AuroraOpenALSoft/Alc/dsound.c

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/**
* OpenAL cross platform audio library
* Copyright (C) 1999-2007 by authors.
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
* Or go to http://www.gnu.org/copyleft/lgpl.html
*/
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#include "config.h"
#define _WIN32_WINNT 0x0500
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#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <dsound.h>
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#include <cguid.h>
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#include <mmreg.h>
#ifndef _WAVEFORMATEXTENSIBLE_
#include <ks.h>
#include <ksmedia.h>
#endif
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#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
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#ifndef DSSPEAKER_5POINT1
#define DSSPEAKER_5POINT1 6
#endif
#ifndef DSSPEAKER_7POINT1
#define DSSPEAKER_7POINT1 7
#endif
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DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
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DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
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static void *ds_handle;
static HRESULT (WINAPI *pDirectSoundCreate)(LPCGUID pcGuidDevice, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter);
static HRESULT (WINAPI *pDirectSoundEnumerateA)(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext);
#define DirectSoundCreate pDirectSoundCreate
#define DirectSoundEnumerateA pDirectSoundEnumerateA
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typedef struct {
// DirectSound Playback Device
LPDIRECTSOUND lpDS;
LPDIRECTSOUNDBUFFER DSpbuffer;
LPDIRECTSOUNDBUFFER DSsbuffer;
volatile int killNow;
ALvoid *thread;
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} DSoundData;
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typedef struct {
ALCchar *name;
GUID guid;
} DevMap;
static const ALCchar dsDevice[] = "DirectSound Default";
static DevMap *DeviceList;
static ALuint NumDevices;
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static void *DSoundLoad(void)
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{
if(!ds_handle)
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{
ALboolean failed = AL_FALSE;
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ds_handle = LoadLibraryA("dsound.dll");
if(ds_handle == NULL)
{
ERROR("Failed to load dsound.dll\n");
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return NULL;
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}
#define LOAD_FUNC(x) do { \
if((p##x = (void*)GetProcAddress((HMODULE)ds_handle, #x)) == NULL) { \
ERROR("Could not load %s from dsound.dll\n", #x); \
failed = AL_TRUE; \
} \
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} while(0)
LOAD_FUNC(DirectSoundCreate);
LOAD_FUNC(DirectSoundEnumerateA);
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#undef LOAD_FUNC
if(failed)
{
FreeLibrary(ds_handle);
ds_handle = NULL;
}
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}
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return ds_handle;
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}
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static BOOL CALLBACK DSoundEnumDevices(LPGUID guid, LPCSTR desc, LPCSTR drvname, LPVOID data)
{
char str[1024];
void *temp;
int count;
ALuint i;
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(void)data;
(void)drvname;
if(NumDevices == 0)
{
temp = realloc(DeviceList, sizeof(DevMap) * (NumDevices+1));
if(temp)
{
DeviceList = temp;
DeviceList[NumDevices].name = strdup(dsDevice);
DeviceList[NumDevices].guid = GUID_NULL;
NumDevices++;
}
}
if(!guid)
return TRUE;
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count = 0;
do {
if(count == 0)
snprintf(str, sizeof(str), "%s via DirectSound", desc);
else
snprintf(str, sizeof(str), "%s #%d via DirectSound", desc, count+1);
count++;
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for(i = 0;i < NumDevices;i++)
{
if(strcmp(str, DeviceList[i].name) == 0)
break;
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}
} while(i != NumDevices);
temp = realloc(DeviceList, sizeof(DevMap) * (NumDevices+1));
