我无法弄清楚如何创建自己的声音播放器，因此我选择使用ChiliTomatoNoodle框架中的一个。
然而，我遇到的问题是我有一个180s的波形文件，它只播放第一秒左右。我该怎么办才能让它玩得更久？
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 ) );
}
提前感谢您的帮助！
答案 0 :(得分：0)
假设您有一个.wav文件，并且您正在将声音文件加载到以下位置：
yourSound = audio.CreateSound("fileName.WAV"); //Capslock on WAV
yourSound.Play();
这样就会在标题中声明Sound：
Sound yourSound;
现在因为你可能已经完成了这个并且这不起作用，它可能与您的文件有关，因为播放声音160秒+应该不是问题。
您是否正在使用.WAV文件进行声音播放？如果是这样，你碰巧转换它（因为它可能是背景声音？）。如果您尝试使用此转换器进行转换：
如果有效，请告诉我！
答案 1 :(得分：0)
你的缓冲区可能只有大到足以播放第一秒左右。您需要做的是设置“通知”。请参阅documentation。
通知是一种要求音频硬件在到达缓冲区中的特定点时通知您的方法。
想法是在缓冲区中间和缓冲区末尾设置通知。当您从中间的通知收到通知时，您将使用更多数据填充缓冲区的前半部分。当您从最后收到通知时，您将使用更多数据填充缓冲区的后半部分。这样，您就可以使用单个缓冲区传输无限量的数据。