/* * RTP 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 "internal.h" #include "libavutil/mathematics.h" #include "libavutil/random_seed.h" #include "libavutil/opt.h" #include "rtpenc.h" static const AVOption options[] = { FF_RTP_FLAG_OPTS(RTPMuxContext, flags), { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM }, { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM }, { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM }, { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM }, { NULL }, }; static const AVClass rtp_muxer_class = { .class_name = "RTP muxer", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, }; #define RTCP_SR_SIZE 28 static int is_supported(enum AVCodecID id) { switch(id) { case AV_CODEC_ID_H261: case AV_CODEC_ID_H263: case AV_CODEC_ID_H263P: case AV_CODEC_ID_H264: case AV_CODEC_ID_HEVC: case AV_CODEC_ID_MPEG1VIDEO: case AV_CODEC_ID_MPEG2VIDEO: case AV_CODEC_ID_MPEG4: case AV_CODEC_ID_AAC: case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: case AV_CODEC_ID_PCM_ALAW: case AV_CODEC_ID_PCM_MULAW: case AV_CODEC_ID_PCM_S8: case AV_CODEC_ID_PCM_S16BE: case AV_CODEC_ID_PCM_S16LE: case AV_CODEC_ID_PCM_U16BE: case AV_CODEC_ID_PCM_U16LE: case AV_CODEC_ID_PCM_U8: case AV_CODEC_ID_MPEG2TS: case AV_CODEC_ID_AMR_NB: case AV_CODEC_ID_AMR_WB: case AV_CODEC_ID_VORBIS: case AV_CODEC_ID_THEORA: case AV_CODEC_ID_VP8: case AV_CODEC_ID_ADPCM_G722: case AV_CODEC_ID_ADPCM_G726: case AV_CODEC_ID_ILBC: case AV_CODEC_ID_MJPEG: case AV_CODEC_ID_SPEEX: case AV_CODEC_ID_OPUS: return 1; default: return 0; } } static int rtp_write_header(AVFormatContext *s1) { RTPMuxContext *s = s1->priv_data; int n, ret = AVERROR(EINVAL); AVStream *st; if (s1->nb_streams != 1) { av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n"); return AVERROR(EINVAL); } st = s1->streams[0]; if (!is_supported(st->codec->codec_id)) { av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id)); return -1; } if (s->payload_type < 0) { /* Re-validate non-dynamic payload types */ if (st->id < RTP_PT_PRIVATE) st->id = ff_rtp_get_payload_type(s1, st->codec, -1); s->payload_type = st->id; } else { /* private option takes priority */ st->id = s->payload_type; } s->base_timestamp = av_get_random_seed(); s->timestamp = s->base_timestamp; s->cur_timestamp = 0; if (!s->ssrc) s->ssrc = av_get_random_seed(); s->first_packet = 1; s->first_rtcp_ntp_time = ff_ntp_time(); if (s1->start_time_realtime != 0 && s1->start_time_realtime != AV_NOPTS_VALUE) /* Round the NTP time to whole milliseconds. */ s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + NTP_OFFSET_US; // Pick a random sequence start number, but in the lower end of the // available range, so that any wraparound doesn't happen immediately. // (Immediate wraparound would be an issue for SRTP.) if (s->seq < 0) { if (s1->flags & AVFMT_FLAG_BITEXACT) { s->seq = 0; } else s->seq = av_get_random_seed() & 0x0fff; } else s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval if (s1->packet_size) { if (s1->pb->max_packet_size) s1->packet_size = FFMIN(s1->packet_size, s1->pb->max_packet_size); } else s1->packet_size = s1->pb->max_packet_size; if (s1->packet_size <= 12) { av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size); return AVERROR(EIO); } s->buf = av_malloc(s1->packet_size); if (!s->buf) { return AVERROR(ENOMEM); } s->max_payload_size = s1->packet_size - 12; s->max_frames_per_packet = 0; if (s1->max_delay > 0) { if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { int frame_size = av_get_audio_frame_duration(st->codec, 0); if (!frame_size) frame_size = st->codec->frame_size; if (frame_size == 0) { av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); } else { s->max_frames_per_packet = av_rescale_q_rnd(s1->max_delay, AV_TIME_BASE_Q, (AVRational){ frame_size, st->codec->sample_rate }, AV_ROUND_DOWN); } } if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { /* FIXME: We should round down here... */ if (st->avg_frame_rate.num > 0 && st->avg_frame_rate.den > 0) { s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, av_inv_q(st->avg_frame_rate)); } else s->max_frames_per_packet = 1; } } avpriv_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; break; case AV_CODEC_ID_MPEG1VIDEO: case AV_CODEC_ID_MPEG2VIDEO: break; case AV_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; case AV_CODEC_ID_H261: if (s1->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) { av_log(s, AV_LOG_ERROR, "Packetizing H261 is experimental and produces incorrect " "packetization for cases where GOBs don't fit into packets " "(even though most receivers may handle it just fine). " "Please set -f_strict experimental in order to enable it.\n"); ret = AVERROR_EXPERIMENTAL; goto fail; } break; case AV_CODEC_ID_H264: /* check for H.264 MP4 syntax */ if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1; } break; case AV_CODEC_ID_HEVC: /* Only check for the standardized hvcC version of extradata, keeping * things simple and similar to the avcC/H264 case above, instead * of trying to handle the pre-standardization versions (as in * libavcodec/hevc.c). */ if (st->codec->extradata_size > 21 && st->codec->extradata[0] == 1) { s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1; } break; case AV_CODEC_ID_VORBIS: case AV_CODEC_ID_THEORA: if (!s->max_frames_per_packet) s->max_frames_per_packet = 15; s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length s->num_frames = 0; goto defaultcase; case AV_CODEC_ID_ADPCM_G722: /* Due to a historical error, the clock rate for G722 in RTP is * 8000, even if the sample rate is 16000. See RFC 3551. */ avpriv_set_pts_info(st, 32, 1, 8000); break; case AV_CODEC_ID_OPUS: if (st->codec->channels > 2) { av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n"); goto fail; } /* The opus RTP RFC says that all opus streams should use 48000 Hz * as clock rate, since all opus sample rates can be expressed in * this clock rate, and sample rate changes on the fly are supported. */ avpriv_set_pts_info(st, 32, 1, 48000); break; case AV_CODEC_ID_ILBC: if (st->codec->block_align != 38 && st->codec->block_align != 50) { av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n"); goto fail; } if (!s->max_frames_per_packet) s->max_frames_per_packet = 1; s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, s->max_payload_size / st->codec->block_align); goto defaultcase; case AV_CODEC_ID_AMR_NB: case AV_CODEC_ID_AMR_WB: if (!s->max_frames_per_packet) s->max_frames_per_packet = 12; if (st->codec->codec_id == AV_CODEC_ID_AMR_NB) n = 31; else n = 61; /* max_header_toc_size + the largest AMR payload must fit */ if (1 + s->max_frames_per_packet + n > s->max_payload_size) { av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); goto fail; } if (st->codec->channels != 1) { av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); goto fail; } s->num_frames = 0; goto defaultcase; case AV_CODEC_ID_AAC: s->num_frames = 0; goto defaultcase; default: defaultcase: if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; break; } return 0; fail: av_freep(&s->buf); return ret; } /* send an rtcp sender report packet */ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye) { RTPMuxContext *s = s1->priv_data; uint32_t rtp_ts; av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); s->last_rtcp_ntp_time = ntp_time; rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, s1->streams[0]->time_base) + s->base_timestamp; avio_w8(s1->pb, RTP_VERSION << 6); avio_w8(s1->pb, RTCP_SR); avio_wb16(s1->pb, 6); /* length in words - 1 */ avio_wb32(s1->pb, s->ssrc); avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time)); avio_wb32(s1->pb, rtp_ts); avio_wb32(s1->pb, s->packet_count); avio_wb32(s1->pb, s->octet_count); if (s->cname) { int len = FFMIN(strlen(s->cname), 255); avio_w8(s1->pb, (RTP_VERSION << 6) + 1); avio_w8(s1->pb, RTCP_SDES); avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */ avio_wb32(s1->pb, s->ssrc); avio_w8(s1->pb, 0x01); /* CNAME */ avio_w8(s1->pb, len); avio_write(s1->pb, s->cname, len); avio_w8(s1->pb, 0); /* END */ for (len = (7 + len) % 4; len % 4; len++) avio_w8(s1->pb, 0); } if (bye) { avio_w8(s1->pb, (RTP_VERSION << 6) | 1); avio_w8(s1->pb, RTCP_BYE); avio_wb16(s1->pb, 1); /* length in words - 1 */ avio_wb32(s1->pb, s->ssrc); } avio_flush(s1->pb); } /* send an rtp packet. sequence number is incremented, but the caller must update the timestamp itself */ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) { RTPMuxContext *s = s1->priv_data; av_dlog(s1, "rtp_send_data size=%d\n", len); /* build the RTP header */ avio_w8(s1->pb, RTP_VERSION << 6); avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); avio_wb16(s1->pb, s->seq); avio_wb32(s1->pb, s->timestamp); avio_wb32(s1->pb, s->ssrc); avio_write(s1->pb, buf1, len); avio_flush(s1->pb); s->seq = (s->seq + 1) & 0xffff; 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 int rtp_send_samples(AVFormatContext *s1, const uint8_t *buf1, int size, int sample_size_bits) { RTPMuxContext *s = s1->priv_data; int len, max_packet_size, n; /* Calculate the number of bytes to get samples aligned on a byte border */ int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8); max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size; /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */ if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0) return AVERROR(EINVAL); n = 0; while (size > 0) { s->buf_ptr = s->buf; len = FFMIN(max_packet_size, size); /* copy data */ memcpy(s->buf_ptr, buf1, len); s->buf_ptr += len; buf1 += len; size -= len; s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits; ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); n += (s->buf_ptr - s->buf); } return 0; } static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size) { RTPMuxContext *s = s1->priv_data; 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) { ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); s->buf_ptr = s->buf + 4; } } if (s->buf_ptr == s->buf + 4) { s->timestamp = s->cur_timestamp; } /* 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); ff_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; } } static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size) { RTPMuxContext *s = s1->priv_data; int len, max_packet_size; max_packet_size = s->max_payload_size; while (size > 0) { len = max_packet_size; if (len > size) len = size; s->timestamp = s->cur_timestamp; ff_rtp_send_data(s1, buf1, len, (len == size)); buf1 += len; size -= len; } } /* 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) { RTPMuxContext *s = s1->priv_data; int len, out_len; s->timestamp = s->cur_timestamp; 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) { ff_rtp_send_data(s1, s->buf, out_len, 0); s->buf_ptr = s->buf; } } } static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size) { RTPMuxContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int frame_duration = av_get_audio_frame_duration(st->codec, 0); int frame_size = st->codec->block_align; int frames = size / frame_size; while (frames > 0) { int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames); if (!s->num_frames) { s->buf_ptr = s->buf; s->timestamp = s->cur_timestamp; } memcpy(s->buf_ptr, buf, n * frame_size); frames -= n; s->num_frames += n; s->buf_ptr += n * frame_size; buf += n * frame_size; s->cur_timestamp += n * frame_duration; if (s->num_frames == s->max_frames_per_packet) { ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1); s->num_frames = 0; } } return 0; } static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) { RTPMuxContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int rtcp_bytes; int size= pkt->size; av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size); rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / RTCP_TX_RATIO_DEN; if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) && !