if(temp)
{
DeviceList = temp;
DeviceList[NumDevices].name = strdup(str);
DeviceList[NumDevices].guid = *guid;
NumDevices++;
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}
return TRUE;
}
static ALuint DSoundProc(ALvoid *ptr)
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{
ALCdevice *pDevice = (ALCdevice*)ptr;
DSoundData *pData = (DSoundData*)pDevice->ExtraData;
DSBCAPS DSBCaps;
DWORD LastCursor = 0;
DWORD PlayCursor;
VOID *WritePtr1, *WritePtr2;
DWORD WriteCnt1, WriteCnt2;
BOOL Playing = FALSE;
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DWORD FrameSize;
DWORD FragSize;
DWORD avail;
HRESULT err;
SetRTPriority();
memset(&DSBCaps, 0, sizeof(DSBCaps));
DSBCaps.dwSize = sizeof(DSBCaps);
err = IDirectSoundBuffer_GetCaps(pData->DSsbuffer, &DSBCaps);
if(FAILED(err))
{
ERROR("Failed to get buffer caps: 0x%lx\n", err);
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aluHandleDisconnect(pDevice);
return 1;
}
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FrameSize = FrameSizeFromDevFmt(pDevice->FmtChans, pDevice->FmtType);
FragSize = pDevice->UpdateSize * FrameSize;
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IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &LastCursor, NULL);
while(!pData->killNow)
{
// Get current play and write cursors
IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &PlayCursor, NULL);
avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes;
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if(avail < FragSize)
{
if(!Playing)
{
err = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING);
if(FAILED(err))
{
ERROR("Failed to play buffer: 0x%lx\n", err);
aluHandleDisconnect(pDevice);
return 1;
}
Playing = TRUE;
}
Sleep(1);
continue;
}
avail -= avail%FragSize;
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// Lock output buffer
WriteCnt1 = 0;
WriteCnt2 = 0;
err = IDirectSoundBuffer_Lock(pData->DSsbuffer, LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
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// If the buffer is lost, restore it and lock
if(err == DSERR_BUFFERLOST)
{
WARN("Buffer lost, restoring...\n");
err = IDirectSoundBuffer_Restore(pData->DSsbuffer);
if(SUCCEEDED(err))
{
Playing = FALSE;
LastCursor = 0;
err = IDirectSoundBuffer_Lock(pData->DSsbuffer, 0, DSBCaps.dwBufferBytes, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
}
}
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// Successfully locked the output buffer
if(SUCCEEDED(err))
{
// If we have an active context, mix data directly into output buffer otherwise fill with silence
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aluMixData(pDevice, WritePtr1, WriteCnt1/FrameSize);
aluMixData(pDevice, WritePtr2, WriteCnt2/FrameSize);
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// Unlock output buffer only when successfully locked
IDirectSoundBuffer_Unlock(pData->DSsbuffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2);
}
else
{
ERROR("Buffer lock error: %#lx\n", err);
aluHandleDisconnect(pDevice);
return 1;
}
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// Update old write cursor location
LastCursor += WriteCnt1+WriteCnt2;
LastCursor %= DSBCaps.dwBufferBytes;
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}
return 0;
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}
static ALCboolean DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
{
DSoundData *pData = NULL;
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LPGUID guid = NULL;
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HRESULT hr;
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if(!DSoundLoad())
return ALC_FALSE;
if(!deviceName)
deviceName = dsDevice;
else if(strcmp(deviceName, dsDevice) != 0)
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{
ALuint i;
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if(!DeviceList)
{
hr = DirectSoundEnumerateA(DSoundEnumDevices, NULL);
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if(FAILED(hr))
ERROR("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr);
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}
for(i = 0;i < NumDevices;i++)
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{
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if(strcmp(deviceName, DeviceList[i].name) == 0)
