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* Copyright (c) 2002 Fabrice Bellard.
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* This file is part of FFmpeg.
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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#include "bitstream.h"
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#include "rtp_internal.h"
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/* TODO: - add RTCP statistics reporting (should be optional).
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- add support for h263/mpeg4 packetized output : IDEA: send a
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buffer to 'rtp_write_packet' contains all the packets for ONE
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frame. Each packet should have a four byte header containing
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the length in big endian format (same trick as
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'url_open_dyn_packet_buf')
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/* statistics functions */
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RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
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static RTPDynamicProtocolHandler mp4v_es_handler= {"MP4V-ES", CODEC_TYPE_VIDEO, CODEC_ID_MPEG4};
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static RTPDynamicProtocolHandler mpeg4_generic_handler= {"mpeg4-generic", CODEC_TYPE_AUDIO, CODEC_ID_AAC};
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static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler)
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handler->next= RTPFirstDynamicPayloadHandler;
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RTPFirstDynamicPayloadHandler= handler;
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void av_register_rtp_dynamic_payload_handlers(void)
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register_dynamic_payload_handler(&mp4v_es_handler);
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register_dynamic_payload_handler(&mpeg4_generic_handler);
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register_dynamic_payload_handler(&ff_h264_dynamic_handler);
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static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len)
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s->last_rtcp_ntp_time = AV_RB64(buf + 8);
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if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE)
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s->first_rtcp_ntp_time = s->last_rtcp_ntp_time;
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s->last_rtcp_timestamp = AV_RB32(buf + 16);
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#define RTP_SEQ_MOD (1<<16)
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* called on parse open packet
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static void rtp_init_statistics(RTPStatistics *s, uint16_t base_sequence) // called on parse open packet.
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memset(s, 0, sizeof(RTPStatistics));
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s->max_seq= base_sequence;
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* called whenever there is a large jump in sequence numbers, or when they get out of probation...
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static void rtp_init_sequence(RTPStatistics *s, uint16_t seq)
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s->bad_seq= RTP_SEQ_MOD + 1;
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* returns 1 if we should handle this packet.
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static int rtp_valid_packet_in_sequence(RTPStatistics *s, uint16_t seq)
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uint16_t udelta= seq - s->max_seq;
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const int MAX_DROPOUT= 3000;
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const int MAX_MISORDER = 100;
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const int MIN_SEQUENTIAL = 2;
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/* source not valid until MIN_SEQUENTIAL packets with sequence seq. numbers have been received */
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if(seq==s->max_seq + 1) {
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if(s->probation==0) {
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rtp_init_sequence(s, seq);
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s->probation= MIN_SEQUENTIAL - 1;
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} else if (udelta < MAX_DROPOUT) {
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// in order, with permissible gap
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if(seq < s->max_seq) {
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//sequence number wrapped; count antother 64k cycles
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s->cycles += RTP_SEQ_MOD;
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} else if (udelta <= RTP_SEQ_MOD - MAX_MISORDER) {
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// sequence made a large jump...
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if(seq==s->bad_seq) {
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// two sequential packets-- assume that the other side restarted without telling us; just resync.
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rtp_init_sequence(s, seq);
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s->bad_seq= (seq + 1) & (RTP_SEQ_MOD-1);
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// duplicate or reordered packet...
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* This function is currently unused; without a valid local ntp time, I don't see how we could calculate the
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* difference between the arrival and sent timestamp. As a result, the jitter and transit statistics values
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* never change. I left this in in case someone else can see a way. (rdm)
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static void rtcp_update_jitter(RTPStatistics *s, uint32_t sent_timestamp, uint32_t arrival_timestamp)
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uint32_t transit= arrival_timestamp - sent_timestamp;
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d= FFABS(transit - s->transit);
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s->jitter += d - ((s->jitter + 8)>>4);
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int rtp_check_and_send_back_rr(RTPDemuxContext *s, int count)
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RTPStatistics *stats= &s->statistics;
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uint32_t extended_max;
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uint32_t expected_interval;
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uint32_t received_interval;
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uint32_t lost_interval;
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uint64_t ntp_time= s->last_rtcp_ntp_time; // TODO: Get local ntp time?
