Как я могу заставить этот код воспроизводить волновой файл дольше?

Я не мог понять, как создать свой собственный звуковой плеер, поэтому я решил использовать его из фреймворка ChiliTomatoNoodle.

Однако у меня проблема в том, что у меня есть файл волны 180-х годов, который воспроизводит только первую секунду или около того. Что мне нужно сделать, чтобы заставить его играть дольше?

sound.h:

#pragma once

#include <windows.h>
#include <mmsystem.h>
#include <dsound.h>
#include <stdio.h>

class DSound;

class Sound
{
friend DSound;
public:
Sound( const Sound& base );
Sound();
~Sound();
const Sound& operator=( const Sound& rhs );
void Play( int attenuation = DSBVOLUME_MAX );
private:
Sound( IDirectSoundBuffer8* pSecondaryBuffer );
private:
IDirectSoundBuffer8* pBuffer;
};

class DSound
{
private:
struct WaveHeaderType
{
char chunkId[4];
unsigned long chunkSize;
char format[4];
char subChunkId[4];
unsigned long subChunkSize;
unsigned short audioFormat;
unsigned short numChannels;
unsigned long sampleRate;
unsigned long bytesPerSecond;
unsigned short blockAlign;
unsigned short bitsPerSample;
char dataChunkId[4];
unsigned long dataSize;
};
public:
DSound( HWND hWnd );
~DSound();
Sound CreateSound( char* wavFileName );
private:
DSound();
private:
IDirectSound8* pDirectSound;
IDirectSoundBuffer* pPrimaryBuffer;
};

Sound.cpp:

#include "Sound.h"#include <assert.h>

#pragma comment(lib, "dsound.lib")
#pragma comment(lib, "dxguid.lib")
#pragma comment(lib, "winmm.lib" )

DSound::DSound( HWND hWnd )
: pDirectSound( NULL ),
pPrimaryBuffer( NULL )
{
HRESULT result;
DSBUFFERDESC bufferDesc;
WAVEFORMATEX waveFormat;

result = DirectSoundCreate8( NULL,&pDirectSound,NULL );
assert( !FAILED( result ) );

// Set the cooperative level to priority so the format of the primary sound buffer can be modified.
result = pDirectSound->SetCooperativeLevel( hWnd,DSSCL_PRIORITY );
assert( !FAILED( result ) );

// Setup the primary buffer description.
bufferDesc.dwSize = sizeof(DSBUFFERDESC);
bufferDesc.dwFlags = DSBCAPS_PRIMARYBUFFER | DSBCAPS_CTRLVOLUME;
bufferDesc.dwBufferBytes = 0;
bufferDesc.dwReserved = 0;
bufferDesc.lpwfxFormat = NULL;
bufferDesc.guid3DAlgorithm = GUID_NULL;

// Get control of the primary sound buffer on the default sound device.
result = pDirectSound->CreateSoundBuffer( &bufferDesc,&pPrimaryBuffer,NULL );
assert( !FAILED( result ) );

// Setup the format of the primary sound bufffer.
// In this case it is a .WAV file recorded at 44,100 samples per second in 16-bit stereo (cd audio format).
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = 44100;
waveFormat.wBitsPerSample = 16;
waveFormat.nChannels = 2;
waveFormat.nBlockAlign = (waveFormat.wBitsPerSample / 8) * waveFormat.nChannels;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;

// Set the primary buffer to be the wave format specified.
result = pPrimaryBuffer->SetFormat( &waveFormat );
assert( !FAILED( result ) );
}

DSound::~DSound()
{
if( pPrimaryBuffer )
{
pPrimaryBuffer->Release();
pPrimaryBuffer = NULL;
}
if( pDirectSound )
{
pDirectSound->Release();
pDirectSound = NULL;
}
}

// must be 44.1k 16bit Stereo PCM Wave
Sound DSound::CreateSound( char* wavFileName )
{
int error;
FILE* filePtr;
unsigned int count;
WaveHeaderType waveFileHeader;
WAVEFORMATEX waveFormat;
DSBUFFERDESC bufferDesc;
HRESULT result;
IDirectSoundBuffer* tempBuffer;
IDirectSoundBuffer8* pSecondaryBuffer;
unsigned char* waveData;
unsigned char* bufferPtr;
unsigned long bufferSize;// Open the wave file in binary.
error = fopen_s( &filePtr,wavFileName,"rb" );
assert( error == 0 );

// Read in the wave file header.
count = fread( &waveFileHeader,sizeof( waveFileHeader ),1,filePtr );
assert( count == 1 );

// Check that the chunk ID is the RIFF format.
assert( (waveFileHeader.chunkId[0] == 'R') &&
(waveFileHeader.chunkId[1] == 'I') &&
(waveFileHeader.chunkId[2] == 'F') &&
(waveFileHeader.chunkId[3] == 'F') );

// Check that the file format is the WAVE format.
assert( (waveFileHeader.format[0] == 'W') &&
(waveFileHeader.format[1] == 'A') &&
(waveFileHeader.format[2] == 'V') &&
(waveFileHeader.format[3] == 'E') );

