Mercurial > libavformat.hg
changeset 2891:a6c922b05571 libavformat
Split the RTP demuxing functions out of rtp.c, to simplify the RTP muxer's dependencies
| author | lucabe |
|---|---|
| date | Fri, 04 Jan 2008 19:33:50 +0000 |
| parents | 31199df992f8 |
| children | 0d82fdf4fa94 |
| files | Makefile rtp.c rtpdec.c |
| diffstat | 3 files changed, 556 insertions(+), 526 deletions(-) [+] |
line wrap: on
line diff
--- a/Makefile Fri Jan 04 17:52:16 2008 +0000 +++ b/Makefile Fri Jan 04 19:33:50 2008 +0000 @@ -121,9 +121,9 @@ OBJS-$(CONFIG_RM_MUXER) += rmenc.o OBJS-$(CONFIG_ROQ_DEMUXER) += idroq.o OBJS-$(CONFIG_ROQ_MUXER) += raw.o -OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o rtsp.o rtp_mpv.o rtp_aac.o +OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_mpv.o rtp_aac.o OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o -OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o rtp.o rtp_h264.o rtp_mpv.o rtp_aac.o +OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o rtp.o rtpdec.o rtp_h264.o rtp_mpv.o rtp_aac.o OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o OBJS-$(CONFIG_SHORTEN_DEMUXER) += raw.o OBJS-$(CONFIG_SIFF_DEMUXER) += siff.o
--- a/rtp.c Fri Jan 04 17:52:16 2008 +0000 +++ b/rtp.c Fri Jan 04 19:33:50 2008 +0000 @@ -26,7 +26,6 @@ #include "network.h" #include "rtp_internal.h" -#include "rtp_h264.h" #include "rtp_mpv.h" #include "rtp_aac.h" @@ -34,15 +33,6 @@ #define RTCP_SR_SIZE 28 -/* 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') -*/ - /* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ AVRtpPayloadType_t AVRtpPayloadTypes[]= { @@ -178,25 +168,6 @@ {-1, "", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1} }; -/* 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}; - -static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) -{ - handler->next= RTPFirstDynamicPayloadHandler; - RTPFirstDynamicPayloadHandler= handler; -} - -void av_register_rtp_dynamic_payload_handlers(void) -{ - register_dynamic_payload_handler(&mp4v_es_handler); - register_dynamic_payload_handler(&mpeg4_generic_handler); - register_dynamic_payload_handler(&ff_h264_dynamic_handler); -} - int rtp_get_codec_info(AVCodecContext *codec, int payload_type) { int i = 0; @@ -255,501 +226,6 @@ return CODEC_ID_NONE; } -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; -#if defined(DEBUG) - printf("sending %d bytes of RR\n", len); -#endif - 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; -} - -/** - * 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, rtp_payload_data_t *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 { - 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: - break; - } - } - // needed to send back RTCP RR in RTSP sessions - s->rtp_ctx = rtpc; - gethostname(s->hostname, sizeof(s->hostname)); - return s; -} - -static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) -{ - int au_headers_length, au_header_size, i; - GetBitContext getbitcontext; - rtp_payload_data_t *infos; - - infos = s->rtp_payload_data; - - if (infos == NULL) - return -1; - - /* decode the first 2 bytes where are stored the AUHeader sections - 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) -{ - switch(s->st->codec->codec_id) { - case CODEC_ID_MP2: - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: - if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { - int64_t addend; - - int delta_timestamp; - /* XXX: is it really necessary to unify the timestamp base ? */ - /* compute pts from timestamp with received ntp_time */ - delta_timestamp = timestamp - s->last_rtcp_timestamp; - /* convert to 90 kHz without overflow */ - addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; - addend = (addend * 5625) >> 14; - pkt->pts = addend + delta_timestamp; - } - break; - case CODEC_ID_AAC: - case CODEC_ID_H264: - case CODEC_ID_MPEG4: - pkt->pts = timestamp; - break; - default: - /* no timestamp info yet */ - break; - } - 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; - 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, pkt, ×tamp, NULL, 0); - 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; - 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 { - // 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; - { - rtp_payload_data_t *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: - if(s->parse_packet) { - rv= s->parse_packet(s, pkt, ×tamp, buf, len); - } else { - 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); -} - /* rtp output */ static int rtp_write_header(AVFormatContext *s1)
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/rtpdec.c Fri Jan 04 19:33:50 2008 +0000 @@ -0,0 +1,554 @@ +/* + * 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 + */ +#include "avformat.h" +#include "mpegts.h" +#include "bitstream.h" + +#include <unistd.h> +#include "network.h" + +#include "rtp_internal.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}; + +static void register_dynamic_payload_handler(RTPDynamicProtocolHandler *handler) +{ + handler->next= RTPFirstDynamicPayloadHandler; + RTPFirstDynamicPayloadHandler= handler; +} + +void av_register_rtp_dynamic_payload_handlers(void) +{ + register_dynamic_payload_handler(&mp4v_es_handler); + register_dynamic_payload_handler(&mpeg4_generic_handler); + 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; +#if defined(DEBUG) + printf("sending %d bytes of RR\n", len); +#endif + 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; +} + +/** + * 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, rtp_payload_data_t *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 { + 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: + break; + } + } + // needed to send back RTCP RR in RTSP sessions + s->rtp_ctx = rtpc; + gethostname(s->hostname, sizeof(s->hostname)); + return s; +} + +static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf) +{ + int au_headers_length, au_header_size, i; + GetBitContext getbitcontext; + rtp_payload_data_t *infos; + + infos = s->rtp_payload_data; + + if (infos == NULL) + return -1; + + /* decode the first 2 bytes where are stored the AUHeader sections + 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) +{ + switch(s->st->codec->codec_id) { + case CODEC_ID_MP2: + case CODEC_ID_MPEG1VIDEO: + case CODEC_ID_MPEG2VIDEO: + if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) { + int64_t addend; + + int delta_timestamp; + /* XXX: is it really necessary to unify the timestamp base ? */ + /* compute pts from timestamp with received ntp_time */ + delta_timestamp = timestamp - s->last_rtcp_timestamp; + /* convert to 90 kHz without overflow */ + addend = (s->last_rtcp_ntp_time - s->first_rtcp_ntp_time) >> 14; + addend = (addend * 5625) >> 14; + pkt->pts = addend + delta_timestamp; + } + break; + case CODEC_ID_AAC: + case CODEC_ID_H264: + case CODEC_ID_MPEG4: + pkt->pts = timestamp; + break; + default: + /* no timestamp info yet */ + break; + } + 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; + 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, pkt, ×tamp, NULL, 0); + 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; + 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 { + // 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; + { + rtp_payload_data_t *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: + if(s->parse_packet) { + rv= s->parse_packet(s, pkt, ×tamp, buf, len); + } else { + 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); +}