X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=28c309ea184853552d2e9549d782c9fac0298953;hb=6ccf76aec73b2cd598bb1e65d126d8a12540c411;hp=647a8073b7f5b570fc2debd2bd7764437b45bb60;hpb=ddf5fb71ee9c8b2d9a23c0f661a84896cd7050fc;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 647a8073b7f..28c309ea184 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -49,6 +49,7 @@ 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: @@ -88,7 +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) { @@ -96,8 +97,8 @@ 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; } @@ -105,7 +106,7 @@ static int rtp_write_header(AVFormatContext *s1) 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); + st->id = ff_rtp_get_payload_type(s1, st->codecpar, -1); s->payload_type = st->id; } else { @@ -139,7 +140,7 @@ 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); @@ -148,38 +149,17 @@ static int rtp_write_header(AVFormatContext *s1) } 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; - } + 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: @@ -189,37 +169,44 @@ 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/H264 case above, instead + * 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->codec->extradata_size > 21 && st->codec->extradata[0] == 1) { - s->nal_length_size = (st->codec->extradata[21] & 0x03) + 1; + 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; + 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->codec->channels > 2) { + if (st->codecpar->channels > 2) { av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n"); goto fail; } @@ -229,20 +216,16 @@ static int rtp_write_header(AVFormatContext *s1) 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; @@ -251,18 +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; } @@ -270,7 +250,7 @@ defaultcase: fail: av_freep(&s->buf); - return AVERROR(EINVAL); + return ret; } /* send an rtcp sender report packet */ @@ -279,7 +259,8 @@ 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}, @@ -324,7 +305,7 @@ 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); @@ -452,6 +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) @@ -473,23 +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); @@ -506,7 +493,7 @@ 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; @@ -519,26 +506,26 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) } 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); @@ -561,7 +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) { @@ -577,7 +567,7 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) ff_rtp_send_h263(s1, pkt->data, size); break; case AV_CODEC_ID_HEVC: - ff_rtp_send_hevc(s1, pkt->data, size); + ff_rtp_send_h264_hevc(s1, pkt->data, size); break; case AV_CODEC_ID_VORBIS: case AV_CODEC_ID_THEORA: @@ -631,4 +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, };