X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=83167eba9c3aef04a17cd244eba9f201e97bd327;hb=0048deb84cb6d22ba7f4fd7c8b4ecc054fcc22d4;hp=41d584381ba5e5703e341f07cd9a21a8d6cab26e;hpb=984b914c55fe480985e702ce945e2f88835c21fe;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 41d584381ba..83167eba9c3 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -28,12 +28,12 @@ #include "rtpenc.h" -//#define DEBUG - static const AVOption options[] = { - FF_RTP_FLAG_OPTS(RTPMuxContext, flags) - { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.dbl = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM }, - { "max_packet_size", "Max packet size", offsetof(RTPMuxContext, max_packet_size), AV_OPT_TYPE_INT, {.dbl = 0 }, 0, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM }, + 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 }, }; @@ -46,34 +46,38 @@ static const AVClass rtp_muxer_class = { #define RTCP_SR_SIZE 28 -static int is_supported(enum CodecID id) +static int is_supported(enum AVCodecID id) { switch(id) { - case CODEC_ID_H263: - case CODEC_ID_H263P: - case CODEC_ID_H264: - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: - case CODEC_ID_MPEG4: - case CODEC_ID_AAC: - case CODEC_ID_MP2: - case CODEC_ID_MP3: - case CODEC_ID_PCM_ALAW: - case CODEC_ID_PCM_MULAW: - case CODEC_ID_PCM_S8: - case CODEC_ID_PCM_S16BE: - case CODEC_ID_PCM_S16LE: - case CODEC_ID_PCM_U16BE: - case CODEC_ID_PCM_U16LE: - case CODEC_ID_PCM_U8: - case CODEC_ID_MPEG2TS: - case CODEC_ID_AMR_NB: - case CODEC_ID_AMR_WB: - case CODEC_ID_VORBIS: - case CODEC_ID_THEORA: - case CODEC_ID_VP8: - case CODEC_ID_ADPCM_G722: - case CODEC_ID_ADPCM_G726: + case AV_CODEC_ID_H263: + case AV_CODEC_ID_H263P: + case AV_CODEC_ID_H264: + 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; @@ -97,42 +101,66 @@ static int rtp_write_header(AVFormatContext *s1) return -1; } - if (s->payload_type < 0) - s->payload_type = ff_rtp_get_payload_type(s1, st->codec); + 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; - s->ssrc = av_get_random_seed(); + 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) /* Round the NTP time to whole milliseconds. */ s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + NTP_OFFSET_US; - - if (s->max_packet_size) { + // 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) + 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) - s->max_packet_size = FFMIN(s->max_packet_size, - s1->pb->max_packet_size); + s1->packet_size = FFMIN(s1->packet_size, + s1->pb->max_packet_size); } else - s->max_packet_size = s1->pb->max_packet_size; - if (s->max_packet_size <= 12) { - av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s->max_packet_size); + 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(s->max_packet_size); + s->buf = av_malloc(s1->packet_size); if (s->buf == NULL) { return AVERROR(ENOMEM); } - s->max_payload_size = s->max_packet_size - 12; + s->max_payload_size = s1->packet_size - 12; s->max_frames_per_packet = 0; - if (s1->max_delay) { + if (s1->max_delay > 0) { if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { - if (st->codec->frame_size == 0) { + 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_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * (int64_t)st->codec->frame_size, AV_ROUND_DOWN); + 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) { @@ -143,60 +171,76 @@ static int rtp_write_header(AVFormatContext *s1) avpriv_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { - case CODEC_ID_MP2: - case CODEC_ID_MP3: + case AV_CODEC_ID_MP2: + case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; break; - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: break; - case CODEC_ID_MPEG2TS: + 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 CODEC_ID_H264: + 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 CODEC_ID_VORBIS: - case CODEC_ID_THEORA: + 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 CODEC_ID_VP8: - av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is " - "incompatible with the latest spec drafts.