{
if(i > 0)
guid = &DeviceList[i].guid;
break;
}
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}
if(i == NumDevices)
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return ALC_FALSE;
}
//Initialise requested device
pData = calloc(1, sizeof(DSoundData));
if(!pData)
{
alcSetError(device, ALC_OUT_OF_MEMORY);
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return ALC_FALSE;
}
//DirectSound Init code
hr = DirectSoundCreate(guid, &pData->lpDS, NULL);
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if(SUCCEEDED(hr))
hr = IDirectSound_SetCooperativeLevel(pData->lpDS, GetForegroundWindow(), DSSCL_PRIORITY);
if(FAILED(hr))
{
if(pData->lpDS)
IDirectSound_Release(pData->lpDS);
free(pData);
ERROR("Device init failed: 0x%08lx\n", hr);
return ALC_FALSE;
}
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device->szDeviceName = strdup(deviceName);
device->ExtraData = pData;
return ALC_TRUE;
}
static void DSoundClosePlayback(ALCdevice *device)
{
DSoundData *pData = device->ExtraData;
IDirectSound_Release(pData->lpDS);
free(pData);
device->ExtraData = NULL;
}
static ALCboolean DSoundResetPlayback(ALCdevice *device)
{
DSoundData *pData = (DSoundData*)device->ExtraData;
DSBUFFERDESC DSBDescription;
WAVEFORMATEXTENSIBLE OutputType;
DWORD speakers;
HRESULT hr;
memset(&OutputType, 0, sizeof(OutputType));
switch(device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
break;
case DevFmtUShort:
device->FmtType = DevFmtShort;
break;
case DevFmtUByte:
case DevFmtShort:
case DevFmtFloat:
break;
}
hr = IDirectSound_GetSpeakerConfig(pData->lpDS, &speakers);
if(SUCCEEDED(hr))
{
if(!(device->Flags&DEVICE_CHANNELS_REQUEST))
{
speakers = DSSPEAKER_CONFIG(speakers);
if(speakers == DSSPEAKER_MONO)
device->FmtChans = DevFmtMono;
else if(speakers == DSSPEAKER_STEREO || speakers == DSSPEAKER_HEADPHONE)
device->FmtChans = DevFmtStereo;
else if(speakers == DSSPEAKER_QUAD)
device->FmtChans = DevFmtQuad;
else if(speakers == DSSPEAKER_5POINT1)
device->FmtChans = DevFmtX51;
else if(speakers == DSSPEAKER_7POINT1)
device->FmtChans = DevFmtX71;
else
ERROR("Unknown system speaker config: 0x%lx\n", speakers);
}
switch(device->FmtChans)
{
case DevFmtMono:
OutputType.dwChannelMask = SPEAKER_FRONT_CENTER;
break;
case DevFmtStereo:
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT;
break;
case DevFmtQuad:
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_BACK_LEFT |
SPEAKER_BACK_RIGHT;
break;
case DevFmtX51:
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_FRONT_CENTER |
SPEAKER_LOW_FREQUENCY |
SPEAKER_BACK_LEFT |
SPEAKER_BACK_RIGHT;
break;
case DevFmtX51Side:
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_FRONT_CENTER |
SPEAKER_LOW_FREQUENCY |
SPEAKER_SIDE_LEFT |
SPEAKER_SIDE_RIGHT;
break;
case DevFmtX61:
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_FRONT_CENTER |
SPEAKER_LOW_FREQUENCY |
SPEAKER_BACK_CENTER |
SPEAKER_SIDE_LEFT |
SPEAKER_SIDE_RIGHT;
break;
case DevFmtX71:
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_FRONT_CENTER |
SPEAKER_LOW_FREQUENCY |
SPEAKER_BACK_LEFT |
SPEAKER_BACK_RIGHT |
SPEAKER_SIDE_LEFT |
SPEAKER_SIDE_RIGHT;
break;
}
OutputType.Format.wFormatTag = WAVE_FORMAT_PCM;
OutputType.Format.nChannels = ChannelsFromDevFmt(device->FmtChans);
OutputType.Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8;
OutputType.Format.nBlockAlign = OutputType.Format.nChannels*OutputType.Format.wBitsPerSample/8;
OutputType.Format.nSamplesPerSec = device->Frequency;
OutputType.Format.nAvgBytesPerSec = OutputType.Format.nSamplesPerSec*OutputType.Format.nBlockAlign;
OutputType.Format.cbSize = 0;
}
if(OutputType.Format.nChannels > 2 || device->FmtType == DevFmtFloat)
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{
OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
OutputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
if(device->FmtType == DevFmtFloat)
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OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
else
OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
}
else
{
if(SUCCEEDED(hr))
{
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_PRIMARYBUFFER;
hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSpbuffer, NULL);
}
if(SUCCEEDED(hr))
