我们将视频编码为 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。
有两种方法可以将 AAC 放入传输流。
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 音频与 ADTS 传输语法）。
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 音频与 LATM 传输语法）。
并且您可以使用所有强力的“新”(MPEG4) AAC 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。
但是经过大量搜索后，我找不到任何有关如何执行其中任何一项的文档。在将其传递到 MPEG-TS 多路复用器（用于输出到 HLS）之前，使用 ADTS 或 LATM 获取编码音频的配置中缺少什么？
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