本篇是用DirectX
Audio和DirectShow播放声音和音乐（2）的续篇。
调整声道平衡
所谓声道平衡就是调节左右声道的大小，如下图所示：
我们一般使用的喇叭或耳机都有左右两个声道，把自己想象成在左右声道两边移动的点，一般情况下在中间，这样听到的来自左右声道的音量是一样的。你可以向左移动，移动过程中左声道音量逐渐变大，右声道音量逐渐变小。当移动到左声道最左边的时候，左声道音量最大（10000），右声道没有声音（-10000）。
DirectSound定义了两个宏帮助把声道平衡调节到最左边和最右边，使用DSBPAN_LEFT将声道调整到最左边，使用DSBPAN_RIGHT
将声道调整到最右边。
通过调用IDirectSoundBuffer8::SetPan函数可以调节声道平衡。
The SetPan method sets the relative volume of the left and right channels.
HRESULT SetPan(
LONG lPan
);
Parameters
- lPan
- Relative volume between the left and right channels.
Return Values
If the method succeeds, the return value is DS_OK. If the method fails, the
return value may be one of the following error values:
|
Return code |
|
DSERR_CONTROLUNAVAIL |
|
DSERR_GENERIC |
|
DSERR_INVALIDPARAM |
|
DSERR_PRIOLEVELNEEDED |
Remarks
The returned value is measured in hundredths of a decibel (dB), in the range
of DSBPAN_LEFT to DSBPAN_RIGHT. These values are defined in Dsound.h as -10,000
and 10,000 respectively. The value DSBPAN_LEFT means the right channel is
attenuated by 100 dB and is effectively silent. The value DSBPAN_RIGHT means the
left channel is silent. The neutral value is DSBPAN_CENTER, defined as 0, which
means that both channels are at full volume. When one channel is attenuated, the
other remains at full volume.
The pan control acts cumulatively with the volume control.
改变播放速度
改变播放速度实际上改变的是声音的音调（pitch）。想象在游戏中通过改变播放速度将英雄的声音变成花栗鼠的声音。使用这种方法可以把一段男性的声音变成女性的声音，通过调用IDirectSoundBuffer8::SetFrequency来改变音调：
The SetFrequency method sets the frequency at which the audio samples are
played.
HRESULT SetFrequency(
DWORD dwFrequency
);
Parameters
- dwFrequency
- Frequency, in hertz (Hz), at which to play the audio samples. A value of
DSBFREQUENCY_ORIGINAL resets the frequency to the default value of the
buffer format.
Return Values
If the method succeeds, the return value is DS_OK. If the method fails, the
return value may be one of the following error values:
|
Return code |
|
DSERR_CONTROLUNAVAIL |
|
DSERR_GENERIC |
|
DSERR_INVALIDPARAM |
|
DSERR_PRIOLEVELNEEDED |
Remarks
Increasing or decreasing the frequency changes the perceived pitch of the
audio data. This method does not affect the format of the buffer.
Before setting the frequency, you should ascertain whether the frequency is
supported by checking the dwMinSecondarySampleRate and dwMaxSecondarySampleRate
members of the DSCAPS structure for the device. Some operating systems do not
support frequencies greater than 100,000 Hz.
This method is not valid for the primary buffer.
如下图所示，它显示了音频缓冲以双倍的速度播放，也就是把播放频率调节为原来的两倍，这样音调就变高。
失去焦点
在很多情况下，其他程序会和你的程序抢占系统资源，然后把那些修改过配置的资源留给你的程序。这种情况多半发生在音频缓存上，所以需要调用IDirectSoundBuffer8::Restore来还原音频设置。如果缓冲区丢失，可以用这个函数找回。
The Restore method restores the memory allocation for a lost sound buffer.
HRESULT Restore();
Parameters
None.