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) { rtcp_send_sr(s1, ff_ntp_time(), 0); s->last_octet_count = s->octet_count; s->first_packet = 0; } s->cur_timestamp = s->base_timestamp + pkt->pts; switch(st->codec->codec_id) { case AV_CODEC_ID_PCM_MULAW: case AV_CODEC_ID_PCM_ALAW: case AV_CODEC_ID_PCM_U8: case AV_CODEC_ID_PCM_S8: return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); case AV_CODEC_ID_PCM_U16BE: case AV_CODEC_ID_PCM_U16LE: case AV_CODEC_ID_PCM_S16BE: case AV_CODEC_ID_PCM_S16LE: return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels); case AV_CODEC_ID_ADPCM_G722: /* The actual sample size is half a byte per sample, but since the * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, * the correct parameter for send_samples_bits is 8 bits per stream * clock. */ return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); case AV_CODEC_ID_ADPCM_G726: return rtp_send_samples(s1, pkt->data, size, st->codec->bits_per_coded_sample * st->codec->channels); case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: rtp_send_mpegaudio(s1, pkt->data, size); break; case AV_CODEC_ID_MPEG1VIDEO: case AV_CODEC_ID_MPEG2VIDEO: ff_rtp_send_mpegvideo(s1, pkt->data, size); break; case AV_CODEC_ID_AAC: if (s->flags & FF_RTP_FLAG_MP4A_LATM) ff_rtp_send_latm(s1, pkt->data, size); else ff_rtp_send_aac(s1, pkt->data, size); break; case AV_CODEC_ID_AMR_NB: case AV_CODEC_ID_AMR_WB: ff_rtp_send_amr(s1, pkt->data, size); break; case AV_CODEC_ID_MPEG2TS: rtp_send_mpegts_raw(s1, pkt->data, size); break; case AV_CODEC_ID_H264: ff_rtp_send_h264_hevc(s1, pkt->data, size); break; case AV_CODEC_ID_H261: ff_rtp_send_h261(s1, pkt->data, size); break; case AV_CODEC_ID_H263: if (s->flags & FF_RTP_FLAG_RFC2190) { int mb_info_size = 0; const uint8_t *mb_info = av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO, &mb_info_size); if (!mb_info) { av_log(s1, AV_LOG_ERROR, "failed to allocate side data\n"); return AVERROR(ENOMEM); } ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size); break; } /* Fallthrough */ case AV_CODEC_ID_H263P: ff_rtp_send_h263(s1, pkt->data, size); break; case AV_CODEC_ID_HEVC: ff_rtp_send_h264_hevc(s1, pkt->data, size); break; case AV_CODEC_ID_VORBIS: case AV_CODEC_ID_THEORA: ff_rtp_send_xiph(s1, pkt->data, size); break; case AV_CODEC_ID_VP8: ff_rtp_send_vp8(s1, pkt->data, size); break; case AV_CODEC_ID_ILBC: rtp_send_ilbc(s1, pkt->data, size); break; case AV_CODEC_ID_MJPEG: ff_rtp_send_jpeg(s1, pkt->data, size); break; case AV_CODEC_ID_OPUS: if (size > s->max_payload_size) { av_log(s1, AV_LOG_ERROR, "Packet size %d too large for max RTP payload size %d\n", size, s->max_payload_size); return AVERROR(EINVAL); } /* Intentional fallthrough */ default: /* better than nothing : send the codec raw data */ rtp_send_raw(s1, pkt->data, size); break; } return 0; } static int rtp_write_trailer(AVFormatContext *s1) { RTPMuxContext *s = s1->priv_data; /* If the caller closes and recreates ->pb, this might actually * be NULL here even if it was successfully allocated at the start. */ if (s1->pb && (s->flags & FF_RTP_FLAG_SEND_BYE)) rtcp_send_sr(s1, ff_ntp_time(), 1); av_freep(&s->buf); return 0; } AVOutputFormat ff_rtp_muxer = { .name = "rtp", .long_name = NULL_IF_CONFIG_SMALL("RTP output"), .priv_data_size = sizeof(RTPMuxContext), .audio_codec = AV_CODEC_ID_PCM_MULAW, .video_codec = AV_CODEC_ID_MPEG4, .write_header = rtp_write_header, .write_packet = rtp_write_packet, .write_trailer = rtp_write_trailer, .priv_class = &rtp_muxer_class, .flags = AVFMT_TS_NONSTRICT, };