X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=28c309ea184853552d2e9549d782c9fac0298953;hb=a670eea56087d0ecd4fbeccf3a9beb9110b7031f;hp=eab765499b416d09f8e8e98c550e411c37332317;hpb=e6153f173a49e5bfa70b0c04d2f82930533597b9;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index eab765499b4..28c309ea184 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, {.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 }, }; @@ -49,9 +49,11 @@ static const AVClass rtp_muxer_class = { 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: @@ -75,6 +77,9 @@ static int is_supported(enum AVCodecID id) 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; @@ -84,7 +89,7 @@ static int is_supported(enum AVCodecID id) static int rtp_write_header(AVFormatContext *s1) { RTPMuxContext *s = s1->priv_data; - int n; + int n, ret = AVERROR(EINVAL); AVStream *st; if (s1->nb_streams != 1) { @@ -92,14 +97,23 @@ static int rtp_write_header(AVFormatContext *s1) return AVERROR(EINVAL); } st = s1->streams[0]; - if (!is_supported(st->codec->codec_id)) { - av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); + if (!is_supported(st->codecpar->codec_id)) { + av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codecpar->codec_id); 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->codecpar, -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; @@ -111,6 +125,13 @@ static int rtp_write_header(AVFormatContext *s1) /* 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) + 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) @@ -119,42 +140,26 @@ static int rtp_write_header(AVFormatContext *s1) } 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); + av_log(s1, AV_LOG_ERROR, "Max packet size %u too low\n", s1->packet_size); return AVERROR(EIO); } s->buf = av_malloc(s1->packet_size); - if (s->buf == NULL) { + 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... */ - s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); - } + if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { + avpriv_set_pts_info(st, 32, 1, st->codecpar->sample_rate); + } else { + avpriv_set_pts_info(st, 32, 1, 90000); } - - avpriv_set_pts_info(st, 32, 1, 90000); - switch(st->codec->codec_id) { + s->buf_ptr = s->buf; + switch(st->codecpar->codec_id) { case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; + avpriv_set_pts_info(st, 32, 1, 90000); break; case AV_CODEC_ID_MPEG1VIDEO: case AV_CODEC_ID_MPEG2VIDEO: @@ -164,45 +169,63 @@ static int rtp_write_header(AVFormatContext *s1) 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 H.261 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; + if (st->codecpar->extradata_size > 4 && st->codecpar->extradata[0] == 1) { + s->nal_length_size = (st->codecpar->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/H.264 case above, instead + * of trying to handle the pre-standardization versions (as in + * libavcodec/hevc.c). */ + if (st->codecpar->extradata_size > 21 && st->codecpar->extradata[0] == 1) { + s->nal_length_size = (st->codecpar->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_VP8: - av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is " - "incompatible with the latest spec drafts.\n"); + s->max_frames_per_packet = 15; break; 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->codecpar->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) { + if (st->codecpar->block_align != 38 && st->codecpar->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; + s->max_frames_per_packet = s->max_payload_size / st->codecpar->block_align; + break; 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) + s->max_frames_per_packet = 50; + if (st->codecpar->codec_id == AV_CODEC_ID_AMR_NB) n = 31; else n = 61; @@ -211,18 +234,15 @@ static int rtp_write_header(AVFormatContext *s1) av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); goto fail; } - if (st->codec->channels != 1) { + if (st->codecpar->channels != 1) { av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); goto fail; } + break; case AV_CODEC_ID_AAC: - s->num_frames = 0; + s->max_frames_per_packet = 50; + break; 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; } @@ -230,29 +250,52 @@ defaultcase: fail: av_freep(&s->buf); - return AVERROR(EINVAL); + return ret; } /* 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; - av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); + av_log(s1, AV_LOG_TRACE, "RTCP: %02x %"PRIx64" %"PRIx32"\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, 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); } @@ -262,10 +305,10 @@ 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); + av_log(s1, AV_LOG_TRACE, "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); @@ -274,7 +317,7 @@ 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++; } @@ -390,6 +433,7 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1, 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) @@ -411,23 +455,28 @@ 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 frame_duration = av_get_audio_frame_duration2(st->codecpar, 0); + int frame_size = st->codecpar->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 > 0 && + av_compare_ts(s->cur_timestamp - s->timestamp, st->time_base, + s1->max_delay, AV_TIME_BASE_Q) >= 0) { + ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1); + s->num_frames = 0; + } 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; + memcpy(s->buf_ptr, buf, frame_size); + frames--; + s->num_frames++; + s->buf_ptr += frame_size; + buf += frame_size; + s->cur_timestamp += frame_duration; if (s->num_frames == s->max_frames_per_packet) { ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1); @@ -444,39 +493,39 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) int rtcp_bytes; int size= pkt->size; - av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size); + av_log(s1, AV_LOG_TRACE, "%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()); + 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) { + switch(st->codecpar->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); + return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->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); + return rtp_send_samples(s1, pkt->data, size, 16 * st->codecpar->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); + return rtp_send_samples(s1, pkt->data, size, 8 * st->codecpar->channels); case AV_CODEC_ID_ADPCM_G726: return rtp_send_samples(s1, pkt->data, size, - st->codec->bits_per_coded_sample * st->codec->channels); + st->codecpar->bits_per_coded_sample * st->codecpar->channels); case AV_CODEC_ID_MP2: case AV_CODEC_ID_MP3: rtp_send_mpegaudio(s1, pkt->data, size); @@ -499,7 +548,10 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) rtp_send_mpegts_raw(s1, pkt->data, size); break; case AV_CODEC_ID_H264: - ff_rtp_send_h264(s1, pkt->data, size); + 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) { @@ -514,6 +566,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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); @@ -524,6 +579,17 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) 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); @@ -536,6 +602,10 @@ 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; @@ -551,4 +621,5 @@ AVOutputFormat ff_rtp_muxer = { .write_packet = rtp_write_packet, .write_trailer = rtp_write_trailer, .priv_class = &rtp_muxer_class, + .flags = AVFMT_TS_NONSTRICT, };