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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
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <dsound.h>
#include <cguid.h>
#include <mmreg.h>
#ifndef _WAVEFORMATEXTENSIBLE_
#include <ks.h>
#include <ksmedia.h>
#endif
#include "alMain.h"
#include "AL/al.h"
#include "AL/alc.h"
#ifndef DSSPEAKER_5POINT1
#define DSSPEAKER_5POINT1 6
#endif
#ifndef DSSPEAKER_7POINT1
#define DSSPEAKER_7POINT1 7
#endif
DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
DEFINE_GUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, 0x00000003, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71);
static void *ds_handle;
static HRESULT (WINAPI *pDirectSoundCreate)(LPCGUID pcGuidDevice, IDirectSound **ppDS, IUnknown *pUnkOuter);
static HRESULT (WINAPI *pDirectSoundEnumerateA)(LPDSENUMCALLBACKA pDSEnumCallback, void *pContext);
static HRESULT (WINAPI *pDirectSoundCaptureCreate)(LPCGUID pcGuidDevice, IDirectSoundCapture **ppDSC, IUnknown *pUnkOuter);
static HRESULT (WINAPI *pDirectSoundCaptureEnumerateA)(LPDSENUMCALLBACKA pDSEnumCallback, void *pContext);
#define DirectSoundCreate pDirectSoundCreate
#define DirectSoundEnumerateA pDirectSoundEnumerateA
#define DirectSoundCaptureCreate pDirectSoundCaptureCreate
#define DirectSoundCaptureEnumerateA pDirectSoundCaptureEnumerateA
typedef struct {
// DirectSound Playback Device
IDirectSound *DS;
IDirectSoundBuffer *DSpbuffer;
IDirectSoundBuffer *DSsbuffer;
IDirectSoundNotify *DSnotify;
HANDLE NotifyEvent;
volatile int killNow;
ALvoid *thread;
} DSoundPlaybackData;
typedef struct {
// DirectSound Capture Device
IDirectSoundCapture *DSC;
IDirectSoundCaptureBuffer *DSCbuffer;
DWORD BufferBytes;
DWORD Cursor;
RingBuffer *Ring;
} DSoundCaptureData;
typedef struct {
ALCchar *name;
GUID guid;
} DevMap;
static DevMap *PlaybackDeviceList;
static ALuint NumPlaybackDevices;
static DevMap *CaptureDeviceList;
static ALuint NumCaptureDevices;
#define MAX_UPDATES 128
static ALCboolean DSoundLoad(void)
{
if(!ds_handle)
{
ds_handle = LoadLib("dsound.dll");
if(ds_handle == NULL)
{
ERR("Failed to load dsound.dll\n");
return ALC_FALSE;
}
#define LOAD_FUNC(f) do { \
p##f = GetSymbol(ds_handle, #f); \
if(p##f == NULL) { \
CloseLib(ds_handle); \
ds_handle = NULL; \
return ALC_FALSE; \
} \
} while(0)
LOAD_FUNC(DirectSoundCreate);
LOAD_FUNC(DirectSoundEnumerateA);
LOAD_FUNC(DirectSoundCaptureCreate);
LOAD_FUNC(DirectSoundCaptureEnumerateA);
#undef LOAD_FUNC
}
return ALC_TRUE;
}
static BOOL CALLBACK DSoundEnumPlaybackDevices(LPGUID guid, LPCSTR desc, LPCSTR drvname, LPVOID data)
{
LPOLESTR guidstr = NULL;
char str[1024];
HRESULT hr;
void *temp;
int count;
ALuint i;
(void)data;
(void)drvname;
if(!guid)
return TRUE;
count = 0;
do {
if(count == 0)
snprintf(str, sizeof(str), "%s", desc);
else
snprintf(str, sizeof(str), "%s #%d", desc, count+1);
count++;
for(i = 0;i < NumPlaybackDevices;i++)
{
if(strcmp(str, PlaybackDeviceList[i].name) == 0)
break;
}
} while(i != NumPlaybackDevices);
hr = StringFromCLSID(guid, &guidstr);
if(SUCCEEDED(hr))
{
TRACE("Got device \"%s\", GUID \"%ls\"\n", str, guidstr);
CoTaskMemFree(guidstr);
}
temp = realloc(PlaybackDeviceList, sizeof(DevMap) * (NumPlaybackDevices+1));
if(temp)
{
PlaybackDeviceList = temp;
PlaybackDeviceList[NumPlaybackDevices].name = strdup(str);
PlaybackDeviceList[NumPlaybackDevices].guid = *guid;
NumPlaybackDevices++;
}
return TRUE;
}
static BOOL CALLBACK DSoundEnumCaptureDevices(LPGUID guid, LPCSTR desc, LPCSTR drvname, LPVOID data)
{
LPOLESTR guidstr = NULL;
char str[1024];
HRESULT hr;
void *temp;
int count;
ALuint i;
