Question

我目前正在尝试学习音频编程。我的目标是打开一个wav文件，提取所有内容并使用RtAudio播放样本。

我制作了一个WaveLoader类，让我提取样本和元数据。我使用this指南来做这件事，我用010编辑器检查了一切是否正确。这是010编辑器的快照，显示了结构和数据。

这就是我将原始样本存储在WaveLoader类中的方法：

        data = new short[wave_data.payloadSize]; // - Allocates memory size of chunk size

        if (!fread(data, 1, wave_data.payloadSize, sound_file))
        {
            throw ("Could not read wav data");
        }

如果我打印出每个样品，我会得到：1，-3,4，-5 ......这似乎没问题。

问题在于我不确定如何玩它们。这就是我所做的：

/*
 * Using PortAudio to play samples
 */
bool Player::Play() 
{
    ShowDevices();
    rt.showWarnings(true);

    RtAudio::StreamParameters oParameters; //, iParameters;
    oParameters.deviceId = rt.getDefaultOutputDevice();
    oParameters.firstChannel = 0;
    oParameters.nChannels = mAudio.channels;

    //iParameters.deviceId = rt.getDefaultInputDevice();
    //iParameters.nChannels = 2;

    unsigned int sampleRate = mAudio.sampleRate;

    // Use a buffer of 512, we need to feed callback with 512 bytes everytime!
    unsigned int nBufferFrames = 512;

    RtAudio::StreamOptions options;
    options.flags = RTAUDIO_SCHEDULE_REALTIME;
    options.flags = RTAUDIO_NONINTERLEAVED;

    //&parameters, NULL, RTAUDIO_FLOAT64,sampleRate, &bufferFrames, &mCallback, (void *)&rawData

    try {
        rt.openStream(&oParameters, NULL, RTAUDIO_SINT16, sampleRate, &nBufferFrames, &mCallback, (void*) &mAudio);
        rt.startStream();
    }
    catch (RtAudioError& e) {
        std::cout << e.getMessage() << std::endl;
        return false;
    }
    return true;
}

/*
* RtAudio Callback
*
*/
int mCallback(void * outputBuffer, void * inputBuffer, unsigned int nBufferFrames, double streamTime, RtAudioStreamStatus status, void * userData)
{
    unsigned int i = 0;
    short *out = static_cast<short*>(outputBuffer);
    auto *data = static_cast<Player::AUDIO_DATA*>(userData);

    // if i is more than our data size, we are done!
    if (i > data->dataSize) return 1;

    // First time callback is called data->ptr is 0, this means that the offset is 0
    // Second time data->ptr is 1, this means offset = nBufferFrames (512) * 1 = 512
    unsigned int offset = nBufferFrames * data->ptr++;

    printf("Offset: %i\n", offset);
    // First time callback is called offset is 0, we are starting from 0 and looping nBufferFrames (512) times, this gives us 512 bytes
    // Second time, the offset is 1, we are starting from 512 bytes and looping to 512 + 512 = 1024 
    for (i = offset; i < offset + nBufferFrames; ++i)
    {
        short sample = data->rawData[i]; // Get raw sample from our struct
        *out++ = sample;                // Pass to output buffer for playback

        printf("Current sample value: %i\n", sample);       // this is showing 1, -3, 4, -5 check 010 editor
    }

    printf("Current time: %f\n", streamTime);
    return 0;
}

内部回调函数，当我打印出样本值时，我得到的结果与010编辑器完全相同？为什么rtaudio不会播放它们。这有什么不对？我是否需要将样本值标准化为介于-1和1之间？

编辑：我试图播放的wav文件：

Chunksize：16
格式：1
频道：1
SampleRate：48000
ByteRate：96000
BlockAlign：2
BitPerSample：16
原始样本总数：2217044字节

Answer 1

由于某种原因，当我将输入参数传递给openStream（）

时，它可以正常工作

    RtAudio::StreamParameters oParameters, iParameters;
    oParameters.deviceId = rt.getDefaultOutputDevice();
    oParameters.firstChannel = 0;
    //oParameters.nChannels = mAudio.channels;
    oParameters.nChannels = mAudio.channels;

    iParameters.deviceId = rt.getDefaultInputDevice();
    iParameters.nChannels = 1;

    unsigned int sampleRate = mAudio.sampleRate;

    // Use a buffer of 512, we need to feed callback with 512 bytes everytime!
    unsigned int nBufferFrames = 512;

    RtAudio::StreamOptions options;
    options.flags = RTAUDIO_SCHEDULE_REALTIME;
    options.flags = RTAUDIO_NONINTERLEAVED;

    //&parameters, NULL, RTAUDIO_FLOAT64,sampleRate, &bufferFrames, &mCallback, (void *)&rawData

    try {
        rt.openStream(&oParameters, &iParameters, RTAUDIO_SINT16, sampleRate, &nBufferFrames, &mCallback, (void*) &mAudio);
        rt.startStream();
    }
    catch (RtAudioError& e) {
        std::cout << e.getMessage() << std::endl;
        return false;
    }
    return true;

当我试图播放麦克风时，它是如此随机。我留下输入参数，我的wav文件突然播放。这是一个错误吗？

RtAudio - 从wav文件播放样本

1 个答案: