如何让这段代码更长时间播放wav文件？

Question

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！

Answer 1

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：

Converter MP3 -> WAV

如果有效，请告诉我！

Answer 2

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。