// Check that the sub chunk ID is the fmt format.
assert( (waveFileHeader.subChunkId[0] == 'f') &&
(waveFileHeader.subChunkId[1] == 'm') &&
(waveFileHeader.subChunkId[2] == 't') &&
(waveFileHeader.subChunkId[3] == ' ') );

// Check that the audio format is WAVE_FORMAT_PCM.
assert( waveFileHeader.audioFormat == WAVE_FORMAT_PCM );

// Check that the wave file was recorded in stereo format.
assert( waveFileHeader.numChannels == 2 );

// Check that the wave file was recorded at a sample rate of 44.1 KHz.
assert( waveFileHeader.sampleRate == 44100 );

// Ensure that the wave file was recorded in 16 bit format.
assert( waveFileHeader.bitsPerSample == 16 );

// Check for the data chunk header.
assert( (waveFileHeader.dataChunkId[0] == 'd') &&
(waveFileHeader.dataChunkId[1] == 'a') &&
(waveFileHeader.dataChunkId[2] == 't') &&
(waveFileHeader.dataChunkId[3] == 'a') );

// Set the wave format of secondary buffer that this wave file will be loaded onto.
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
waveFormat.nSamplesPerSec = 44100;
waveFormat.wBitsPerSample = 16;
waveFormat.nChannels = 2;
waveFormat.nBlockAlign = (waveFormat.wBitsPerSample / 8) * waveFormat.nChannels;
waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
waveFormat.cbSize = 0;

// Set the buffer description of the secondary sound buffer that the wave file will be loaded onto.
bufferDesc.dwSize = sizeof(DSBUFFERDESC);
bufferDesc.dwFlags = DSBCAPS_CTRLVOLUME;
bufferDesc.dwBufferBytes = waveFileHeader.dataSize;
bufferDesc.dwReserved = 0;
bufferDesc.lpwfxFormat = &waveFormat;
bufferDesc.guid3DAlgorithm = GUID_NULL;

// Create a temporary sound buffer with the specific buffer settings.
result = pDirectSound->CreateSoundBuffer( &bufferDesc,&tempBuffer,NULL );
assert( !FAILED( result ) );

// Test the buffer format against the direct sound 8 interface and create the secondary buffer.
result = tempBuffer->QueryInterface( IID_IDirectSoundBuffer8,(void**)&pSecondaryBuffer );
assert( !FAILED( result ) );

// Release the temporary buffer.
tempBuffer->Release();
tempBuffer = 0;

// Move to the beginning of the wave data which starts at the end of the data chunk header.
fseek( filePtr,sizeof(WaveHeaderType),SEEK_SET );

// Create a temporary buffer to hold the wave file data.
waveData = new unsigned char[ waveFileHeader.dataSize ];
assert( waveData );

// Read in the wave file data into the newly created buffer.
count = fread( waveData,1,waveFileHeader.dataSize,filePtr );
assert( count == waveFileHeader.dataSize);

// Close the file once done reading.
error = fclose( filePtr );
assert( error == 0 );

// Lock the secondary buffer to write wave data into it.
result = pSecondaryBuffer->Lock( 0,waveFileHeader.dataSize,(void**)&bufferPtr,(DWORD*)&bufferSize,NULL,0,0 );
assert( !FAILED( result ) );

// Copy the wave data into the buffer.
memcpy( bufferPtr,waveData,waveFileHeader.dataSize );

// Unlock the secondary buffer after the data has been written to it.
result = pSecondaryBuffer->Unlock( (void*)bufferPtr,bufferSize,NULL,0 );
assert( !FAILED( result ) );

// Release the wave data since it was copied into the secondary buffer.
delete [] waveData;
waveData = NULL;

return Sound( pSecondaryBuffer );
}

Sound::Sound( IDirectSoundBuffer8* pSecondaryBuffer )
: pBuffer( pSecondaryBuffer )
{}

Sound::Sound()
: pBuffer( NULL )
{}

Sound::Sound( const Sound& base )
: pBuffer( base.pBuffer )
{
pBuffer->AddRef();
}

Sound::~Sound()
{
if( pBuffer )
{
pBuffer->Release();
pBuffer = NULL;
}
}

const Sound& Sound::operator=( const Sound& rhs )
{
this->~Sound();
pBuffer = rhs.pBuffer;
pBuffer->AddRef();
return rhs;
}

// attn is the attenuation value in units of 0.01 dB (larger
// negative numbers give a quieter sound, 0 for full volume)
void Sound::Play( int attn )
{
attn = max( attn,DSBVOLUME_MIN );
HRESULT result;

// check that we have a valid buffer
assert( pBuffer != NULL );

// Set position at the beginning of the sound buffer.
result = pBuffer->SetCurrentPosition( 0 );
assert( !FAILED( result ) );

// Set volume of the buffer to attn
result = pBuffer->SetVolume( attn );
assert( !FAILED( result ) );

// Play the contents of the secondary sound buffer.
result = pBuffer->Play( 0,0,0 );
assert( !FAILED( result ) );
}

Заранее спасибо за помощь!