将 AAC 的 avcodec 配置配置为 HLS/MPEG-TS（[mpegts @ 0x7fc4c00343c0] AAC 比特流不是 ADTS 格式并且缺少额外数据）
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【中文标题】将 AAC 的 avcodec 配置配置为 HLS/MPEG-TS（[mpegts @ 0x7fc4c00343c0] AAC 比特流不是 ADTS 格式并且缺少额外数据）【英文标题】：avcodec config for AAC into HLS/MPEG-TS ([mpegts @ 0x7fc4c00343c0] AAC bitstream not in ADTS format and extradata missing) 【发布时间】：2021-04-08 20:30:28 【问题描述】：我们正在将视频编码为 H264，并将原始 PCM 样本编码为 AAC，以用于 HLS 流式传输。视频运行良好，但在 libavcodec 中配置 AAC 编码器时遇到问题。
This SO question 说：
有两种方法可以将 AAC 放入传输流中。
1.使用 ADTS 语法（MPEG2 风格）。
在这种情况下，PMT 的 stream_type 应指定为 0x0F（ISO/IEC 13818-7 带有 ADTS 传输语法的音频）。
因此，您只能使用“旧”(MPEG2) AAC 版本，而不能使用 SBR 和 PS。
2。使用 LATM+LOAS/AudiosyncStream 语法（MPEG4 风格）。
在这种情况下，PMT 的 stream_type 应指定为 0x11（ISO/IEC 14496-3 使用 LATM 传输语法的音频）。
您可以使用“新”(MPEG4) AAC 功能的所有力量，包括 SBR 和 PS。
此外，DVB 标准 ETSI TS 101 154 要求：HEv1/HEv2 AAC 应 使用 LATM 语法传输。
但经过大量搜索后，我找不到任何有关如何执行其中任何一项的文档。在将编码音频传递到 MPEG-TS 多路复用器（用于输出到 HLS）之前，下面的配置缺少什么来获取带有 ADTS 或 LATM 的编码音频？
设置 AAC 编解码器的当前代码给出错误[mpegts @ 0x7fc4c00343c0] AAC bitstream not in ADTS format and extradata missing
AAC 编码器设置（为简洁起见，删除了错误检查）
/// Set up Encoder ///
mpAudioCodec = avcodec_find_encoder(AV_CODEC_ID_AAC);
mpAudioCodecContext = avcodec_alloc_context3(mpAudioCodec);
mpAudioCodecContext->bit_rate = DEFAULT_AUD_BITRATE;
mpAudioCodecContext->sample_rate = DEFAULT_AUD_SAMPLE_RATE;
mpAudioCodecContext->channel_layout = DEFAULT_AUD_CHAN_LAYOUT;
mpAudioCodecContext->channels = 2;
mpAudioCodecContext->sample_fmt = AV_SAMPLE_FMT_FLTP; // S16 not supported. Must convert
mpAudioCodecContext->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
rc = avcodec_open2(mpAudioCodecContext, mpAudioCodec, 0);
HLS 多路复用器设置
avformat_alloc_output_context2(&mpOutputMux, 0, "hls", path.c_str());
// VIDEO TRACK
mpVideoTrack = avformat_new_stream(mpOutputMux, 0);
mpVideoTrack->id = 0;
mpVideoTrack->codecpar->codec_type = AVMEDIA_TYPE_VIDEO;
mpVideoTrack->codecpar->codec_id = AV_CODEC_ID_H264;
mpVideoTrack->time_base = (AVRational) 1, mFrameRate ;
mpVideoTrack->avg_frame_rate = (AVRational) mFrameRate, 1 ;
// AUDIO TRACK
mpAudioTrack = avformat_new_stream(mpOutputMux, 0);
mpAudioTrack->id = 1;
mpAudioTrack->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
mpAudioTrack->codecpar->codec_id = DEFAULT_AUDIO_CODEC;
mpAudioTrack->codecpar->sample_rate = mpAudioCodecContext->sample_rate;
mpAudioTrack->time_base.den = mpAudioCodecContext->sample_rate;
mpAudioTrack->time_base.num = 1;
AVDictionary *hlsOptions = NULL;
av_dict_set(&hlsOptions, "hls_segment_type", "mpegts", 0);
av_dict_set(&hlsOptions, "segment_list_type", "m3u8", 0);
av_dict_set_int(&hlsOptions, "hls_list_size", mPlaylistSize, 0);
av_dict_set_int(&hlsOptions, "hls_time", mChunkDurSec, 0);
av_dict_set(&hlsOptions, "hls_flags", "delete_segments", 0);
av_dict_set(&hlsOptions, "hls_segment_filename", segPath.c_str(), 0);
av_dict_set_int(&hlsOptions, "reference_stream", mpVideoTrack->index, 0);
av_dict_set(&hlsOptions, "segment_list_flags", "cache+live", 0);
int ret = avformat_write_header(mpOutputMux, &hlsOptions);
编码循环
int bytesCopied = mAudEsBuffer.popData(mpPcmS16Buf, mpPcmAudioFrame->nb_samples);
// resample to float
int rc = swr_convert(mpAudioResampleCtx, mpPcmAudioFrame->data, mpPcmAudioFrame->nb_samples, (const uint8_t**) &mpPcmS16Buf, mpPcmAudioFrame->nb_samples);
/* Set a timestamp based on the sample rate for the container. */
mCurAudPts += mpPcmAudioFrame->nb_samples;
mpPcmAudioFrame->pts = mCurAudPts;
// send frame for encoding to AAC
rc = avcodec_send_frame(mpAudioCodecContext, mpPcmAudioFrame);
/* read all the available output packets (in general there may be any number of them */
while (rc >= 0)
// need to init packet every time??