hr = IDirectSoundBuffer_SetFormat(pData->DSpbuffer,&OutputType.Format);
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}
if(SUCCEEDED(hr))
{
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_GLOBALFOCUS|DSBCAPS_GETCURRENTPOSITION2;
DSBDescription.dwBufferBytes=device->UpdateSize * device->NumUpdates *
OutputType.Format.nBlockAlign;
DSBDescription.lpwfxFormat=&OutputType.Format;
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hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSsbuffer, NULL);
}
if(SUCCEEDED(hr))
{
SetDefaultWFXChannelOrder(device);
pData->thread = StartThread(DSoundProc, device);
if(!pData->thread)
hr = E_FAIL;
}
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if(FAILED(hr))
{
if (pData->DSsbuffer)
IDirectSoundBuffer_Release(pData->DSsbuffer);
pData->DSsbuffer = NULL;
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if (pData->DSpbuffer)
IDirectSoundBuffer_Release(pData->DSpbuffer);
pData->DSpbuffer = NULL;
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return ALC_FALSE;
}
return ALC_TRUE;
}
static void DSoundStopPlayback(ALCdevice *device)
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{
DSoundData *pData = device->ExtraData;
if(!pData->thread)
return;
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pData->killNow = 1;
StopThread(pData->thread);
pData->thread = NULL;
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pData->killNow = 0;
IDirectSoundBuffer_Release(pData->DSsbuffer);
pData->DSsbuffer = NULL;
if (pData->DSpbuffer)
IDirectSoundBuffer_Release(pData->DSpbuffer);
pData->DSpbuffer = NULL;
}
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static ALCboolean DSoundOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName)
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{
(void)pDevice;
(void)deviceName;
return ALC_FALSE;
}
static void DSoundCloseCapture(ALCdevice *pDevice)
{
(void)pDevice;
}
static void DSoundStartCapture(ALCdevice *pDevice)
{
(void)pDevice;
}
static void DSoundStopCapture(ALCdevice *pDevice)
{
(void)pDevice;
}
static void DSoundCaptureSamples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples)
{
(void)pDevice;
(void)pBuffer;
(void)lSamples;
}
static ALCuint DSoundAvailableSamples(ALCdevice *pDevice)
{
(void)pDevice;
return 0;
}
static const BackendFuncs DSoundFuncs = {
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DSoundOpenPlayback,
DSoundClosePlayback,
DSoundResetPlayback,
DSoundStopPlayback,
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DSoundOpenCapture,
DSoundCloseCapture,
DSoundStartCapture,
DSoundStopCapture,
DSoundCaptureSamples,
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DSoundAvailableSamples
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};
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void alcDSoundInit(BackendFuncs *FuncList)
{
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*FuncList = DSoundFuncs;
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}
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void alcDSoundDeinit(void)
{
ALuint i;
for(i = 0;i < NumDevices;++i)
free(DeviceList[i].name);
free(DeviceList);
DeviceList = NULL;
NumDevices = 0;
if(ds_handle)
FreeLibrary(ds_handle);
ds_handle = NULL;
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}
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void alcDSoundProbe(enum DevProbe type)
{
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HRESULT hr;
ALuint i;
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if(!DSoundLoad()) return;
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switch(type)
{
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case DEVICE_PROBE:
AppendDeviceList(dsDevice);
break;
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case ALL_DEVICE_PROBE:
for(i = 0;i < NumDevices;++i)
free(DeviceList[i].name);
free(DeviceList);
DeviceList = NULL;
NumDevices = 0;
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hr = DirectSoundEnumerateA(DSoundEnumDevices, NULL);
if(FAILED(hr))
ERROR("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr);
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else
{
for(i = 0;i < NumDevices;i++)
AppendAllDeviceList(DeviceList[i].name);
}
break;
case CAPTURE_DEVICE_PROBE:
break;
}
}