X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtp.c;h=0a3c411f51e9125f1449b9c0f54e5a59a86d2781;hb=c3386bd5b4d3662f94e902a0fe3e9e869e29967d;hp=359b1f2c9e923b25b2262fa94f737ba3d1768583;hpb=fead30d4440bc7b75006ae60f2742c63a05168b3;p=ffmpeg diff --git a/libavformat/rtp.c b/libavformat/rtp.c index 359b1f2c9e9..0a3c411f51e 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -1,1095 +1,151 @@ /* * RTP input/output format - * Copyright (c) 2002 Fabrice Bellard. + * Copyright (c) 2002 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; 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 -#include -#include -#include -#include -#include - -#include "rtp_internal.h" -#include "rtp_h264.h" -//#define DEBUG - - -/* TODO: - add RTCP statistics reporting (should be optional). +#include "libavutil/opt.h" +#include "avformat.h" - - 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') -*/ +#include "rtp.h" /* 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; -} - -/* return < 0 if unknown payload type */ -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 - -/** - * some rtp servers assume client is dead if they don't hear from them... - * so we send a Receiver Report to the provided ByteIO context - * (we don't have access to the rtcp handle from here) - */ -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. +/* payload types >= 96 are dynamic; + * payload types between 72 and 76 are reserved for RTCP conflict avoidance; + * all the other payload types not present in the table are unassigned or + * reserved */ -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; -} +static const struct { + int pt; + const char enc_name[6]; + enum AVMediaType codec_type; + enum AVCodecID codec_id; + int clock_rate; + int audio_channels; +} rtp_payload_types[] = { + {0, "PCMU", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_MULAW, 8000, 1}, + {3, "GSM", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1}, + {4, "G723", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_G723_1, 8000, 1}, + {5, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1}, + {6, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 16000, 1}, + {7, "LPC", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1}, + {8, "PCMA", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_ALAW, 8000, 1}, + {9, "G722", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_ADPCM_G722, 8000, 1}, + {10, "L16", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_S16BE, 44100, 2}, + {11, "L16", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_PCM_S16BE, 44100, 1}, + {12, "QCELP", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_QCELP, 8000, 1}, + {13, "CN", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1}, + {14, "MPA", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_MP2, -1, -1}, + {14, "MPA", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_MP3, -1, -1}, + {15, "G728", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1}, + {16, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 11025, 1}, + {17, "DVI4", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 22050, 1}, + {18, "G729", AVMEDIA_TYPE_AUDIO, AV_CODEC_ID_NONE, 8000, 1}, + {25, "CelB", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_NONE, 90000, -1}, + {26, "JPEG", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_MJPEG, 90000, -1}, + {28, "nv", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_NONE, 90000, -1}, + {31, "H261", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_H261, 90000, -1}, + {32, "MPV", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_MPEG1VIDEO, 90000, -1}, + {32, "MPV", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_MPEG2VIDEO, 90000, -1}, + {33, "MP2T", AVMEDIA_TYPE_DATA, AV_CODEC_ID_MPEG2TS, 90000, -1}, + {34, "H263", AVMEDIA_TYPE_VIDEO, AV_CODEC_ID_H263, 90000, -1}, + {-1, "", AVMEDIA_TYPE_UNKNOWN, AV_CODEC_ID_NONE, -1, -1} +}; -/** - * 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) +int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type) { - unsigned int ssrc, h; - int payload_type, seq, ret; - AVStream *st; - uint32_t timestamp; - int rv= 0; + int i = 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 + for (i = 0; rtp_payload_types[i].pt >= 0; i++) + if (rtp_payload_types[i].pt == payload_type) { + if (rtp_payload_types[i].codec_id != AV_CODEC_ID_NONE) { + codec->codec_type = rtp_payload_types[i].codec_type; + codec->codec_id = rtp_payload_types[i].codec_id; + if (rtp_payload_types[i].audio_channels > 0) + codec->channels = rtp_payload_types[i].audio_channels; + if (rtp_payload_types[i].clock_rate > 0) + codec->sample_rate = rtp_payload_types[i].clock_rate; 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; } - } + return -1; } -/* 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; +int ff_rtp_get_payload_type(AVFormatContext *fmt, + AVCodecContext *codec, int idx) +{ + int i; + AVOutputFormat *ofmt = fmt ? fmt->oformat : NULL; + + /* Was the payload type already specified for the RTP muxer? */ + if (ofmt && ofmt->priv_class && fmt->priv_data) { + int64_t payload_type; + if (av_opt_get_int(fmt->priv_data, "payload_type", 0, &payload_type) >= 0 && + payload_type >= 0) + return (int)payload_type; + } + + /* static payload type */ + for (i = 0; rtp_payload_types[i].pt >= 0; ++i) + if (rtp_payload_types[i].codec_id == codec->codec_id) { + if (codec->codec_id == AV_CODEC_ID_H263 && (!fmt || !fmt->oformat || + !fmt->oformat->priv_class || !fmt->priv_data || + !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190"))) + continue; + /* G722 has 8000 as nominal rate even if the sample rate is 16000, + * see section 4.5.2 in RFC 3551. */ + if (codec->codec_id == AV_CODEC_ID_ADPCM_G722 && + codec->sample_rate == 16000 && codec->channels == 1) + return rtp_payload_types[i].pt; + if (codec->codec_type == AVMEDIA_TYPE_AUDIO && + ((rtp_payload_types[i].clock_rate > 0 && + codec->sample_rate != rtp_payload_types[i].clock_rate) || + (rtp_payload_types[i].audio_channels > 0 && + codec->channels != rtp_payload_types[i].audio_channels))) + continue; + return rtp_payload_types[i].pt; } - 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 (idx < 0) + idx = codec->codec_type == AVMEDIA_TYPE_AUDIO; - 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++; + /* dynamic payload type */ + return RTP_PT_PRIVATE + idx; } -static void rtp_send_raw(AVFormatContext *s1, - const uint8_t *buf1, int size) +const char *ff_rtp_enc_name(int payload_type) { - RTPDemuxContext *s = s1->priv_data; - AVStream *st = s1->streams[0]; - int len, max_packet_size; - - max_packet_size = s->max_payload_size; + int i; - while (size > 0) { - len = max_packet_size; - if (len > size) - len = size; + for (i = 0; rtp_payload_types[i].pt >= 0; i++) + if (rtp_payload_types[i].pt == payload_type) + return rtp_payload_types[i].enc_name; - /* 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++; + return ""; } -/* 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) +enum AVCodecID ff_rtp_codec_id(const char *buf, enum AVMediaType codec_type) { - RTPDemuxContext *s = s1->priv_data; - int len, out_len; + int i; - 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; + for (i = 0; rtp_payload_types[i].pt >= 0; i++) + if (!strcmp(buf, rtp_payload_types[i].enc_name) && (codec_type == rtp_payload_types[i].codec_type)) + return rtp_payload_types[i].codec_id; - 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; - } - } + return AV_CODEC_ID_NONE; } - -/* 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, -};