#include "bitstream.h"
#include <unistd.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <netinet/in.h>
-#ifndef __BEOS__
-# include <arpa/inet.h>
-#else
-# include "barpainet.h"
-#endif
-#include <netdb.h>
+#include "network.h"
#include "rtp_internal.h"
-
-//#define RTP_H264
-#ifdef RTP_H264
- #include "rtp_h264.h"
-#endif
+#include "rtp_h264.h"
//#define DEBUG
RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
-static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_MPEG4AAC};
+static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
{
RTPFirstDynamicPayloadHandler= handler;
}
-void av_register_rtp_dynamic_payload_handlers()
+void av_register_rtp_dynamic_payload_handlers(void)
{
register_dynamic_payload_handler(&mp4v_es_handler);
register_dynamic_payload_handler(&mpeg4_generic_handler);
-#ifdef RTP_H264
register_dynamic_payload_handler(&ff_h264_dynamic_handler);
-#endif
}
int rtp_get_codec_info(AVCodecContext *codec, int payload_type)
return -1;
}
-/* return < 0 if unknown payload type */
int rtp_get_payload_type(AVCodecContext *codec)
{
int i, payload_type;
return 0;
}
+#define RTP_SEQ_MOD (1<<16)
+
/**
- * 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
- * (we don't have access to the rtcp handle from here)
- */
+* 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
+
int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
{
ByteIOContext pb;
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) /
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 */
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;
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) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
case CODEC_ID_MPEG4:
-#ifdef RTP_H264
case CODEC_ID_H264:
-#endif
st->need_parsing = 1;
break;
default:
/* decode the first 2 bytes where are stored the AUHeader sections
length in bits */
- au_headers_length = BE_16(buf);
+ au_headers_length = AV_RB16(buf);
if (au_headers_length > RTP_MAX_PACKET_LENGTH)
return -1;
pkt->pts = addend + delta_timestamp;
}
break;
- case CODEC_ID_MPEG4AAC:
+ case CODEC_ID_AAC:
case CODEC_ID_H264:
case CODEC_ID_MPEG4:
pkt->pts = timestamp;
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;
// moved from below, verbatim. this is because this section handles packets, and the lower switch handles
// timestamps.
// TODO: Put this into a dynamic packet handler...
- case CODEC_ID_MPEG4AAC:
+ case CODEC_ID_AAC:
if (rtp_parse_mp4_au(s, buf))
return -1;
{
projects / ffmpeg / blobdiff