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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。

我创建了一个 WaveLoader 类，让我提取样本和元数据。我使用本指南来执行此操作，并使用 010 编辑器检查了一切是否正确。这是显示结构和数据的 010 编辑器的快照。

010 编辑器

这就是我在 WaveLoader 类中存储原始样本的方式：

        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本，我会得到：1、-3、4、-5 ... 这似乎还可以。

问题是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;
}

在回调函数内部，当我打印出样本值时，我得到的结果与 010 编辑器一模一样吗？为什么 rtaudio 不播放它们。这里有什么问题？我需要将样本值标准化到 -1 和 1 之间吗？

编辑：我要播放的 wav 文件：

  • 块大小：16
  • 格式：1
  • 频道：1
  • 采样率：48000
  • 字节率：96000
  • 块对齐：2
  • BitPerSample：16
  • 原始样本总大小：2217044 字节
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1 回答 1

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出于某种原因，当我将输入参数传递给 openStream()

    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;

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

于 2016-09-22T19:07:50.360 回答