Return Values
If the method succeeds, the return value is DS_OK. If the method fails, the
return value may be one of the following error values:
Return code
- DSERR_BUFFERLOST
- DSERR_INVALIDCALL
- DSERR_PRIOLEVELNEEDED
Remarks
If the application does not have the input focus,
IDirectSoundBuffer8::Restore might not succeed. For example, if the application
with the input focus has the DSSCL_WRITEPRIMARY cooperative level, no other
application will be able to restore its buffers. Similarly, an application with
the DSSCL_WRITEPRIMARY cooperative level must have the input focus to restore
its primary buffer.
After DirectSound restores the buffer memory, the application must rewrite
the buffer with valid sound data. DirectSound cannot restore the contents of the
memory, only the memory itself.
The application can receive notification that a buffer is lost when it
specifies that buffer in a call to the Lock or Play method. These methods return
DSERR_BUFFERLOST to indicate a lost buffer. The GetStatus method can also be
used to retrieve the status of the sound buffer and test for the
DSBSTATUS_BUFFERLOST flag.
使用这个函数会导致缓存中的音频数据丢失，调用完此函数后需要重新加载。在创建音频缓存的时候使用 DSBCAPS_LOCSOFTWARE标志，这样DirectSound将在系统内存中分配缓冲区，因此数据基本上不可能丢失，也就不必担心丢失资源了。
加载声音到音频缓冲
最简单的方法就是通过Windows 最广泛使用的数字音频文件 ----
波表文件，这种文件通常以.WAV作为它的扩展名。一个波表文件通常由两部分构成，一部分是文件开头的波表文件头，另外一部分是紧随其后的原始音频数据。这些原始音频数据可能是经过压缩的，也可能是未经压缩的。如果是压缩过的，操作起来会复杂很多，如果没有压缩过，操作起来就很容易。
下面的结构表示一个波表文件的文件头，通过观察能看出波表文件的文件头结构。
// .WAV file header
struct WAVE_HEADER
{
char riff_sig[4]; // 'RIFF'
long waveform_chunk_size; // 8
char wave_sig[4]; // 'WAVE'
char format_sig[4]; // 'fmt ' (notice space after)
long format_chunk_size; // 16;
short format_tag; // WAVE_FORMAT_PCM
short channels; // # of channels
long sample_rate; // sampling rate
long bytes_per_sec; // bytes per second
short block_align; // sample block alignment
short bits_per_sample; // bits per second
char data_sig[4]; // 'data'
long data_size; // size of waveform data
};
处理文件头非常简单，只需要打开文件，读取数据（读取数据的大小和WAVE_HEADER结构的大小一致）、填充 WAVE_HEADER结构就可以了。这个结构包含了我们所需要的所有关于音频文件的信息。你可以通过签名段来判断一个文件是否是波形文件，签名段在
WAVE_HEADER中是"*Sig"。请仔细查看 WAVE_HEADER中每个段的特征，如果不符合特征，说明所读取的不是一个波形文件。尤其是要检查签名段，如果签名段不是'WAVE'则说明加载了错误的音频文件。
有了必要的音频数据的结构信息后，就可以基于这些信息创建音频缓存，把音频数据放入其中，然后执行各种各样的操作。
可以编写两个函数来实现这样的功能，
Create_Buffer_From_WAV读取并解析波表文件头，并且创建单独的音频缓冲区
，Load_Sound_Data读取音频数据到缓冲区。
IDirectSound8* g_ds; // directsound component
IDirectSoundBuffer8* g_ds_buffer; // sound buffer object
//--------------------------------------------------------------------------------
// Create wave header information from wave file.