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if (!s->rtp_ctx || (count < 1))
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/* TODO: I think this is way too often; RFC 1889 has algorithm for this */
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/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
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s->octet_count += count;
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rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
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rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !?
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s->last_octet_count = s->octet_count;
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if (url_open_dyn_buf(&pb) < 0)
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put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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put_be16(pb, 7); /* length in words - 1 */
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put_be32(pb, s->ssrc); // our own SSRC
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put_be32(pb, s->ssrc); // XXX: should be the server's here!
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// some placeholders we should really fill...
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extended_max= stats->cycles + stats->max_seq;
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expected= extended_max - stats->base_seq + 1;
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lost= expected - stats->received;
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lost= FFMIN(lost, 0xffffff); // clamp it since it's only 24 bits...
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expected_interval= expected - stats->expected_prior;
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stats->expected_prior= expected;
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received_interval= stats->received - stats->received_prior;
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stats->received_prior= stats->received;
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lost_interval= expected_interval - received_interval;
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if (expected_interval==0 || lost_interval<=0) fraction= 0;
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else fraction = (lost_interval<<8)/expected_interval;
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fraction= (fraction<<24) | lost;
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put_be32(pb, fraction); /* 8 bits of fraction, 24 bits of total packets lost */
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put_be32(pb, extended_max); /* max sequence received */
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put_be32(pb, stats->jitter>>4); /* jitter */
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if(s->last_rtcp_ntp_time==AV_NOPTS_VALUE)
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put_be32(pb, 0); /* last SR timestamp */
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put_be32(pb, 0); /* delay since last SR */
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uint32_t middle_32_bits= s->last_rtcp_ntp_time>>16; // this is valid, right? do we need to handle 64 bit values special?
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uint32_t delay_since_last= ntp_time - s->last_rtcp_ntp_time;
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put_be32(pb, middle_32_bits); /* last SR timestamp */
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put_be32(pb, delay_since_last); /* delay since last SR */
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put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */
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len = strlen(s->hostname);
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put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */
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put_be32(pb, s->ssrc);
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put_buffer(pb, s->hostname, len);
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for (len = (6 + len) % 4; len % 4; len++) {
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put_flush_packet(pb);
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len = url_close_dyn_buf(pb, &buf);
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if ((len > 0) && buf) {
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printf("sending %d bytes of RR\n", len);
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result= url_write(s->rtp_ctx, buf, len);
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printf("result from url_write: %d\n", result);
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* open a new RTP parse context for stream 'st'. 'st' can be NULL for
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* MPEG2TS streams to indicate that they should be demuxed inside the
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* rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
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* TODO: change this to not take rtp_payload data, and use the new dynamic payload system.
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RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, rtp_payload_data_t *rtp_payload_data)
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s = av_mallocz(sizeof(RTPDemuxContext));
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s->payload_type = payload_type;
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s->last_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->first_rtcp_ntp_time = AV_NOPTS_VALUE;
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s->rtp_payload_data = rtp_payload_data;
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rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp?
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if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) {
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s->ts = mpegts_parse_open(s->ic);
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switch(st->codec->codec_id) {
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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st->need_parsing = AVSTREAM_PARSE_FULL;
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// needed to send back RTCP RR in RTSP sessions
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gethostname(s->hostname, sizeof(s->hostname));
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static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
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int au_headers_length, au_header_size, i;
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GetBitContext getbitcontext;
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rtp_payload_data_t *infos;
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infos = s->rtp_payload_data;
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/* decode the first 2 bytes where the AUHeader sections are stored
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au_headers_length = AV_RB16(buf);
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if (au_headers_length > RTP_MAX_PACKET_LENGTH)
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infos->au_headers_length_bytes = (au_headers_length + 7) / 8;
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/* skip AU headers length section (2 bytes) */
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init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8);
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/* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
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au_header_size = infos->sizelength + infos->indexlength;
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if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
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infos->nb_au_headers = au_headers_length / au_header_size;
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infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers);
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/* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
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In my test, the FAAD decoder does not behave correctly when sending each AU one by one
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but does when sending the whole as one big packet... */
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infos->au_headers[0].size = 0;
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infos->au_headers[0].index = 0;
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for (i = 0; i < infos->nb_au_headers; ++i) {
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infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength);
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infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength);
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infos->nb_au_headers = 1;
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* This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc.
