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我无法弄清楚如何创建自己的声音播放器，所以我选择使用ChiliTomatoNoodle的框架。如何让此代码播放更长的波形文件？
但是，我遇到的问题是我有一个180s的波形文件，这只是播放第一秒左右。我需要做些什么才能延长比赛时间？
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));
}
感谢您的帮助提前！
你的意思是在循环播放声音？然后在“Play”调用中设置“循环”标志。 –