diff options
Diffstat (limited to 'contrib/ffmpeg/libavformat/rtp.c')
| -rw-r--r-- | contrib/ffmpeg/libavformat/rtp.c | 1085 |
1 files changed, 1085 insertions, 0 deletions
diff --git a/contrib/ffmpeg/libavformat/rtp.c b/contrib/ffmpeg/libavformat/rtp.c new file mode 100644 index 000000000..493a89cf3 --- /dev/null +++ b/contrib/ffmpeg/libavformat/rtp.c @@ -0,0 +1,1085 @@ +/* + * RTP input/output 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') +*/ + +/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */ +AVRtpPayloadType_t AVRtpPayloadTypes[]= +{ + {0, "PCMU", CODEC_TYPE_AUDIO, CODEC_ID_PCM_MULAW, 8000, 1}, + {1, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {2, "Reserved", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {3, "GSM", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {4, "G723", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {5, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {6, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 16000, 1}, + {7, "LPC", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {8, "PCMA", CODEC_TYPE_AUDIO, CODEC_ID_PCM_ALAW, 8000, 1}, + {9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2}, + {11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1}, + {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1}, + {13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1}, + {15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {16, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 11025, 1}, + {17, "DVI4", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 22050, 1}, + {18, "G729", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1}, + {19, "reserved", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, + {20, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, + {21, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, + {22, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, + {23, "unassigned", CODEC_TYPE_AUDIO, CODEC_ID_NONE, -1, -1}, + {24, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, + {25, "CelB", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, + {26, "JPEG", CODEC_TYPE_VIDEO, CODEC_ID_MJPEG, 90000, -1}, + {27, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, + {28, "nv", CODEC_TYPE_VIDEO, CODEC_ID_NONE, 90000, -1}, + {29, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, + {30, "unassigned", CODEC_TYPE_VIDEO, CODEC_ID_NONE, -1, -1}, + {31, "H261", CODEC_TYPE_VIDEO, CODEC_ID_H261, 90000, -1}, + {32, "MPV", CODEC_TYPE_VIDEO, CODEC_ID_MPEG1VIDEO, 90000, -1}, + {33, "MP2T", CODEC_TYPE_DATA, CODEC_ID_MPEG2TS, 90000, -1}, + {34, "H263", CODEC_TYPE_VIDEO, CODEC_ID_H263, 90000, -1}, + {35, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {36, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {37, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {38, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {39, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {40, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {41, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {42, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {43, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {44, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {45, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {46, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {47, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {48, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {49, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {50, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {51, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {52, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {53, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {54, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {55, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {56, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {57, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {58, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {59, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {60, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {61, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {62, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {63, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {64, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {65, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {66, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {67, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {68, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {69, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {70, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {71, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {72, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {73, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {74, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {75, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {76, "reserved for RTCP conflict avoidance", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {77, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {78, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {79, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {80, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {81, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {82, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {83, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {84, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {85, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {86, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {87, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {88, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {89, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {90, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {91, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {92, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {93, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {94, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {95, "unassigned", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {96, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {97, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {98, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {99, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {100, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {101, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {102, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {103, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {104, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {105, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {106, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {107, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {108, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {109, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {110, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {111, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {112, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {113, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {114, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {115, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {116, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {117, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {118, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {119, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {120, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {121, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {122, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {123, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {124, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {125, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {126, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {127, "dynamic", CODEC_TYPE_UNKNOWN, CODEC_ID_NONE, -1, -1}, + {-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() +{ + 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) +{ + if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) { + codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type; + codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id; + if (AVRtpPayloadTypes[payload_type].audio_channels > 0) + codec->channels = AVRtpPayloadTypes[payload_type].audio_channels; + if (AVRtpPayloadTypes[payload_type].clock_rate > 0) + codec->sample_rate = AVRtpPayloadTypes[payload_type].clock_rate; + return 0; + } + return -1; +} + +int rtp_get_payload_type(AVCodecContext *codec) +{ + int i, payload_type; + + /* compute the payload type */ + for (payload_type = -1, i = 0; AVRtpPayloadTypes[i].pt >= 0; ++i) + if (AVRtpPayloadTypes[i].codec_id == codec->codec_id) { + if (codec->codec_id == CODEC_ID_PCM_S16BE) + if (codec->channels != AVRtpPayloadTypes[i].audio_channels) + continue; + payload_type = AVRtpPayloadTypes[i].pt; + } + return payload_type; +} + +static inline uint32_t decode_be32(const uint8_t *p) +{ + return (p[0] << 24) | (p[1] << 16) | (p[2] << 8) | p[3]; +} + +static inline uint64_t decode_be64(const uint8_t *p) +{ + return ((uint64_t)decode_be32(p) << 32) | decode_be32(p + 4); +} + +static int rtcp_parse_packet(RTPDemuxContext *s, const unsigned char *buf, int len) +{ + if (buf[1] != 200) + return -1; + s->last_rtcp_ntp_time = decode_be64(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 = decode_be32(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(AVRtpPayloadTypes[payload_type].enc_name, "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 = 1; + 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 doesnt 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: + 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 = (buf[2] << 8) | buf[3]; + timestamp = decode_be32(buf + 4); + ssrc = decode_be32(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 = decode_be32(buf); + len -= 4; + buf += 4; + av_new_packet(pkt, len); + memcpy(pkt->data, buf, len); + break; + case CODEC_ID_MPEG1VIDEO: + /* better than nothing: skip mpeg video RTP header */ + if (len <= 4) + return -1; + h = decode_be32(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; + s->buf_ptr = buf; + 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(AVRtpPayloadTypes[s->payload_type].enc_name, "MP2T")) { + mpegts_parse_close(s->ts); + } + av_free(s); +} + +/* rtp output */ + +static int rtp_write_header(AVFormatContext *s1) +{ + RTPDemuxContext *s = s1->priv_data; + int payload_type, max_packet_size, n; + AVStream *st; + + if (s1->nb_streams != 1) + return -1; + st = s1->streams[0]; + + payload_type = rtp_get_payload_type(st->codec); + if (payload_type < 0) + payload_type = RTP_PT_PRIVATE; /* private payload type */ + s->payload_type = payload_type; + +// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly + s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ + s->timestamp = s->base_timestamp; + s->ssrc = 0; /* FIXME: was random(), what should this be? */ + s->first_packet = 1; + + max_packet_size = url_fget_max_packet_size(&s1->pb); + if (max_packet_size <= 12) + return AVERROR_IO; + s->max_payload_size = max_packet_size - 12; + + switch(st->codec->codec_id) { + case CODEC_ID_MP2: + case CODEC_ID_MP3: + s->buf_ptr = s->buf + 4; + s->cur_timestamp = 0; + break; + case CODEC_ID_MPEG1VIDEO: + s->cur_timestamp = 0; + break; + case CODEC_ID_MPEG2TS: + n = s->max_payload_size / TS_PACKET_SIZE; + if (n < 1) + n = 1; + s->max_payload_size = n * TS_PACKET_SIZE; + s->buf_ptr = s->buf; + break; + default: + s->buf_ptr = s->buf; + break; + } + + return 0; +} + +/* send an rtcp sender report packet */ +static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) +{ + RTPDemuxContext *s = s1->priv_data; +#if defined(DEBUG) + printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); +#endif + put_byte(&s1->pb, (RTP_VERSION << 6)); + put_byte(&s1->pb, 200); + put_be16(&s1->pb, 6); /* length in words - 1 */ + put_be32(&s1->pb, s->ssrc); + put_be64(&s1->pb, ntp_time); + put_be32(&s1->pb, s->timestamp); + put_be32(&s1->pb, s->packet_count); + put_be32(&s1->pb, s->octet_count); + put_flush_packet(&s1->pb); +} + +/* send an rtp packet. sequence number is incremented, but the caller + must update the timestamp itself */ +static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) +{ + RTPDemuxContext *s = s1->priv_data; + +#ifdef DEBUG + printf("rtp_send_data size=%d\n", len); +#endif + + /* build the RTP header */ + put_byte(&s1->pb, (RTP_VERSION << 