J'ai l'intention de créer un programme qui visualisera la forme d'onde audio d'un fichier .wav.
Jusqu'à présent, j'ai commencé par lire correctement la partie d'en-tête dudit fichier wav. Le code que j'utilise serait le suivant:
#include <iostream>
#include <string>
#include <fstream>
using namespace std;
using std::string;
using std::fstream;
typedef struct WAV_HEADER{
char RIFF[4]; // RIFF Header Magic header
unsigned long ChunkSize; // RIFF Chunk Size
char WAVE[4]; // WAVE Header
char fmt[4]; // FMT header
unsigned long Subchunk1Size; // Size of the fmt chunk
unsigned short AudioFormat; // Audio format 1=PCM,6=mulaw,7=alaw, 257=IBM Mu-Law, 258=IBM A-Law, 259=ADPCM
unsigned short NumOfChan; // Number of channels 1=Mono 2=Sterio
unsigned long SamplesPerSec; // Sampling Frequency in Hz
unsigned long bytesPerSec; // bytes per second
unsigned short blockAlign; // 2=16-bit mono, 4=16-bit stereo
unsigned short bitsPerSample; // Number of bits per sample
char Subchunk2ID[4]; // "data" string
unsigned long Subchunk2Size; // Sampled data length
}wav_hdr;
// Function prototypes
int getFileSize(FILE *inFile);
int main(int argc,char *argv[]){
wav_hdr wavHeader;
FILE *wavFile;
int headerSize = sizeof(wav_hdr),filelength = 0;
string answer;
do{
string input;
string answer;
const char* filePath;
cout << "Pick wav file from the Windows Media File: ";
cin >> input;
cin.get();
cout << endl;
path = "C:\\Windows\\Media\\" + input + ".wav";
filePath = path.c_str();
wavFile = fopen( filePath , "r" );
if(wavFile == NULL){
printf("Can not able to open wave file\n");
//exit(EXIT_FAILURE);
}
fread(&wavHeader,headerSize,1,wavFile);
filelength = getFileSize(wavFile);
fclose(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 form 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;
cout << endl << endl << "Try something else? (y/n)";
cin >> answer;
//cin.get();
cout << endl << endl;
}while( answer == "y" );
getchar();
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;
}
Je l'ai essayé plusieurs fois et les données qu'il donne semblent cohérentes via différents fichiers wav dans le dossier Media du dossier Windows.
L'étape suivante serait alors de stocker les données réelles du fichier wav dans un vecteur. Cependant, je ne sais absolument pas comment procéder. Les solutions en ligne que j'ai trouvées sont allées jusqu'à la lecture du fichier d'en-tête.
Des idées sur la façon de stocker (et, espérons-le, d'afficher) les données réelles du fichier wav? Merci!
Cette image est tirée de n cours de Stanford
Vous pouvez donc voir que les données audio se produisent immédiatement après les en-têtes que vous avez déjà lus et il y aura Subchunk2Size
octets de données audio.
Le pseudocode pour cela serait
ReadRIFF();
ReadFMT();
int32 chunk2Id = Read32(BigEndian);
int32 chunk2Size = Read32(LittleEndian);
for (int i = 0; i < chunk2Size; i++)
{
audioData[i] = ReadByte();
}
Si l'audio est stéréo, vous aurez deux flux audio dans data
. Si l'audio est compressé (mp3, aac, etc.), vous devrez d'abord le décompresser.
Je sais que c'est un vieux post, mais vos paramètres de fread sont commutés, voici une version plus correcte (nécessite g ++ - 4.7 ou supérieur avec le drapeau -std = c ++ 11 comme ceci "g ++ -std = c ++ 11 WaveReader. cpp -o WaveReader ").
#include <iostream>
#include <string>
#include <fstream>
#include <cstdint>
using std::cin;
using std::cout;
using std::endl;
using std::fstream;
using std::string;
typedef struct WAV_HEADER
{
/* RIFF Chunk Descriptor */
uint8_t RIFF[4]; // RIFF Header Magic header
uint32_t ChunkSize; // RIFF Chunk Size
uint8_t WAVE[4]; // WAVE Header
/* "fmt" sub-chunk */
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
/* "data" sub-chunk */
uint8_t Subchunk2ID[4]; // "data" string
uint32_t Subchunk2Size; // Sampled data length
} wav_hdr;
// Function prototypes
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: %s\n", 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;
}