/rtp.c')
-rw-r--r-- | libavformat/rtp.c | 152 |
1 files changed, 143 insertions, 9 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index cd77ad586c..b30000bebd 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -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; |