/* * RTP input format * Copyright (c) 2002 Fabrice Bellard * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg 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 * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ /* needed for gethostname() */ #define _XOPEN_SOURCE 600 #include "libavcodec/get_bits.h" #include "avformat.h" #include "mpegts.h" #include #include "network.h" #include "rtpdec.h" #include "rtpdec_formats.h" //#define DEBUG /* TODO: - add RTCP statistics reporting (should be optional). - add support for h263/mpeg4 packetized output : IDEA: send a buffer to 'rtp_write_packet' contains all the packets for ONE frame. Each packet should have a four byte header containing the length in big endian format (same trick as 'url_open_dyn_packet_buf') */ /* statistics functions */ RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL; void ff_register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) { handler->next= RTPFirstDynamicPayloadHandler; RTPFirstDynamicPayloadHandler= handler; } void av_register_rtp_dynamic_payload_handlers(void) { ff_register_dynamic_payload_handler(&ff_mp4v_es_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mpeg4_generic_dynamic_handler); ff_register_dynamic_payload_handler(&ff_amr_nb_dynamic_handler); ff_register_dynamic_payload_handler(&ff_amr_wb_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h263_1998_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h263_2000_dynamic_handler); ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); ff_register_dynamic_payload_handler(&ff_vorbis_dynamic_handler); ff_register_dynamic_payload_handler(&ff_theora_dynamic_handler); ff_register_dynamic_payload_handler(&ff_qdm2_dynamic_handler); ff_register_dynamic_payload_handler(&ff_svq3_dynamic_handler); ff_register_dynamic_payload_handler(&ff_mp4a_latm_dynamic_handler); ff_register_dynamic_payload_handler(&ff_vp8_dynamic_handler); ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler); ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler); ff_register_dynamic_payload_handler(&ff_qt_rtp_aud_handler); ff_register_dynamic_payload_handler(&ff_qt_rtp_vid_handler); ff_register_dynamic_payload_handler(&ff_quicktime_rtp_aud_handler); ff_register_dynamic_payload_handler(&ff_quicktime_rtp_vid_handler); } static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) { int payload_len; while (len >= 2) { switch (buf[1]) { case RTCP_SR: if (len < 16) { av_log(NULL, AV_LOG_ERROR, "Invalid length for RTCP SR packet\n"); return AVERROR_INVALIDDATA; } payload_len = (AV_RB16(buf + 2) + 1) * 4; s->last_rtcp_ntp_time = AV_RB64(buf + 8); if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = s->last_rtcp_ntp_time; s->last_rtcp_timestamp = AV_RB32(buf + 16); buf += payload_len; len -= payload_len; break; case RTCP_BYE: return -RTCP_BYE; default: return -1; } } return -1; } #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 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) / RTCP_TX_RATIO_DEN; rtcp_bytes /= 50; // mmu_man: that's enough for me... VLC sends much less btw !? if (rtcp_bytes < 28) return -1; s->last_octet_count = s->octet_count; if (url_open_dyn_buf(&pb) < 0) return -1; // Receiver Report put_byte(pb, (RTP_VERSION << 6) + 1); /* 1 report block */ put_byte(pb, RTCP_RR); put_be16(pb, 7); /* length in words - 1 */ // our own SSRC: we use the server's SSRC + 1 to avoid conflicts put_be32(pb, s->ssrc + 1); put_be32(pb, s->ssrc); // server SSRC // some placeholders we should really fill... // 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_byte(pb, RTCP_SDES); len = strlen(s->hostname); put_be16(pb, (6 + len + 3) / 4); /* length in words - 1 */ put_be32(pb, s->ssrc); put_byte(pb, 0x01); put_byte(pb, len); put_buffer(pb, s->hostname, len); // padding for (len = (6 + len) % 4; len % 4; len++) { put_byte(pb, 0); } put_flush_packet(pb); len = url_close_dyn_buf(pb, &buf); if ((len > 0) && buf) { int result; dprintf(s->ic, "sending %d bytes of RR\n", len); result= url_write(s->rtp_ctx, buf, len); dprintf(s->ic, "result from url_write: %d\n", result); av_free(buf); } return 0; } void rtp_send_punch_packets(URLContext* rtp_handle) { ByteIOContext *pb; uint8_t *buf; int len; /* Send a small RTP packet */ if (url_open_dyn_buf(&pb) < 0) return; put_byte(pb, (RTP_VERSION << 6)); put_byte(pb, 0); /* Payload type */ put_be16(pb, 0); /* Seq */ put_be32(pb, 0); /* Timestamp */ put_be32(pb, 0); /* SSRC */ put_flush_packet(pb); len = url_close_dyn_buf(pb, &buf); if ((len > 0) && buf) url_write(rtp_handle, buf, len); av_free(buf); /* Send a minimal RTCP RR */ if (url_open_dyn_buf(&pb) < 0) return; put_byte(pb, (RTP_VERSION << 6)); put_byte(pb, RTCP_RR); /* receiver report */ put_be16(pb, 1); /* length in words - 1 */ put_be32(pb, 0); /* our own SSRC */ put_flush_packet(pb); len = url_close_dyn_buf(pb, &buf); if ((len > 0) && buf) url_write(rtp_handle, buf, len); av_free(buf); } /** * open a new RTP parse context for stream 'st'. 'st' can be NULL for * MPEG2TS streams to indicate that they should be demuxed inside the * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned) */ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, int queue_size) { RTPDemuxContext *s; s = av_mallocz(sizeof(RTPDemuxContext)); if (!s) return NULL; s->payload_type = payload_type; s->last_rtcp_ntp_time = AV_NOPTS_VALUE; s->first_rtcp_ntp_time = AV_NOPTS_VALUE; s->ic = s1; s->st = st; s->queue_size = queue_size; rtp_init_statistics(&s->statistics, 0); // do we know the initial sequence from sdp? if (!strcmp(ff_rtp_enc_name(payload_type), "MP2T")) { s->ts = ff_mpegts_parse_open(s->ic); if (s->ts == NULL) { av_free(s); return NULL; } } else { av_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { case CODEC_ID_MPEG1VIDEO: case CODEC_ID_MPEG2VIDEO: case CODEC_ID_MP2: case CODEC_ID_MP3: case CODEC_ID_MPEG4: case CODEC_ID_H263: case CODEC_ID_H264: st->need_parsing = AVSTREAM_PARSE_FULL; break; case CODEC_ID_ADPCM_G722: av_set_pts_info(st, 32, 1, st->codec->sample_rate); /* According to RFC 3551, the stream clock rate is 8000 * even if the sample rate is 16000. */ if (st->codec->sample_rate == 8000) st->codec->sample_rate = 16000; break; default: if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { av_set_pts_info(st, 32, 1, st->codec->sample_rate); } break; } } // needed to send back RTCP RR in RTSP sessions s->rtp_ctx = rtpc; gethostname(s->hostname, sizeof(s->hostname)); return s; } void rtp_parse_set_dynamic_protocol(RTPDemuxContext *s, PayloadContext *ctx, RTPDynamicProtocolHandler *handler) { s->dynamic_protocol_context = ctx; s->parse_packet = handler->parse_packet; } /** * This was the second switch in rtp_parse packet. Normalizes time, if required, sets stream_index, etc. */ static void finalize_packet(RTPDemuxContext *s, AVPacket *pkt, uint32_t timestamp) { if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE && timestamp != RTP_NOTS_VALUE) { int64_t addend; int delta_timestamp; /* compute pts from timestamp with received ntp_time */ delta_timestamp = timestamp - s->last_rtcp_timestamp; /* convert to the PTS timebase */ addend = av_rescale(s->last_rtcp_ntp_time - s->first_rtcp_ntp_time, s->st->time_base.den, (uint64_t)s->st->time_base.num << 32); pkt->pts = s->range_start_offset + addend + delta_timestamp; } } static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt, const