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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。
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;
//¶meters, 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