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static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp)
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switch(s->st->codec->codec_id) {
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
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/* XXX: is it really necessary to unify the timestamp base ? */
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/* compute pts from timestamp with received ntp_time */
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delta_timestamp = timestamp - s->last_rtcp_timestamp;
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/* convert to 90 kHz without overflow */
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addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14;
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addend = (addend * 5625) >> 14;
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pkt->pts = addend + delta_timestamp;
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pkt->pts = timestamp;
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/* no timestamp info yet */
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pkt->stream_index = s->st->index;
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* Parse an RTP or RTCP packet directly sent as a buffer.
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* @param s RTP parse context.
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* @param pkt returned packet
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* @param buf input buffer or NULL to read the next packets
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* @param len buffer len
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* @return 0 if a packet is returned, 1 if a packet is returned and more can follow
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* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
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int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
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const uint8_t *buf, int len)
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unsigned int ssrc, h;
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int payload_type, seq, ret, flags = 0;
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/* return the next packets, if any */
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if(s->st && s->parse_packet) {
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timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned....
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rv= s->parse_packet(s, pkt, ×tamp, NULL, 0, flags);
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finalize_packet(s, pkt, timestamp);
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// TODO: Move to a dynamic packet handler (like above)
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if (s->read_buf_index >= s->read_buf_size)
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ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
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s->read_buf_size - s->read_buf_index);
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s->read_buf_index += ret;
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if (s->read_buf_index < s->read_buf_size)
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if ((buf[0] & 0xc0) != (RTP_VERSION << 6))
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if (buf[1] >= 200 && buf[1] <= 204) {
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rtcp_parse_packet(s, buf, len);
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payload_type = buf[1] & 0x7f;
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seq = AV_RB16(buf + 2);
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timestamp = AV_RB32(buf + 4);
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ssrc = AV_RB32(buf + 8);
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/* store the ssrc in the RTPDemuxContext */
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/* NOTE: we can handle only one payload type */
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if (s->payload_type != payload_type)
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// only do something with this if all the rtp checks pass...
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if(!rtp_valid_packet_in_sequence(&s->statistics, seq))
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av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
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payload_type, seq, ((s->seq + 1) & 0xffff));
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/* specific MPEG2TS demux support */
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ret = mpegts_parse_packet(s->ts, pkt, buf, len);
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s->read_buf_size = len - ret;
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memcpy(s->buf, buf + ret, s->read_buf_size);
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s->read_buf_index = 0;
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} else if (s->parse_packet) {
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rv = s->parse_packet(s, pkt, ×tamp, buf, len, flags);
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// at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise.
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switch(st->codec->codec_id) {
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/* better than nothing: skip mpeg audio RTP header */
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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case CODEC_ID_MPEG1VIDEO:
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case CODEC_ID_MPEG2VIDEO:
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/* better than nothing: skip mpeg video RTP header */
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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// moved from below, verbatim. this is because this section handles packets, and the lower switch handles
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// TODO: Put this into a dynamic packet handler...
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if (rtp_parse_mp4_au(s, buf))
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rtp_payload_data_t *infos = s->rtp_payload_data;
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buf += infos->au_headers_length_bytes + 2;
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len -= infos->au_headers_length_bytes + 2;
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/* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
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av_new_packet(pkt, infos->au_headers[0].size);
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memcpy(pkt->data, buf, infos->au_headers[0].size);
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buf += infos->au_headers[0].size;
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len -= infos->au_headers[0].size;
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s->read_buf_size = len;
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av_new_packet(pkt, len);
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memcpy(pkt->data, buf, len);
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// now perform timestamp things....
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finalize_packet(s, pkt, timestamp);
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void rtp_parse_close(RTPDemuxContext *s)
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// TODO: fold this into the protocol specific data fields.
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if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) {
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mpegts_parse_close(s->ts);
added
removed
ffmpeg/libavformat/rtpdec.c