//--------------------------------------------------------------------------------
IDirectSoundBuffer8* Create_Buffer_From_WAV(FILE* fp, WAVE_HEADER* wave_header)
{
IDirectSoundBuffer* ds_buffer_main;
IDirectSoundBuffer8* ds_buffer_second;
DSBUFFERDESC ds_buffer_desc;
WAVEFORMATEX wave_format;
// read in the header from beginning of file
fseek(fp, 0, SEEK_SET);
fread(wave_header, 1, sizeof(WAVE_HEADER), fp);
// check the sig fields. returning if an error.
if(memcmp(wave_header->riff_sig, "RIFF", 4) || memcmp(wave_header->wave_sig, "WAVE", 4) ||
memcmp(wave_header->format_sig, "fmt ", 4) || memcmp(wave_header->data_sig, "data", 4))
{
return NULL;
}
// setup the playback format
ZeroMemory(&wave_format, sizeof(WAVEFORMATEX));
wave_format.wFormatTag = WAVE_FORMAT_PCM;
wave_format.nChannels = wave_header->channels;
wave_format.nSamplesPerSec = wave_header->sample_rate;
wave_format.wBitsPerSample = wave_header->bits_per_sample;
wave_format.nBlockAlign = wave_format.wBitsPerSample / 8 * wave_format.nChannels;
wave_format.nAvgBytesPerSec = wave_format.nSamplesPerSec * wave_format.nBlockAlign;
// create the sound buffer using the header data
ZeroMemory(&ds_buffer_desc, sizeof(DSBUFFERDESC));
ds_buffer_desc.dwSize = sizeof(DSBUFFERDESC);
ds_buffer_desc.dwFlags = DSBCAPS_CTRLVOLUME;
ds_buffer_desc.dwBufferBytes = wave_header->data_size;
ds_buffer_desc.lpwfxFormat = &wave_format;
// create main sound buffer
if(FAILED(g_ds->CreateSoundBuffer(&ds_buffer_desc, &ds_buffer_main, NULL)))
return NULL;
// get newer interface
if(FAILED(ds_buffer_main->QueryInterface(IID_IDirectSoundBuffer8, (void**)&ds_buffer_second)))
{
ds_buffer_main->Release();
return NULL;
}
// return the interface
return ds_buffer_second;
}
//--------------------------------------------------------------------------------
// Load sound data from second directsound buffer.
//--------------------------------------------------------------------------------
BOOL Load_Sound_Data(IDirectSoundBuffer8* ds_buffer, long lock_pos, long lock_size, FILE* fp)
{
BYTE* ptr1;
BYTE* ptr2;
DWORD size1, size2;
if(lock_size == 0)
return FALSE;
// lock the sound buffer at position specified
if(FAILED(ds_buffer->Lock(lock_pos, lock_size, (void**)&ptr1, &size1, (void**)&ptr2, &size2, 0)))
return FALSE;
// read in the data
fread(ptr1, 1, size1, fp);
if(ptr2 != NULL)
fread(ptr2, 1, size2, fp);
// unlock it
ds_buffer->Unlock(ptr1, size1, ptr2, size2);
return TRUE;
}
接着编写一个函数
Load_WAV封装刚才那两个函数，从文件名加载波形文件信息。
//--------------------------------------------------------------------------------
// Load wave file.
//--------------------------------------------------------------------------------
IDirectSoundBuffer8* Load_WAV(char* filename)
{
IDirectSoundBuffer8* ds_buffer;
WAVE_HEADER wave_header = {0};
FILE* fp;
// open the source file
if((fp = fopen(filename, "rb")) == NULL)
return NULL;
// create the sound buffer
if((ds_buffer = Create_Buffer_From_WAV(fp, &wave_header)) == NULL)
{
fclose(fp);
return NULL;
}
// read in the data
fseek(fp, sizeof(WAVE_HEADER), SEEK_SET);
// load sound data
Load_Sound_Data(ds_buffer, 0, wave_header.data_size, fp);
// close the source file
fclose(fp);
// return the new sound buffer fully loaded with sound
return ds_buffer;
}
以下给出完整示例：
点击下载源码和工程
/***************************************************************************************
PURPOSE:
Wave Playing Demo
***************************************************************************************/
#include <windows.h>
#include <stdio.h>
#include <dsound.h>
#include "resource.h"
#pragma comment(lib, "dxguid.lib")
#pragma comment(lib, "dsound.lib")
#pragma warning(disable : 4996)
#define Safe_Release(p) if((p)) (p)->Release();
// .WAV file header
struct WAVE_HEADER
{
char riff_sig[4]; // 'RIFF'
long waveform_chunk_size; // 8
char wave_sig[4]; // 'WAVE'
char format_sig[4]; // 'fmt ' (notice space after)
long format_chunk_size; // 16;
short format_tag; // WAVE_FORMAT_PCM
short channels; // # of channels
long sample_rate; // sampling rate
long bytes_per_sec; // bytes per second
short block_align; // sample block alignment
short bits_per_sample; // bits per second
char data_sig[4]; // 'data'
long data_size; // size of waveform data
};
// window handles, class and caption text.
HWND g_hwnd;
char g_class_name[] = "WavPlayClass";
IDirectSound8* g_ds; // directsound component
IDirectSoundBuffer8* g_ds_buffer; // sound buffer object
//--------------------------------------------------------------------------------
// Create wave header information from wave file.
//--------------------------------------------------------------------------------
IDirectSoundBuffer8* Create_Buffer_From_WAV(FILE* fp, WAVE_HEADER* wave_header)
{
IDirectSoundBuffer* ds_buffer_main;
IDirectSoundBuffer8* ds_buffer_second;
DSBUFFERDESC ds_buffer_desc;
WAVEFORMATEX wave_format;
// read in the header from beginning of file
fseek(fp, 0, SEEK_SET);
fread(wave_header, 1, sizeof(WAVE_HEADER), fp);
// check the sig fields. returning if an error.
if(memcmp(wave_header->riff_sig, "RIFF", 4) || memcmp(wave_header->wave_sig, "WAVE", 4) ||
memcmp(wave_header->format_sig, "fmt ", 4) || memcmp(wave_header->data_sig, "data", 4))
{
return NULL;
}
// setup the playback format
ZeroMemory(&wave_format, sizeof(WAVEFORMATEX));
wave_format.wFormatTag = WAVE_FORMAT_PCM;
wave_format.nChannels = wave_header->channels;
wave_format.nSamplesPerSec = wave_header->sample_rate;
wave_format.wBitsPerSample = wave_header->bits_per_sample;
wave_format.nBlockAlign = wave_format.wBitsPerSample / 8 * wave_format.nChannels;
wave_format.nAvgBytesPerSec = wave_format.nSamplesPerSec * wave_format.nBlockAlign;
// create the sound buffer using the header data
ZeroMemory(&ds_buffer_desc, sizeof(DSBUFFERDESC));
ds_buffer_desc.dwSize = sizeof(DSBUFFERDESC);
ds_buffer_desc.dwFlags = DSBCAPS_CTRLVOLUME;
ds_buffer_desc.dwBufferBytes = wave_header->data_size;
ds_buffer_desc.lpwfxFormat = &wave_format;
// create main sound buffer
if(FAILED(g_ds->CreateSoundBuffer(&ds_buffer_desc, &ds_buffer_main, NULL)))
return NULL;
// get newer interface
if(FAILED(ds_buffer_main->QueryInterface(IID_IDirectSoundBuffer8, (void**)&ds_buffer_second)))
{
ds_buffer_main->Release();
return NULL;
}
// return the interface
return ds_buffer_second;
}
//--------------------------------------------------------------------------------
// Load sound data from second directsound buffer.
//--------------------------------------------------------------------------------
BOOL Load_Sound_Data(IDirectSoundBuffer8* ds_buffer, long lock_pos, long lock_size, FILE* fp)
{
BYTE* ptr1;
BYTE* ptr2;
DWORD size1, size2;
if(lock_size == 0)
return FALSE;
// lock the sound buffer at position specified
if(FAILED(ds_buffer->Lock(lock_pos, lock_size, (void**)&ptr1, &size1, (void**)&ptr2, &size2, 0)))
return FALSE;
// read in the data
fread(ptr1, 1, size1, fp);
if(ptr2 != NULL)
fread(ptr2, 1, size2, fp);
// unlock it
ds_buffer->Unlock(ptr1, size1, ptr2, size2);
return TRUE;
}
//--------------------------------------------------------------------------------
// Load wave file.
