第1年8月9日 ffmpeg音频 fltp

1.

由于最新ffmpeg 库的sample_fmt必须以AV_SAMPLE_FMT_FLTP这种方式进行存储，而ENCODING_PCM_16BIT 是AV_SAMPLE_FMT_S16格式的。如果是单声道的话两者区别不大，都可以存在AVFrame->data[0] 里面，只是AV_SAMPLE_FMT_FLTP是以浮点数的方式存储，后者是以有符号整形16位存贮的。如果是双声道的话二者的区别就比较大了。

AV_SAMPLE_FMT_S16是非平坦的，左右声道以LRLR....的方式连续的存在一个数组里面，AV_SAMPLE_FMT_FLTP是平坦的存贮方式，左右声道是分开存在两个数组里面的。

如何将Android 以 CHANNEL_IN_STEREO(双声道) ENCODING_PCM_16BIT (AV_SAMPLE_FMT_S16) 录制方式用ffmpeg 用AV_SAMPLE_FMT_FLTP编码为aac音频文件呢?其实ffmepg 已经提供了这样的接口，就是swr_convert 函数，转换方式如下:

 https://www.bilibili.com/video/BV1jz4y1Z7pf?p=1

https://blog.csdn.net/XIAIBIANCHENG/article/details/72810495

https://www.jianshu.com/p/be78830eefbd

2.音频渲染

- (AudioFrame *) handleAudioFrame
{
    if (!_audioFrame->data[0])
        return nil;

    id<AudioManager> audioManager = [AudioManager audioManager];

    const NSUInteger numChannels = audioManager.numOutputChannels;
    NSInteger numFrames;

    void * audioData;

    if (_swrContext) {

        const NSUInteger ratio = MAX(1, audioManager.samplingRate / _audioCodecCtx->sample_rate) *
                                 MAX(1, audioManager.numOutputChannels / _audioCodecCtx->channels) * 2;

        const int bufSize = av_samples_get_buffer_size(NULL,
                                                       audioManager.numOutputChannels,
                                                       _audioFrame->nb_samples * ratio,
                                                       AV_SAMPLE_FMT_S16,
                                                       1);

        if (!_swrBuffer || _swrBufferSize < bufSize) {
            _swrBufferSize = bufSize;
            _swrBuffer = realloc(_swrBuffer, _swrBufferSize);
        }

        Byte *outbuf[2] = { _swrBuffer, 0 };

        numFrames = swr_convert(_swrContext,
                                outbuf,
                                _audioFrame->nb_samples * ratio,
                                (const uint8_t **)_audioFrame->data,
                                _audioFrame->nb_samples);

        if (numFrames < 0) {
            LoggerAudio(0, @"fail resample audio");
            return nil;
        }

        //int64_t delay = swr_get_delay(_swrContext, audioManager.samplingRate);
        //if (delay > 0)
        //    LoggerAudio(0, @"resample delay %lld", delay);

        audioData = _swrBuffer;

    } else {

        if (_audioCodecCtx->sample_fmt != AV_SAMPLE_FMT_S16) {
            NSAssert(false, @"bucheck, audio format is invalid");
            return nil;
        }

        audioData = _audioFrame->data[0];
        numFrames = _audioFrame->nb_samples;
    }

    const NSUInteger numElements = numFrames * numChannels;
    NSMutableData *data = [NSMutableData dataWithLength:numElements * sizeof(float)];

    float scale = 1.0 / (float)INT16_MAX ;
    vDSP_vflt16((SInt16 *)audioData, 1, data.mutableBytes, 1, numElements);
    vDSP_vsmul(data.mutableBytes, 1, &scale, data.mutableBytes, 1, numElements);

    AudioFrame *frame = [[AudioFrame alloc] init];
    frame.position = av_frame_get_best_effort_timestamp(_audioFrame) * _audioTimeBase;
    frame.duration = av_frame_get_pkt_duration(_audioFrame) * _audioTimeBase;
    frame.samples = data;

    if (frame.duration == 0) {
        // sometimes ffmpeg can't determine the duration of audio frame
        // especially of wma/wmv format
        // so in this case must compute duration
        frame.duration = frame.samples.length / (sizeof(float) * numChannels * audioManager.samplingRate);
    }

#if DEBUG
    LoggerAudio(2, @"AFD: %.4f %.4f | %.4f ",
                frame.position,
                frame.duration,
                frame.samples.length / (8.0 * 44100.0));
#endif

    return frame;
}

https://blog.csdn.net/xwl198937/article/details/52386331