uint8_t *buf, int len) { unsigned int ssrc, h; int payload_type, seq, ret, flags = 0; int ext; AVStream *st; uint32_t timestamp; int rv= 0; ext = buf[0] & 0x10; payload_type = buf[1] & 0x7f; if (buf[1] & 0x80) flags |= RTP_FLAG_MARKER; seq = AV_RB16(buf + 2); timestamp = AV_RB32(buf + 4); ssrc = AV_RB32(buf + 8); /* store the ssrc in the RTPDemuxContext */ s->ssrc = ssrc; /* NOTE: we can handle only one payload type */ if (s->payload_type != payload_type) return -1; st = s->st; // 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; } s->seq = seq; len -= 12; buf += 12; /* RFC 3550 Section 5.3.1 RTP Header Extension handling */ if (ext) { if (len < 4) return -1; /* calculate the header extension length (stored as number * of 32-bit words) */ ext = (AV_RB16(buf + 2) + 1) << 2; if (len < ext) return -1; // skip past RTP header extension len -= ext; buf += ext; } if (!st) { /* specific MPEG2TS demux support */ ret = ff_mpegts_parse_packet(s->ts, pkt, buf, len); if (ret < 0) { s->prev_ret = -1; return -1; } if (ret < len) { s->read_buf_size = len - ret; memcpy(s->buf, buf + ret, s->read_buf_size); s->read_buf_index = 0; s->prev_ret = 1; return 1; } s->prev_ret = 0; return 0; } else if (s->parse_packet) { rv = s->parse_packet(s->ic, s->dynamic_protocol_context, s->st, pkt, ×tamp, buf, len, flags); } else { // at this point, the RTP header has been stripped; This is ASSUMING that there is only 1 CSRC, which in't wise. switch(st->codec->codec_id) { case CODEC_ID_MP2: case CODEC_ID_MP3: /* better than nothing: skip mpeg audio RTP header */ if (len <= 4) return -1; h = AV_RB32(buf); len -= 4; buf += 4; av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; case CODEC_ID_MPEG1VIDEO: case CODEC_ID_MPEG2VIDEO: /* better than nothing: skip mpeg video RTP header */ if (len <= 4) return -1; h = AV_RB32(buf); buf += 4; len -= 4; if (h & (1 << 26)) { /* mpeg2 */ if (len <= 4) return -1; buf += 4; len -= 4; } av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; default: av_new_packet(pkt, len); memcpy(pkt->data, buf, len); break; } pkt->stream_index = st->index; } // now perform timestamp things.... finalize_packet(s, pkt, timestamp); s->prev_ret = rv; return rv; } void ff_rtp_reset_packet_queue(RTPDemuxContext *s) { while (s->queue) { RTPPacket *next = s->queue->next; av_free(s->queue->buf); av_free(s->queue); s->queue = next; } s->seq = 0; s->queue_len = 0; s->prev_ret = 0; } static void enqueue_packet(RTPDemuxContext *s, uint8_t *buf, int len) { uint16_t seq = AV_RB16(buf + 2); RTPPacket *cur = s->queue, *prev = NULL, *packet; /* Find the correct place in the queue to insert the packet */ while (cur) { int16_t diff = seq - cur->seq; if (diff < 0) break; prev = cur; cur = cur->next; } packet = av_mallocz(sizeof(*packet)); if (!packet) return; packet->recvtime = av_gettime(); packet->seq = seq; packet->len = len; packet->buf = buf; packet->next = cur; if (prev) prev->next = packet; else s->queue = packet; s->queue_len++; } static int has_next_packet(RTPDemuxContext *s) { return s->queue && s->queue->seq == s->seq + 1; } int64_t ff_rtp_queued_packet_time(RTPDemuxContext *s) { return s->queue ? s->queue->recvtime : 0; } static int rtp_parse_queued_packet(RTPDemuxContext *s, AVPacket *pkt) { int rv; RTPPacket *next; if (s->queue_len <= 0) return -1; if (!has_next_packet(s)) av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING, "RTP: missed %d packets\n", s->queue->seq - s->seq - 1); /* Parse the first packet in the queue, and dequeue it */ rv = rtp_parse_packet_internal(s, pkt, s->queue->buf, s->queue->len); next = s->queue->next; av_free(s->queue->buf); av_free(s->queue); s->queue = next; s->queue_len--; return rv ? rv : has_next_packet(s); } /** * Parse an RTP or RTCP packet directly sent as a buffer. * @param s RTP parse context. * @param pkt returned packet * @param bufptr pointer to the input buffer or NULL to read the next packets * @param len buffer len * @return 0 if a packet is returned, 1 if a packet is returned and more can follow * (use buf as NULL to read the next). -1 if no packet (error or no more packet). */ int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt, uint8_t **bufptr, int len) { uint8_t* buf = bufptr ? *bufptr : NULL; int ret, flags = 0; uint32_t timestamp; int rv= 0; if (!buf) { /* If parsing of the previous packet actually returned 0, there's * nothing more to be parsed from that packet, but we may have * indicated that we can return the next enqueued packet. */ if (!s->prev_ret) return rtp_parse_queued_packet(s, pkt); /* return the next packets, if any */ if(s->st && s->parse_packet) { /* timestamp should be overwritten by parse_packet, if not, * the packet is left with pts == AV_NOPTS_VALUE */ timestamp = RTP_NOTS_VALUE; rv= s->parse_packet(s->ic, s->dynamic_protocol_context, s->st, pkt, ×tamp, NULL, 0, flags); finalize_packet(s, pkt, timestamp); s->prev_ret = rv; return rv ? rv : has_next_packet(s); } else { // TODO: Move to a dynamic packet handler (like above) if (s->read_buf_index >= s->read_buf_size) { s->prev_ret = -1; return -1; } ret = ff_mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, s->read_buf_size - s->read_buf_index); if (ret < 0) { s->prev_ret = -1; return -1; } s->read_buf_index += ret; if (s->read_buf_index < s->read_buf_size) return 1; else { s->prev_ret = 0; return has_next_packet(s); } } } if (len < 12) return -1; if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) return -1; if (buf[1] >= RTCP_SR && buf[1] <= RTCP_APP) { return rtcp_parse_packet(s, buf, len); } if (s->seq == 0 || s->queue_size <= 1) { /* First packet, or no reordering */ return rtp_parse_packet_internal(s, pkt, buf, len); } else { uint16_t seq = AV_RB16(buf + 2); int16_t diff = seq - s->seq; if (diff < 0) { /* Packet older than the previously emitted one, drop */ av_log(s->st ? s->st->codec : NULL, AV_LOG_WARNING, "RTP: dropping old packet received too late\n"); return -1; } else if (diff <= 1) { /* Correct packet */ rv = rtp_parse_packet_internal(s, pkt, buf, len); return rv ? rv : has_next_packet(s); } else { /* Still missing some packet, enqueue this one. */ enqueue_packet(s, buf, len); *bufptr = NULL; /* Return the first enqueued packet if the queue is full, * even if we're missing something */ if (s->queue_len >= s->queue_size) return rtp_parse_queued_packet(s, pkt); return -1; } } } void rtp_parse_close(RTPDemuxContext *s) { ff_rtp_reset_packet_queue(s); if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { ff_mpegts_parse_close(s->ts); } av_free(s); } int ff_parse_fmtp(AVStream *stream, PayloadContext *data, const char *p, int (*parse_fmtp)(AVStream *stream, PayloadContext *data, char *attr, char *value)) { char attr[256]; char *value; int res; int value_size = strlen(p) + 1; if (!(value = av_malloc(value_size))) { av_log(stream, AV_LOG_ERROR, "Failed to allocate data for FMTP."); return AVERROR(ENOMEM); } // remove protocol identifier while (*p && *p == ' ') p++; // strip spaces while (*p && *p != ' ') p++; // eat protocol identifier while (*p && *p == ' ') p++; // strip trailing spaces while (ff_rtsp_next_attr_and_value(&p, attr, sizeof(attr), value, value_size)) { res = parse_fmtp(stream, data, attr, value); if (res < 0 && res != AVERROR_PATCHWELCOME) { av_free(value); return res; } } av_free(value); return 0; }