将音频样本转换为 AVFrame FFmpeg & C++

Posted 2023-02-22

【中文标题】将音频样本转换为 AVFrame FFmpeg & C++

【英文标题】：Convert audio samples to AVFrame FFmpeg & C++

【发布时间】：2021-01-30 12:17:35

【问题描述】：

我真的需要帮助来纠正我目前正在使用的方法。该方法应将样本转换并写入ac3文件。

输入样本是 AV_SAMPLE_FMT_FLT 格式的 BYTE*

对于编码器，样本必须具有 AV_SAMPLE_FMT_FLTP 格式

bool AddAudiosample(AVFormatContext * pFormatContext, AVStream * pStream, BYTE * audiodata, uint32_t sampleCount, uint64_t devicets)

    AVCodecContext * pCodecCxt = NULL;
    bool res = true;

    pCodecCxt = pStream->codec;

    AVFrame*  pFLTAudioFrame = NULL;
    pFLTAudioFrame = av_frame_alloc();

    AVFrame*  pFLTPAudioFrame = NULL;
    pFLTPAudioFrame = av_frame_alloc();

    ProcessData(pFLTAudioFrame, pFLTPAudioFrame, (uint8_t *)audiodata, sampleCount, devicets);

    swr_convert(pSmplConvertCtx, pFLTPAudioFrame->data, pFLTPAudioFrame->nb_samples, (const uint8_t **)pFLTAudioFrame->data, pFLTAudioFrame->nb_samples);

    AVPacket pkt;
    av_init_packet(&pkt);

    pkt.flags |= AV_PKT_FLAG_KEY;
    pkt.stream_index = pStream->index;
    pkt.data = pAudioEncodeBuffer;
    pkt.size = pFLTPAudioFrame->pkt_size;

    int gotOutput = 0;
    auto ret = avcodec_encode_audio2(pCodecCxt, &pkt, pFLTPAudioFrame, &gotOutput);
    if (ret < 0)
    
        exit(1);
    
    if (gotOutput)
    
        pkt.pts = av_rescale_q(pCodecCxt->coded_frame->pts, pCodecCxt->time_base, pStream->time_base);
        ret = av_interleaved_write_frame(pFormatContext, &pkt);
        if (ret < 0)
        
            exit(1);
        
    

    return res;


void ProcessData(AVFrame *inputframe, AVFrame *outputFrame, uint8_t* data, uint32_t sample_count, uint64_t device_ts)

    inputframe->nb_samples = sample_count;
    inputframe->format = AV_SAMPLE_FMT_FLT;
    inputframe->sample_rate = mWFX->nSamplesPerSec;
    inputframe->channels = mWFX->nChannels;
    inputframe->pkt_size = sample_count*mWFX->nBlockAlign;

    av_samples_fill_arrays(inputframe->data, inputframe->linesize, data, mWFX->nChannels, sample_count, AV_SAMPLE_FMT_FLT, 1);

    outputFrame->nb_samples = inputframe->nb_samples;
    outputFrame->format = AV_SAMPLE_FMT_FLTP;
    outputFrame->sample_rate = inputframe->sample_rate;
    outputFrame->channels = inputframe->channels;
    outputFrame->pkt_size = sample_count*mWFX->nBlockAlign;

    av_samples_fill_arrays(outputFrame->data, outputFrame->linesize, pAudioEncodeBuffer, mWFX->nChannels, sample_count, AV_SAMPLE_FMT_FLTP, 1);

这里是缓冲包的上下文重采样设置和计算：

    pSmplConvertCtx = swr_alloc();
    if (!pSmplConvertCtx)
    
         fprintf(stderr, "Could not allocate resampler context\n");
         exit(1);
    

    av_opt_set_int       (pSmplConvertCtx, "in_channel_count",   pCodecCxt->channels,       0);
    av_opt_set_int       (pSmplConvertCtx, "in_sample_rate",     pCodecCxt->sample_rate,    0);
    av_opt_set_sample_fmt(pSmplConvertCtx, "in_sample_fmt",      AV_SAMPLE_FMT_FLT,         0);
    av_opt_set_int       (pSmplConvertCtx, "out_channel_count",  pCodecCxt->channels,       0);
    av_opt_set_int       (pSmplConvertCtx, "out_sample_rate",    pCodecCxt->sample_rate,    0);
    av_opt_set_sample_fmt(pSmplConvertCtx, "out_sample_fmt",     pCodecCxt->sample_fmt,     0);

    if ((swr_init(pSmplConvertCtx)) < 0)
    
        fprintf(stderr, "Failed to initialize the resampling context\n");
        exit(1);
    

    nSizeAudioEncodeBuffer = av_samples_get_buffer_size(NULL, pCodecCxt->channels, pCodecCxt->sample_rate, pCodecCxt->sample_fmt, 1);
    if (pAudioEncodeBuffer == NULL)
    
        pAudioEncodeBuffer = (uint8_t * )av_malloc(nSizeAudioEncodeBuffer);
    

【问题讨论】：

不确定，这可能指向正确的方向：***.com/questions/45241185/… 我对 libav 不是很熟悉。

【参考方案1】：

这里是过程函数：

process_data(AVFrame *frame, uint8_t* data, uint32_t sample_count, uint64_t device_ts)

    int sample_size = _bit_per_sample / 8 * _channel_num;

    //wasapi time unit is 100ns,so time base is NS_PER_SEC
    frame->pts = _use_device_ts ? device_ts * 100 : av_gettime_relative();

    if(_use_device_ts == false)
        frame->pts -= (int64_t)sample_count * NS_PER_SEC / (int64_t)_sample_rate;

    frame->pkt_dts = frame->pts;
    frame->nb_samples = sample_count;
    frame->format = _fmt;
    frame->sample_rate = _sample_rate;
    frame->channels = _channel_num;
    frame->pkt_size = sample_count*sample_size;

    av_samples_fill_arrays(frame->data, frame->linesize, data, _channel_num, sample_count, _fmt, 1);

    if (_on_data) _on_data(frame, _cb_extra_index);

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【讨论】：

你的输出文件听起来像什么？

我正在尝试创建一个 AVI 文件。它有两个音频和视频流

视频写入正确。我有音频问题

输出文件没有声音。

您必须确保视频和音频流的pkt设置正确