| /* |
| * 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/bitstream.h" |
| #include "avformat.h" |
| #include "mpegts.h" |
| |
| #include <unistd.h> |
| #include "network.h" |
| |
| #include "rtpdec.h" |
| #include "rtp_h264.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; |
| |
| 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_AAC}; |
| |
| 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(&mp4v_es_handler); |
| ff_register_dynamic_payload_handler(&mpeg4_generic_handler); |
| ff_register_dynamic_payload_handler(&ff_h264_dynamic_handler); |
| } |
| |
| static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) |
| { |
| if (buf[1] != 200) |
| return -1; |
| 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); |
| 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 |
| |
| 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, 201); |
| put_be16(pb, 7); /* length in words - 1 */ |
| 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... |
| // 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, 202); |
| 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; |
| } |
| |
| /** |
| * 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) |
| * TODO: change this to not take rtp_payload data, and use the new dynamic payload system. |
| */ |
| RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *rtpc, int payload_type, RTPPayloadData *rtp_payload_data) |
| { |
| 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->rtp_payload_data = rtp_payload_data; |
| 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 = 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_H264: |
| st->need_parsing = AVSTREAM_PARSE_FULL; |
| break; |
| default: |
| if (st->codec->codec_type == CODEC_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; |
| } |
| |
| static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) |
| { |
| int au_headers_length, au_header_size, i; |
| GetBitContext getbitcontext; |
| RTPPayloadData *infos; |
| |
| infos = s->rtp_payload_data; |
| |
| if (infos == NULL) |
| return -1; |
| |
| /* decode the first 2 bytes where the AUHeader sections are stored |
| length in bits */ |
| au_headers_length = AV_RB16(buf); |
| |
| if (au_headers_length > RTP_MAX_PACKET_LENGTH) |
| return -1; |
| |
| infos->au_headers_length_bytes = (au_headers_length + 7) / 8; |
| |
| /* skip AU headers length section (2 bytes) */ |
| buf += 2; |
| |
| init_get_bits(&getbitcontext, buf, infos->au_headers_length_bytes * 8); |
| |
| /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */ |
| au_header_size = infos->sizelength + infos->indexlength; |
| if (au_header_size <= 0 || (au_headers_length % au_header_size != 0)) |
| return -1; |
| |
| infos->nb_au_headers = au_headers_length / au_header_size; |
| infos->au_headers = av_malloc(sizeof(struct AUHeaders) * infos->nb_au_headers); |
| |
| /* XXX: We handle multiple AU Section as only one (need to fix this for interleaving) |
| In my test, the FAAD decoder does not behave correctly when sending each AU one by one |
| but does when sending the whole as one big packet... */ |
| infos->au_headers[0].size = 0; |
| infos->au_headers[0].index = 0; |
| for (i = 0; i < infos->nb_au_headers; ++i) { |
| infos->au_headers[0].size += get_bits_long(&getbitcontext, infos->sizelength); |
| infos->au_headers[0].index = get_bits_long(&getbitcontext, infos->indexlength); |
| } |
| |
| infos->nb_au_headers = 1; |
| |
| return 0; |
| } |
| |
| /** |
| * 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) { |
| 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 = addend + delta_timestamp; |
| } |
| pkt->stream_index = s->st->index; |
| } |
| |
| /** |
| * Parse an RTP or RTCP packet directly sent as a buffer. |
| * @param s RTP parse context. |
| * @param pkt returned packet |
| * @param buf 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, |
| const uint8_t *buf, int len) |
| { |
| unsigned int ssrc, h; |
| int payload_type, seq, ret, flags = 0; |
| AVStream *st; |
| uint32_t timestamp; |
| int rv= 0; |
| |
| if (!buf) { |
| /* return the next packets, if any */ |
| if(s->st && s->parse_packet) { |
| timestamp= 0; ///< Should not be used if buf is NULL, but should be set to the timestamp of the packet returned.... |
| rv= s->parse_packet(s->ic, s->dynamic_protocol_context, |
| s->st, pkt, ×tamp, NULL, 0, flags); |
| finalize_packet(s, pkt, timestamp); |
| return rv; |
| } else { |
| // TODO: Move to a dynamic packet handler (like above) |
| if (s->read_buf_index >= s->read_buf_size) |
| return -1; |
| ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index, |
| s->read_buf_size - s->read_buf_index); |
| if (ret < 0) |
| return -1; |
| s->read_buf_index += ret; |
| if (s->read_buf_index < s->read_buf_size) |
| return 1; |
| else |
| return 0; |
| } |
| } |
| |
| if (len < 12) |
| return -1; |
| |
| if ((buf[0] & 0xc0) != (RTP_VERSION << 6)) |
| return -1; |
| if (buf[1] >= 200 && buf[1] <= 204) { |
| rtcp_parse_packet(s, buf, len); |
| return -1; |
| } |
| 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; |
| |
| if (!st) { |
| /* specific MPEG2TS demux support */ |
| ret = mpegts_parse_packet(s->ts, pkt, buf, len); |
| if (ret < 0) |
| 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; |
| return 1; |
| } |
| } 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: |
| /* 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; |
| // 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_AAC: |
| if (rtp_parse_mp4_au(s, buf)) |
| return -1; |
| { |
| RTPPayloadData *infos = s->rtp_payload_data; |
| if (infos == NULL) |
| return -1; |
| buf += infos->au_headers_length_bytes + 2; |
| len -= infos->au_headers_length_bytes + 2; |
| |
| /* XXX: Fixme we only handle the case where rtp_parse_mp4_au define |
| one au_header */ |
| av_new_packet(pkt, infos->au_headers[0].size); |
| memcpy(pkt->data, buf, infos->au_headers[0].size); |
| buf += infos->au_headers[0].size; |
| len -= infos->au_headers[0].size; |
| } |
| s->read_buf_size = len; |
| rv= 0; |
| break; |
| default: |
| av_new_packet(pkt, len); |
| memcpy(pkt->data, buf, len); |
| break; |
| } |
| |
| // now perform timestamp things.... |
| finalize_packet(s, pkt, timestamp); |
| } |
| return rv; |
| } |
| |
| void rtp_parse_close(RTPDemuxContext *s) |
| { |
| // TODO: fold this into the protocol specific data fields. |
| if (!strcmp(ff_rtp_enc_name(s->payload_type), "MP2T")) { |
| mpegts_parse_close(s->ts); |
| } |
| av_free(s); |
| } |