(void)data;
(void)drvname;
if(!guid)
return TRUE;
count = 0;
do {
if(count == 0)
snprintf(str, sizeof(str), "%s", desc);
else
snprintf(str, sizeof(str), "%s #%d", desc, count+1);
count++;
for(i = 0;i < NumCaptureDevices;i++)
{
if(strcmp(str, CaptureDeviceList[i].name) == 0)
break;
}
} while(i != NumCaptureDevices);
hr = StringFromCLSID(guid, &guidstr);
if(SUCCEEDED(hr))
{
TRACE("Got device \"%s\", GUID \"%ls\"\n", str, guidstr);
CoTaskMemFree(guidstr);
}
temp = realloc(CaptureDeviceList, sizeof(DevMap) * (NumCaptureDevices+1));
if(temp)
{
CaptureDeviceList = temp;
CaptureDeviceList[NumCaptureDevices].name = strdup(str);
CaptureDeviceList[NumCaptureDevices].guid = *guid;
NumCaptureDevices++;
}
return TRUE;
}
static ALuint DSoundPlaybackProc(ALvoid *ptr)
{
ALCdevice *Device = (ALCdevice*)ptr;
DSoundPlaybackData *data = (DSoundPlaybackData*)Device->ExtraData;
DSBCAPS DSBCaps;
DWORD LastCursor = 0;
DWORD PlayCursor;
VOID *WritePtr1, *WritePtr2;
DWORD WriteCnt1, WriteCnt2;
BOOL Playing = FALSE;
DWORD FrameSize;
DWORD FragSize;
DWORD avail;
HRESULT err;
SetRTPriority();
memset(&DSBCaps, 0, sizeof(DSBCaps));
DSBCaps.dwSize = sizeof(DSBCaps);
err = IDirectSoundBuffer_GetCaps(data->DSsbuffer, &DSBCaps);
if(FAILED(err))
{
ERR("Failed to get buffer caps: 0x%lx\n", err);
aluHandleDisconnect(Device);
return 1;
}
FrameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
FragSize = Device->UpdateSize * FrameSize;
IDirectSoundBuffer_GetCurrentPosition(data->DSsbuffer, &LastCursor, NULL);
while(!data->killNow)
{
// Get current play cursor
IDirectSoundBuffer_GetCurrentPosition(data->DSsbuffer, &PlayCursor, NULL);
avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes;
if(avail < FragSize)
{
if(!Playing)
{
err = IDirectSoundBuffer_Play(data->DSsbuffer, 0, 0, DSBPLAY_LOOPING);
if(FAILED(err))
{
ERR("Failed to play buffer: 0x%lx\n", err);
aluHandleDisconnect(Device);
return 1;
}
Playing = TRUE;
}
avail = WaitForSingleObjectEx(data->NotifyEvent, 2000, FALSE);
if(avail != WAIT_OBJECT_0)
ERR("WaitForSingleObjectEx error: 0x%lx\n", avail);
continue;
}
avail -= avail%FragSize;
// Lock output buffer
WriteCnt1 = 0;
WriteCnt2 = 0;
err = IDirectSoundBuffer_Lock(data->DSsbuffer, LastCursor, avail, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
// If the buffer is lost, restore it and lock
if(err == DSERR_BUFFERLOST)
{
WARN("Buffer lost, restoring...\n");
err = IDirectSoundBuffer_Restore(data->DSsbuffer);
if(SUCCEEDED(err))
{
Playing = FALSE;
LastCursor = 0;
err = IDirectSoundBuffer_Lock(data->DSsbuffer, 0, DSBCaps.dwBufferBytes, &WritePtr1, &WriteCnt1, &WritePtr2, &WriteCnt2, 0);
}
}
// Successfully locked the output buffer
if(SUCCEEDED(err))
{
// If we have an active context, mix data directly into output buffer otherwise fill with silence
aluMixData(Device, WritePtr1, WriteCnt1/FrameSize);
aluMixData(Device, WritePtr2, WriteCnt2/FrameSize);
// Unlock output buffer only when successfully locked
IDirectSoundBuffer_Unlock(data->DSsbuffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2);
}
else
{
ERR("Buffer lock error: %#lx\n", err);
aluHandleDisconnect(Device);
return 1;
}
// Update old write cursor location
LastCursor += WriteCnt1+WriteCnt2;
LastCursor %= DSBCaps.dwBufferBytes;
}
return 0;
}
static ALCenum DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
{
DSoundPlaybackData *data = NULL;
LPGUID guid = NULL;
HRESULT hr;
if(!PlaybackDeviceList)
{
hr = DirectSoundEnumerateA(DSoundEnumPlaybackDevices, NULL);
if(FAILED(hr))
ERR("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr);
}
if(!deviceName && NumPlaybackDevices > 0)
{
