我编写了两个函数,它们应该将音频浮点缓冲区导出到.wav文件中,但是我在播放导出的文件时遇到了问题。 Audacity像它应该的那样播放它(听起来和我的应用程序完全一样),然而,Ableton(DAW软件)似乎误解了wav的某些部分,所以它听起来真的是扭曲的。 (就像失真效应一样)
我认为,能够以某种方式假设样本深度(较小),因此实际样本会超出极限。
我有两个函数,一个从两个浮点缓冲区创建一个int32_t
缓冲区(左右混合到一个缓冲区中),另一个函数写入.wav文件,包括格式块等.I猜猜哪个地方有问题。
班级成员/结构
// static I use in the export function
static const int FORMAT_PCM = 1;
static const int CHANNEL_COUNT = 2; // fix stereo
static const int BYTES_PER_SAMPLE = 4; // fix bytes per sample, 32bit audio
// a function I found in the internet, helps writting the bytes to the file
template <typename T>
static void write(std::ofstream& stream, const T& t) {
stream.write((const char*)&t, sizeof(T));
};
// used "structure" to store the buffer
class StereoAudioBuffer {
public:
StereoAudioBuffer(int length) : sizeInSamples(2*length){
samples = new int32_t[2*length];
};
~StereoAudioBuffer() {delete samples;};
int32_t *samples;
const int sizeInSamples;
};
转换功能
StereoAudioBuffer* WaveExport::convertTo32BitStereo(
float *leftSamples,
float*rightSamples,
int length)
{
StereoAudioBuffer *buffer = new StereoAudioBuffer(length);
float max = 0;
// find max sample
for(int i = 0; i < length; i++) {
if(abs(leftSamples[i]) > max) {
max = abs(leftSamples[i]);
}
if(abs(rightSamples[i]) > max) {
max = abs(rightSamples[i]);
}
}
// normalise and scale to size(int32_t)
float factor = 2147483000.0f / max;
for(int i = 0; i < length; i++) {
buffer->samples[2*i] = leftSamples[i] * factor ;
buffer->samples[2*i+1] = rightSamples[i] * factor;
}
return buffer;
}
导出功能(此代码的一部分来自互联网,遗憾的是,我再也找不到来源了
void WaveExport::writeStereoWave(
const char *path,
StereoAudioBuffer* buffer,
int sampleRate)
{
std::ofstream stream(path, std::ios::binary);
// RIFF
stream.write("RIFF", 4);
// FILE SIZE
write<int>(stream, 36 + buffer->sizeInSamples * BYTES_PER_SAMPLE); // 32 bits -> 4 bytes
// WAVE
stream.write("WAVE", 4);
// FORMAT CHUNK
stream.write("fmt ", 4);
write<int>(stream, 16);
write<short>(stream, FORMAT_PCM); // Format
write<short>(stream, CHANNEL_COUNT); // Channels
write<int>(stream, sampleRate); // Sample Rate
write<int>(stream, sampleRate * CHANNEL_COUNT * BYTES_PER_SAMPLE); // Byterate
write<short>(stream, CHANNEL_COUNT * BYTES_PER_SAMPLE); // Frame size
write<short>(stream, 8 * BYTES_PER_SAMPLE); // Bits per sample
int dataChunkSize = buffer->sizeInSamples * BYTES_PER_SAMPLE;
// SAMPLES
stream.write("data", 4);
stream.write((const char*)&dataChunkSize, 4);
stream.write((const char*)buffer->samples, BYTES_PER_SAMPLE*buffer->sizeInSamples);
}
有人知道怎么写.wav文件,也许可以告诉我我做错了什么或错过了什么? 谢谢!
答案 0 :(得分:0)
没有问题。我使用的是32位.wav,在应用程序中不支持,我用于播放。
我将导出函数更改为使用int16_t,16位深度,并且工作正常。