将 AAC 的 avcodec 配置转换为 HLS/MPEG-TS[mpegts @ 0x7fc4c00343c0] AAC 比特流不是 ADTS 格式且缺少额外数据

如何解决将 AAC 的 avcodec 配置转换为 HLS/MPEG-TS[mpegts @ 0x7fc4c00343c0] AAC 比特流不是 ADTS 格式且缺少额外数据

我们正在将视频编码为 H264，并将原始 PCM 样本编码为 AAC 以进行 HLS 流式传输。视频工作正常，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 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）。

并且您可以使用“新”(MPEG4) AAC 功能的所有功能，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以获取带有 ADTS 或 LATM 的编码音频？

设置 AAC 编解码器的当前代码给出了错误 [mpegts @ 0x7fc4c00343c0] AAC bitstream not in ADTS format and exTradata missing

AAC 编码器设置（为简洁起见，已删除错误检查）

/// 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 MUX 设置

avformat_alloc_output_context2(&mpOutputMux,"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,"hls_time",mChunkDurSec,"hls_flags","delete_segments","hls_segment_filename",segPath.c_str(),0);

av_dict_set_int(&hlsOptions,"reference_stream",mpVideoTrack->index,"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",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