deviceName = PlaybackDeviceList[0].name;
guid = &PlaybackDeviceList[0].guid;
}
else
{
ALuint i;
for(i = 0;i < NumPlaybackDevices;i++)
{
if(strcmp(deviceName, PlaybackDeviceList[i].name) == 0)
{
guid = &PlaybackDeviceList[i].guid;
break;
}
}
if(i == NumPlaybackDevices)
return ALC_INVALID_VALUE;
}
//Initialise requested device
data = calloc(1, sizeof(DSoundPlaybackData));
if(!data)
return ALC_OUT_OF_MEMORY;
hr = DS_OK;
data->NotifyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
if(data->NotifyEvent == NULL)
hr = E_FAIL;
//DirectSound Init code
if(SUCCEEDED(hr))
hr = DirectSoundCreate(guid, &data->DS, NULL);
if(SUCCEEDED(hr))
hr = IDirectSound_SetCooperativeLevel(data->DS, GetForegroundWindow(), DSSCL_PRIORITY);
if(FAILED(hr))
{
if(data->DS)
IDirectSound_Release(data->DS);
if(data->NotifyEvent)
CloseHandle(data->NotifyEvent);
free(data);
ERR("Device init failed: 0x%08lx\n", hr);
return ALC_INVALID_VALUE;
}
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void DSoundClosePlayback(ALCdevice *device)
{
DSoundPlaybackData *data = device->ExtraData;
if(data->DSnotify)
IDirectSoundNotify_Release(data->DSnotify);
data->DSnotify = NULL;
if(data->DSsbuffer)
IDirectSoundBuffer_Release(data->DSsbuffer);
data->DSsbuffer = NULL;
if(data->DSpbuffer != NULL)
IDirectSoundBuffer_Release(data->DSpbuffer);
data->DSpbuffer = NULL;
IDirectSound_Release(data->DS);
CloseHandle(data->NotifyEvent);
free(data);
device->ExtraData = NULL;
}
static ALCboolean DSoundResetPlayback(ALCdevice *device)
{
DSoundPlaybackData *data = (DSoundPlaybackData*)device->ExtraData;
DSBUFFERDESC DSBDescription;
WAVEFORMATEXTENSIBLE OutputType;
DWORD speakers;
HRESULT hr;
memset(&OutputType, 0, sizeof(OutputType));
if(data->DSnotify)
IDirectSoundNotify_Release(data->DSnotify);
data->DSnotify = NULL;
if(data->DSsbuffer)
IDirectSoundBuffer_Release(data->DSsbuffer);
data->DSsbuffer = NULL;
if(data->DSpbuffer != NULL)
IDirectSoundBuffer_Release(data->DSpbuffer);
data->DSpbuffer = NULL;
switch(device->FmtType)
{
case DevFmtByte:
device->FmtType = DevFmtUByte;
break;
case DevFmtFloat:
if((device->Flags&DEVICE_SAMPLE_TYPE_REQUEST))
break;
/* fall-through */
case DevFmtUShort:
device->FmtType = DevFmtShort;
break;
case DevFmtUInt:
device->FmtType = DevFmtInt;
break;
case DevFmtUByte:
case DevFmtShort:
case DevFmtInt:
break;
}
hr = IDirectSound_GetSpeakerConfig(data->DS, &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
ERR("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;
}
retry_open:
hr = S_OK;
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)
{
OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
OutputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
if(device->FmtType == DevFmtFloat)
OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
else
OutputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
if(data->DSpbuffer)
IDirectSoundBuffer_Release(data->DSpbuffer);
data->DSpbuffer = NULL;
}
else
{
if(SUCCEEDED(hr))
{
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_PRIMARYBUFFER;
hr = IDirectSound_CreateSoundBuffer(data->DS, &DSBDescription, &data->DSpbuffer, NULL);
}
if(SUCCEEDED(hr))
hr = IDirectSoundBuffer_SetFormat(data->DSpbuffer,&OutputType.Format);
}
if(SUCCEEDED(hr))
{
if(device->NumUpdates > MAX_UPDATES)
{
device->UpdateSize = (device->UpdateSize*device->NumUpdates +
MAX_UPDATES-1) / MAX_UPDATES;
device->NumUpdates = MAX_UPDATES;
}
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_CTRLPOSITIONNOTIFY|DSBCAPS_GETCURRENTPOSITION2|DSBCAPS_GLOBALFOCUS;
DSBDescription.dwBufferBytes=device->UpdateSize * device->NumUpdates *
OutputType.Format.nBlockAlign;
