如何使用C++生成音频波形

How to generate a waveform of audio wav file using C++

本文关键字:音频 波形 何使用 C++      更新时间:2023-10-16

我目前正在使用C++来实现我的代码。这个想法是我想通过程序读取 wav 文件,然后在屏幕上输出其波形。我已经在那里找到了一些参考资料:C++读取WAV文件的数据部分。这是我读取音频文件的代码。我不知道如何生成波形.

#include <iostream>
#include <string>
#include <fstream>
using namespace std;
using std::string;
using std::fstream;

typedef struct  WAV_HEADER
{
/* for the part of RIEF Chunk Descriptor */
uint8_t             RIFF[4];        // RIFF Header      Magic header
uint32_t            ChunkSize;      // RIFF Chunk Size  
uint8_t             WAVE[4];        // WAVE Header  
/* for the part of "fmt" subChunk */    
uint8_t             fmt[4];         // FMT header       
uint32_t            Subchunk1Size;  // Size of the fmt chunk                                
uint16_t            AudioFormat;    // Audio format 1=PCM,6=mulaw,7=alaw, 257=IBM Mu-Law, 258=IBM A-Law, 259=ADPCM 
uint16_t            NumOfChan;      // Number of channels 1=Mono 2=Sterio                   
uint32_t             SamplesPerSec;  // Sampling Frequency in Hz                             
uint32_t             bytesPerSec;    // bytes per second 
uint16_t             blockAlign;     // 2=16-bit mono, 4=16-bit stereo 
uint16_t             bitsPerSample;  // Number of bits per sample
/* For the part of "data" subChunk */      
uint8_t              Subchunk2ID[4]; // "data"  string   
uint32_t             Subchunk2Size;  // Sampled data length    
}wav_hdr; 

int getFileSize(FILE* inFile);
int main(int argc, char* argv[])
{
wav_hdr wavHeader;
int headerSize = sizeof(wav_hdr), filelength = 0;
const char* filePath;
string input;
if (argc <= 1)
{
cout << "Input wave file name: ";
cin >> input;
cin.get();
filePath = input.c_str();
}
else
{
filePath = argv[1];
cout << "Input wave file name: " << filePath << endl;
}
FILE* wavFile = fopen(filePath, "r");
if (wavFile == nullptr)
{
fprintf(stderr, "Unable to open wave file: %sn", filePath);
return 1;
}
//Read the header
size_t bytesRead = fread(&wavHeader, 1, headerSize, wavFile);
cout << "Header Read " << bytesRead << " bytes." << endl;
if (bytesRead > 0)
{
//Read the data
uint16_t bytesPerSample = wavHeader.bitsPerSample / 8;      //Number     of bytes per sample
uint64_t numSamples = wavHeader.ChunkSize / bytesPerSample; //How many samples are in the wav file?
static const uint16_t BUFFER_SIZE = 4096;
int8_t* buffer = new int8_t[BUFFER_SIZE];
while ((bytesRead = fread(buffer, sizeof buffer[0], BUFFER_SIZE / (sizeof buffer[0]), wavFile)) > 0)
{
/** DO SOMETHING WITH THE WAVE DATA HERE **/
cout << "Read " << bytesRead << " bytes." << endl;
}
delete [] buffer;
buffer = nullptr;
filelength = getFileSize(wavFile);
cout << "File is :" << filelength << " bytes." << endl;
cout << "RIFF header :" << wavHeader.RIFF[0] << wavHeader.RIFF[1] << wavHeader.RIFF[2] << wavHeader.RIFF[3] << endl;
cout << "WAVE header :" << wavHeader.WAVE[0] << wavHeader.WAVE[1] << wavHeader.WAVE[2] << wavHeader.WAVE[3] << endl;
cout << "FMT :" << wavHeader.fmt[0] << wavHeader.fmt[1] << wavHeader.fmt[2] << wavHeader.fmt[3] << endl;
cout << "Data size :" << wavHeader.ChunkSize << endl;
// Display the sampling Rate from the header
cout << "Sampling Rate :" << wavHeader.SamplesPerSec << endl;
cout << "Number of bits used :" << wavHeader.bitsPerSample << endl;
cout << "Number of channels :" << wavHeader.NumOfChan << endl;
cout << "Number of bytes per second :" << wavHeader.bytesPerSec << endl;
cout << "Data length :" << wavHeader.Subchunk2Size << endl;
cout << "Audio Format :" << wavHeader.AudioFormat << endl;
// Audio format 1=PCM,6=mulaw,7=alaw, 257=IBM Mu-Law, 258=IBM A-Law, 259=ADPCM
cout << "Block align :" << wavHeader.blockAlign << endl;
cout << "Data string :" << wavHeader.Subchunk2ID[0] << wavHeader.Subchunk2ID[1] << wavHeader.Subchunk2ID[2] << wavHeader.Subchunk2ID[3] << endl;

}
fclose(wavFile);
return 0;
}
// find the file size
int getFileSize(FILE* inFile)
{
int fileSize = 0;
fseek(inFile, 0, SEEK_END);
fileSize = ftell(inFile);
fseek(inFile, 0, SEEK_SET);
return fileSize;
}

我会把它作为带有代码要点的评论而不是答案,因为它没有直接回答你的问题,但我没有足够的声誉。

我做了你想用Qt做的事情,我的大部分代码都来自Qt的文档。您需要找到样本的峰值,然后绘制它。让我分享一些代码,这些代码可以让您了解您可能想要如何做到这一点。

void Waveform::appendSamples()
{
buffer = audioDecoder->read();
qreal peak = getPeakValue(buffer.format());
const qint16 *data = buffer.constData<qint16>();
int count = buffer.sampleCount() / 2;
for (int i = 0; i < count; i += 1200){ // I want 40 samples per second currently assuming 48kHz
double val = data[i]/peak;         
samples.append(val * 300); // *300 for scaling
}
}
qreal Waveform::getPeakValue(const QAudioFormat &format)
{
qreal ret(0);
if (format.isValid()){
switch (format.sampleType()) {
case QAudioFormat::Unknown:
break;
case QAudioFormat::Float:
if (format.sampleSize() != 32)
ret = 0;
else
ret = 1.00003;
break;
case QAudioFormat::SignedInt:
if (format.sampleSize() == 32)
ret = INT_MAX;
else if (format.sampleSize() == 16)
ret = SHRT_MAX;
else if (format.sampleSize() == 8)
ret = CHAR_MAX;
break;
case QAudioFormat::UnSignedInt:
if (format.sampleSize() == 32)
ret = UINT_MAX;
else if (format.sampleSize() == 16)
ret = USHRT_MAX;
else if (format.sampleSize() == 8)
ret = UCHAR_MAX;
break;
default:
break;
}
}
return ret;
}

对于Qt中的绘图部分(可能对您没有用(

QSGNode* WaveformRenderer::updatePaintNode(QSGNode *oldNode, UpdatePaintNodeData *)
{
QSGSimpleRectNode *n = static_cast<QSGSimpleRectNode *>(oldNode);
if(!n){
n = new QSGSimpleRectNode();
n->setColor(Qt::red);
}
QVector<double> samples = wav->getSamples(); //retrieve the samples
int numberOfSamples = samples.size();
setItemWidth(qreal(numberOfSamples * 2)); //signal stuff for qml
for(int i = 0; i < numberOfSamples; ++i){
QSGSimpleRectNode *temp = new QSGSimpleRectNode();
temp->setColor(Qt::green);
temp->setRect(i * 2, height()/2 - samples[i], 2, samples[i] * 2);
n->appendChildNode(temp);
}
return n;
}