/* Set the packet data and size so that it is recognized as being empty. */
av_init_packet(mpEncAudioPacket);
mpEncAudioPacket->data = NULL;
mpEncAudioPacket->size = 0;
rc = avcodec_receive_packet(mpAudioCodecContext, mpEncAudioPacket);
if (rc < 0)
printf("TqHlsLib::readAndMuxAudio() - Error encoding audio frame: %s\n", av_make_error_string(mpErr, TQERRLEN, rc));
return HLS_DEC_ERROR;
TRACE(("%T %t TqHlsLib::readAndMuxAudio() - Got an encoded audio packet. %u bytes\n",
mpEncAudioPacket->size ));
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(mpEncAudioPacket, mpAudioTrack->time_base, mpAudioTrack->time_base);
mpEncAudioPacket->stream_index = mpAudioTrack->index;
/* Write the compressed frame to the media file. */
rc = av_interleaved_write_frame(mpOutputMux, mpEncAudioPacket);
if (rc < 0)
fprintf(stderr, "TqHlsLib::addVideoH264Packet - Error while writing audio packet: %s\n",
av_make_error_string(mpErr, TQERRLEN, ret));
// return some error here
av_packet_unref(mpEncAudioPacket);
输出
[mpegts @ 0x7fb280144e00] AAC bitstream not in ADTS format and extradata missing
20:24:52.327418 24388 TqHlsLib::readAndMuxAudio() - Got an encoded audio packet. 185 bytes
[mpegts @ 0x7fb280144e00] AAC bitstream not in ADTS format and extradata missing
20:24:52.372975 24388 TqHlsLib::readAndMuxAudio() - Got an encoded audio packet. 188 bytes
[mpegts @ 0x7fb280144e00] AAC bitstream not in ADTS format and extradata missing
【问题讨论】：
【参考方案1】：经过大量搜索，我找到了两种将 ADTS 标头添加到音频的方法，以便正确地 MUX MPEG-TS。
第一个是为 ADTS 设置一个单独的AVFormatContext，使用编码的 AAC 数据包创建一个AVStream，然后使用send_frame / receive_frame 获取相同的 AAC 数据，但这次使用 ADTS已应用标头。
这增加了很多复杂性和延迟。
最后，我只是在每个编码的 AAC 数据包中手动添加了一个 ADTS 标头，然后再传递给av_interleaved_write_frame。
为了帮助以后的人，这里是代码：
/*
ADTS HEADER: 7 Bytes. See ISO 13818-7 (2004)
AAAAAAAA AAAABCCD EEFFFFGH HHIJKLMM MMMMMMMM MMMOOOOO OOOOOOPP
A - Sync 0xFFFx
B 1 MPEG Version: 0 for MPEG-4, 1 for MPEG-2
C 2 Layer: always 0
D 1 protection absent, Warning, set to 1 if there is no CRC and 0 if there is CRC
E 2 profile, the MPEG-4 Audio Object Type minus 1
F 4 MPEG-4 Sampling Frequency Index (15 is forbidden)
G 1 private bit, guaranteed never to be used by MPEG, set to 0 when encoding, ignore when decoding
H 3 MPEG-4 Channel Configuration (in the case of 0, the channel configuration is sent via an inband PCE)
I 1 originality, set to 0 when encoding, ignore when decoding
J 1 home, set to 0 when encoding, ignore when decoding
K 1 copyrighted id bit, the next bit of a centrally registered copyright identifier, set to 0 when encoding, ignore when decoding
L 1 copyright id start, signals that this frame's copyright id bit is the first bit of the copyright id, set to 0 when encoding, ignore when decoding
M 13 frame length, this value must include 7 or 9 bytes of header length: FrameLength = (ProtectionAbsent == 1 ? 7 : 9) + size(AACFrame)
O 11 Buffer fullness
P 2 Number of AAC frames (RDBs) in ADTS frame minus 1, for maximum compatibility always use 1 AAC frame per ADTS frame
Q 16 CRC if protection absent is 0
*/
void
addADTSHeader(uint8_t *inBuf, size_t inLen, const ADTSInfo &adtsInfo, std::vector<uint8_t> &outVec)
outVec.clear();
uint8_t b;
// 0: Sync I
outVec.push_back(0xFF);
// 1: Sync II + BCCD
b = 0xF0;
b |= ((adtsInfo.mpegVersion == 2) ? 0 : 1) << 3;
b |= 1; // no protection
outVec.push_back(b);
// 2: EEFFFFGH
int sampleFreqIdx = 4; // default 44100
int i = 0;
while (ADTS_SAMPLE_RATES[i] != LAST_INDEX)
if (ADTS_SAMPLE_RATES[i] == adtsInfo.sampleRate)
sampleFreqIdx = i;
break;
i++;
b = (adtsInfo.objectType - 1) << 6; // EE
b |= sampleFreqIdx << 2;
b |= (adtsInfo.channelConfig & 0x07) >> 2; // high bit of channel config
outVec.push_back(b);
// 3: HHIJKLMM
b = (adtsInfo.channelConfig & 0x03) << 6; // low two bits of channel config
b |= ((7 + inLen) >> 11) & 0x03; // high two bits of frame length
outVec.push_back(b);
// 4: MMMMMMMM
b = (7 + inLen) >> 3; // Frame length middle
outVec.push_back(b);
// 5: MMMOOOOO
b = ((7 + inLen) & 0x07) << 5; // low three bits of frame length
b |= 0x1F; // buffer fullness 0x7FF VBR -> N/A
outVec.push_back(b);
// 6: OOOOOOPP
b = 0x3F << 2; // buffer fullness 0x7FF VBR -> N/A
b |= 0; // 1 AAC frame per ADTS
outVec.push_back(b);
outVec.insert(outVec.end(), inBuf, inBuf + inLen);
【讨论】：
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