//--------------------------------------------------------------------------------
IDirectSoundBuffer8* Load_WAV(char* filename)
{
IDirectSoundBuffer8* ds_buffer;
WAVE_HEADER wave_header = {0};
FILE* fp;
// open the source file
if((fp = fopen(filename, "rb")) == NULL)
return NULL;
// create the sound buffer
if((ds_buffer = Create_Buffer_From_WAV(fp, &wave_header)) == NULL)
{
fclose(fp);
return NULL;
}
// read in the data
fseek(fp, sizeof(WAVE_HEADER), SEEK_SET);
// load sound data
Load_Sound_Data(ds_buffer, 0, wave_header.data_size, fp);
// close the source file
fclose(fp);
// return the new sound buffer fully loaded with sound
return ds_buffer;
}
//--------------------------------------------------------------------------------
// Window procedure.
//--------------------------------------------------------------------------------
long WINAPI Window_Proc(HWND hwnd, UINT msg, WPARAM wParam, LPARAM lParam)
{
switch(msg)
{
case WM_DESTROY:
PostQuitMessage(0);
return 0;
}
return (long) DefWindowProc(hwnd, msg, wParam, lParam);
}
//--------------------------------------------------------------------------------
// Main function, routine entry.
//--------------------------------------------------------------------------------
int WINAPI WinMain(HINSTANCE inst, HINSTANCE, LPSTR cmd_line, int cmd_show)
{
WNDCLASS win_class;
MSG msg;
// create window class and register it
win_class.style = CS_HREDRAW | CS_VREDRAW;
win_class.lpfnWndProc = Window_Proc;
win_class.cbClsExtra = 0;
win_class.cbWndExtra = DLGWINDOWEXTRA;
win_class.hInstance = inst;
win_class.hIcon = LoadIcon(inst, IDI_APPLICATION);
win_class.hCursor = LoadCursor(NULL, IDC_ARROW);
win_class.hbrBackground = (HBRUSH) (COLOR_BTNFACE + 1);
win_class.lpszMenuName = NULL;
win_class.lpszClassName = g_class_name;
if(! RegisterClass(&win_class))
return FALSE;
// create the main window
g_hwnd = CreateDialog(inst, MAKEINTRESOURCE(IDD_WAVPLAY), 0, NULL);
ShowWindow(g_hwnd, cmd_show);
UpdateWindow(g_hwnd);
// initialize and configure directsound
// creates and initializes an object that supports the IDirectSound8 interface
if(FAILED(DirectSoundCreate8(NULL, &g_ds, NULL)))
{
MessageBox(NULL, "Unable to create DirectSound object", "Error", MB_OK);
return 0;
}
// set the cooperative level of the application for this sound device
g_ds->SetCooperativeLevel(g_hwnd, DSSCL_NORMAL);
// load a sound to play
g_ds_buffer = Load_WAV("test.wav");
if(g_ds_buffer)
{
// play sound looping
g_ds_buffer->SetCurrentPosition(0);
// set volume
g_ds_buffer->SetVolume(DSBVOLUME_MAX);
// play sound
g_ds_buffer->Play(0, 0, DSBPLAY_LOOPING);
}
// start message pump, waiting for signal to quit.
ZeroMemory(&msg, sizeof(MSG));
while(msg.message != WM_QUIT)
{
if(PeekMessage(&msg, NULL, 0, 0, PM_REMOVE))
{
TranslateMessage(&msg);
DispatchMessage(&msg);
}
}
// release directsound objects
g_ds->Release();
UnregisterClass(g_class_name, inst);
return (int) msg.wParam;
}
运行截图：
阅读下篇：
用DirectX
Audio和DirectShow播放声音和音乐（4）