6)); + put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); + put_be16(&s1->pb, s->seq); + put_be32(&s1->pb, s->timestamp); + put_be32(&s1->pb, s->ssrc); + + put_buffer(&s1->pb, buf1, len); + put_flush_packet(&s1->pb); + + s->seq++; + s->octet_count += len; + s->packet_count++; +} + +/* send an integer number of samples and compute time stamp and fill + the rtp send buffer before sending. */ +static void rtp_send_samples(AVFormatContext *s1, + const uint8_t *buf1, int size, int sample_size) +{ + RTPDemuxContext *s = s1->priv_data; + int len, max_packet_size, n; + + max_packet_size = (s->max_payload_size / sample_size) * sample_size; + /* not needed, but who nows */ + if ((size % sample_size) != 0) + av_abort(); + while (size > 0) { + len = (max_packet_size - (s->buf_ptr - s->buf)); + if (len > size) + len = size; + + /* copy data */ + memcpy(s->buf_ptr, buf1, len); + s->buf_ptr += len; + buf1 += len; + size -= len; + n = (s->buf_ptr - s->buf); + /* if buffer full, then send it */ + if (n >= max_packet_size) { + rtp_send_data(s1, s->buf, n, 0); + s->buf_ptr = s->buf; + /* update timestamp */ + s->timestamp += n / sample_size; + } + } +} + +/* NOTE: we suppose that exactly one frame is given as argument here */ +/* XXX: test it */ +static void rtp_send_mpegaudio(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPDemuxContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int len, count, max_packet_size; + + max_packet_size = s->max_payload_size; + + /* test if we must flush because not enough space */ + len = (s->buf_ptr - s->buf); + if ((len + size) > max_packet_size) { + if (len > 4) { + rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); + s->buf_ptr = s->buf + 4; + /* 90 KHz time stamp */ + s->timestamp = s->base_timestamp + + (s->cur_timestamp * 90000LL) / st->codec->sample_rate; + } + } + + /* add the packet */ + if (size > max_packet_size) { + /* big packet: fragment */ + count = 0; + while (size > 0) { + len = max_packet_size - 4; + if (len > size) + len = size; + /* build fragmented packet */ + s->buf[0] = 0; + s->buf[1] = 0; + s->buf[2] = count >> 8; + s->buf[3] = count; + memcpy(s->buf + 4, buf1, len); + rtp_send_data(s1, s->buf, len + 4, 0); + size -= len; + buf1 += len; + count += len; + } + } else { + if (s->buf_ptr == s->buf + 4) { + /* no fragmentation possible */ + s->buf[0] = 0; + s->buf[1] = 0; + s->buf[2] = 0; + s->buf[3] = 0; + } + memcpy(s->buf_ptr, buf1, size); + s->buf_ptr += size; + } + s->cur_timestamp += st->codec->frame_size; +} + +/* NOTE: a single frame must be passed with sequence header if + needed. XXX: use slices. */ +static void rtp_send_mpegvideo(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPDemuxContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int len, h, max_packet_size; + uint8_t *q; + + max_packet_size = s->max_payload_size; + + while (size > 0) { + /* XXX: more correct headers */ + h = 0; + if (st->codec->sub_id == 2) + h |= 1 << 26; /* mpeg 2 indicator */ + q = s->buf; + *q++ = h >> 24; + *q++ = h >> 16; + *q++ = h >> 8; + *q++ = h; + + if (st->codec->sub_id == 2) { + h = 0; + *q++ = h >> 24; + *q++ = h >> 16; + *q++ = h >> 8; + *q++ = h; + } + + len = max_packet_size - (q - s->buf); + if (len > size) + len = size; + + memcpy(q, buf1, len); + q += len; + + /* 90 KHz time stamp */ + s->timestamp = s->base_timestamp + + av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps + rtp_send_data(s1, s->buf, q - s->buf, (len == size)); + + buf1 += len; + size -= len; + } + s->cur_timestamp++; +} + +static void rtp_send_raw(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPDemuxContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int len, max_packet_size; + + max_packet_size = s->max_payload_size; + + while (size > 0) { + len = max_packet_size; + if (len > size) + len = size; + + /* 90 KHz time stamp */ + s->timestamp = s->base_timestamp + + av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps + rtp_send_data(s1, buf1, len, (len == size)); + + buf1 += len; + size -= len; + } + s->cur_timestamp++; +} + +/* NOTE: size is assumed to be an integer multiple of TS_PACKET_SIZE */ +static void rtp_send_mpegts_raw(AVFormatContext *s1, + const uint8_t *buf1, int size) +{ + RTPDemuxContext *s = s1->priv_data; + int len, out_len; + + while (size >= TS_PACKET_SIZE) { + len = s->max_payload_size - (s->buf_ptr - s->buf); + if (len > size) + len = size; + memcpy(s->buf_ptr, buf1, len); + buf1 += len; + size -= len; + s->buf_ptr += len; + + out_len = s->buf_ptr - s->buf; + if (out_len >= s->max_payload_size) { + rtp_send_data(s1, s->buf, out_len, 0); + s->buf_ptr = s->buf; + } + } +} + +/* write an RTP packet. 'buf1' must contain a single specific frame. */ +static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) +{ + RTPDemuxContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int rtcp_bytes; + int64_t ntp_time; + int size= pkt->size; + uint8_t *buf1= pkt->data; + +#ifdef DEBUG + printf("%d: write len=%d\n", pkt->stream_index, size); +#endif + + /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ + rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / + RTCP_TX_RATIO_DEN; + if (s->first_packet || rtcp_bytes >= 28) { + /* compute NTP time */ + /* XXX: 90 kHz timestamp hardcoded */ + ntp_time = (pkt->pts << 28) / 5625; + rtcp_send_sr(s1, ntp_time); + s->last_octet_count = s->octet_count; + s->first_packet = 0; + } + + switch(st->codec->codec_id) { + case CODEC_ID_PCM_MULAW: + case CODEC_ID_PCM_ALAW: + case CODEC_ID_PCM_U8: + case CODEC_ID_PCM_S8: + rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); + break; + case CODEC_ID_PCM_U16BE: + case CODEC_ID_PCM_U16LE: + case CODEC_ID_PCM_S16BE: + case CODEC_ID_PCM_S16LE: + rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); + break; + case CODEC_ID_MP2: + case CODEC_ID_MP3: + rtp_send_mpegaudio(s1, buf1, size); + break; + case CODEC_ID_MPEG1VIDEO: + rtp_send_mpegvideo(s1, buf1, size); + break; + case CODEC_ID_MPEG2TS: + rtp_send_mpegts_raw(s1, buf1, size); + break; + default: + /* better than nothing : send the codec raw data */ + rtp_send_raw(s1, buf1, size); + break; + } + return 0; +} + +static int rtp_write_trailer(AVFormatContext *s1) +{ + // RTPDemuxContext *s = s1->priv_data; + return 0; +} + +AVOutputFormat rtp_muxer = { + "rtp", + "RTP output format", + NULL, + NULL, + sizeof(RTPDemuxContext), + CODEC_ID_PCM_MULAW, + CODEC_ID_NONE, + rtp_write_header, + rtp_write_packet, + rtp_write_trailer, +}; |