2016-09-22 105 views
0

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,提取所有内容并用RtAudio播放样本。RtAudio - 从wav文件播放示例

我做了一个WaveLoader类,让我提取样本和元数据。我使用this指南来做到这一点,并且我使用010编辑器检查了一切正确。这是010编辑器的快照,显示结构和数据。

010 Editor

这也是我如何存储WaveLoader类中的原始样本:

 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获得:1,-3,4,-5 ...这似乎确定。

问题是,我不知道如何玩他们。这是我做了什么:

/* 
* 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; 
} 

内的回调函数,当我打印出来的样本值,我得到酷似010的编辑器?为什么没有rtaudio播放它们。这里有什么问题?我是否需要将样本值归一化到-1和1之间?

编辑: 我试图播放的wav文件:

  • CHUNKSIZE:16
  • 格式:1
  • 通道:1
  • 采样率:48000
  • ByteRate:96000
  • BlockAlign:2
  • BitPerSample:16
  • 原始样本的总规模:2217044个字节

回答

0

出于某种原因,当我通过输入参数的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; 

它是那么的随意,当我试图播放它的工作原理我话筒。我离开输入参数,我的wav文件突然播放。这是一个错误吗?