\n"); - break; - case CODEC_ID_ADPCM_G722: + 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 CODEC_ID_AMR_NB: - case CODEC_ID_AMR_WB: + 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 == CODEC_ID_AMR_NB) + 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"); - return -1; + goto fail; } if (st->codec->channels != 1) { av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); - return -1; + goto fail; } - case CODEC_ID_AAC: + case AV_CODEC_ID_AAC: s->num_frames = 0; default: defaultcase: @@ -208,10 +252,14 @@ defaultcase: } return 0; + +fail: + av_freep(&s->buf); + return AVERROR(EINVAL); } /* send an rtcp sender report packet */ -static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) +static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time, int bye) { RTPMuxContext *s = s1->priv_data; uint32_t rtp_ts; @@ -221,15 +269,37 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) 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, 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_wb32(s1->pb, ntp_time / 1000000); - avio_wb32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); + 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); } @@ -242,7 +312,7 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) 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, 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); @@ -251,15 +321,15 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) avio_write(s1->pb, buf1, len); avio_flush(s1->pb); - s->seq++; + 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 void rtp_send_samples(AVFormatContext *s1, - const uint8_t *buf1, int size, int sample_size_bits) +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; @@ -269,7 +339,7 @@ static void rtp_send_samples(AVFormatContext *s1, 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) - av_abort(); + return AVERROR(EINVAL); n = 0; while (size > 0) { s->buf_ptr = s->buf; @@ -284,6 +354,7 @@ static void rtp_send_samples(AVFormatContext *s1, 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, @@ -383,6 +454,36 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1, } } +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; @@ -397,61 +498,57 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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()); + 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 CODEC_ID_PCM_MULAW: - case CODEC_ID_PCM_ALAW: - case CODEC_ID_PCM_U8: - case CODEC_ID_PCM_S8: - rtp_send_samples(s1, pkt->data, size, 8 * 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, pkt->data, size, 16 * st->codec->channels); - break; - case CODEC_ID_ADPCM_G722: + 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. */ - rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); - break; - case CODEC_ID_ADPCM_G726: - rtp_send_samples(s1, pkt->data, size, - st->codec->bits_per_coded_sample * st->codec->channels); - break; - case CODEC_ID_MP2: - case CODEC_ID_MP3: + 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 CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: ff_rtp_send_mpegvideo(s1, pkt->data, size); break; - case CODEC_ID_AAC: + 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 CODEC_ID_AMR_NB: - case CODEC_ID_AMR_WB: + case AV_CODEC_ID_AMR_NB: + case AV_CODEC_ID_AMR_WB: ff_rtp_send_amr(s1, pkt->data, size); break; - case CODEC_ID_MPEG2TS: + case AV_CODEC_ID_MPEG2TS: rtp_send_mpegts_raw(s1, pkt->data, size); break; - case CODEC_ID_H264: + case AV_CODEC_ID_H264: ff_rtp_send_h264(s1, pkt->data, size); break; - case CODEC_ID_H263: + case AV_CODEC_ID_H263: if (s->flags & FF_RTP_FLAG_RFC2190) { int mb_info_size = 0; const uint8_t *mb_info = @@ -461,16 +558,30 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) break; } /* Fallthrough */ - case CODEC_ID_H263P: + case AV_CODEC_ID_H263P: ff_rtp_send_h263(s1, pkt->data, size); break; - case CODEC_ID_VORBIS: - case CODEC_ID_THEORA: + case AV_CODEC_ID_VORBIS: + case AV_CODEC_ID_THEORA: ff_rtp_send_xiph(s1, pkt->data, size); break; - case CODEC_ID_VP8: + 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); @@ -483,6 +594,10 @@ 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; @@ -490,12 +605,12 @@ static int rtp_write_trailer(AVFormatContext *s1) AVOutputFormat ff_rtp_muxer = { .name = "rtp", - .long_name = NULL_IF_CONFIG_SMALL("RTP output format"), + .long_name = NULL_IF_CONFIG_SMALL("RTP output"), .priv_data_size = sizeof(RTPMuxContext), - .audio_codec = CODEC_ID_PCM_MULAW, - .video_codec = CODEC_ID_MPEG4, + .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, + .priv_class = &rtp_muxer_class, };