DSBDescription.lpwfxFormat=&OutputType.Format;
hr = IDirectSound_CreateSoundBuffer(data->DS, &DSBDescription, &data->DSsbuffer, NULL);
if(FAILED(hr) && device->FmtType == DevFmtFloat)
{
device->FmtType = DevFmtShort;
goto retry_open;
}
}
if(SUCCEEDED(hr))
{
hr = IDirectSoundBuffer_QueryInterface(data->DSsbuffer, &IID_IDirectSoundNotify, (LPVOID *)&data->DSnotify);
if(SUCCEEDED(hr))
{
DSBPOSITIONNOTIFY notifies[MAX_UPDATES];
ALuint i;
for(i = 0;i < device->NumUpdates;++i)
{
notifies[i].dwOffset = i * device->UpdateSize *
OutputType.Format.nBlockAlign;
notifies[i].hEventNotify = data->NotifyEvent;
}
if(IDirectSoundNotify_SetNotificationPositions(data->DSnotify, device->NumUpdates, notifies) != DS_OK)
hr = E_FAIL;
}
}
if(FAILED(hr))
{
if(data->DSnotify != NULL)
IDirectSoundNotify_Release(data->DSnotify);
data->DSnotify = NULL;
if(data->DSsbuffer != NULL)
IDirectSoundBuffer_Release(data->DSsbuffer);
data->DSsbuffer = NULL;
if(data->DSpbuffer != NULL)
IDirectSoundBuffer_Release(data->DSpbuffer);
data->DSpbuffer = NULL;
return ALC_FALSE;
}
ResetEvent(data->NotifyEvent);
SetDefaultWFXChannelOrder(device);
return ALC_TRUE;
}
static ALCboolean DSoundStartPlayback(ALCdevice *device)
{
DSoundPlaybackData *data = (DSoundPlaybackData*)device->ExtraData;
data->thread = StartThread(DSoundPlaybackProc, device);
if(data->thread == NULL)
return ALC_FALSE;
return ALC_TRUE;
}
static void DSoundStopPlayback(ALCdevice *device)
{
DSoundPlaybackData *data = device->ExtraData;
if(!data->thread)
return;
data->killNow = 1;
StopThread(data->thread);
data->thread = NULL;
data->killNow = 0;
IDirectSoundBuffer_Stop(data->DSsbuffer);
}
static ALCenum DSoundOpenCapture(ALCdevice *device, const ALCchar *deviceName)
{
DSoundCaptureData *data = NULL;
WAVEFORMATEXTENSIBLE InputType;
DSCBUFFERDESC DSCBDescription;
LPGUID guid = NULL;
HRESULT hr, hrcom;
ALuint samples;
if(!CaptureDeviceList)
{
/* Initialize COM to prevent name truncation */
hrcom = CoInitialize(NULL);
hr = DirectSoundCaptureEnumerateA(DSoundEnumCaptureDevices, NULL);
if(FAILED(hr))
ERR("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr);
if(SUCCEEDED(hrcom))
CoUninitialize();
}
if(!deviceName && NumCaptureDevices > 0)
{
deviceName = CaptureDeviceList[0].name;
guid = &CaptureDeviceList[0].guid;
}
else
{
ALuint i;
for(i = 0;i < NumCaptureDevices;i++)
{
if(strcmp(deviceName, CaptureDeviceList[i].name) == 0)
{
guid = &CaptureDeviceList[i].guid;
break;
}
}
if(i == NumCaptureDevices)
return ALC_INVALID_VALUE;
}
switch(device->FmtType)
{
case DevFmtByte:
case DevFmtUShort:
case DevFmtUInt:
WARN("%s capture samples not supported\n", DevFmtTypeString(device->FmtType));
return ALC_INVALID_ENUM;
case DevFmtUByte:
case DevFmtShort:
case DevFmtInt:
case DevFmtFloat:
break;
}
//Initialise requested device
data = calloc(1, sizeof(DSoundCaptureData));
if(!data)
return ALC_OUT_OF_MEMORY;
hr = DS_OK;
//DirectSoundCapture Init code
if(SUCCEEDED(hr))
hr = DirectSoundCaptureCreate(guid, &data->DSC, NULL);
if(SUCCEEDED(hr))
{
memset(&InputType, 0, sizeof(InputType));
switch(device->FmtChans)
{
case DevFmtMono:
InputType.dwChannelMask = SPEAKER_FRONT_CENTER;
break;
case DevFmtStereo:
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT;
break;
case DevFmtQuad:
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_BACK_LEFT |
SPEAKER_BACK_RIGHT;
break;
case DevFmtX51:
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_FRONT_CENTER |
SPEAKER_LOW_FREQUENCY |
SPEAKER_BACK_LEFT |
SPEAKER_BACK_RIGHT;
break;
case DevFmtX51Side:
InputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_FRONT_CENTER |
SPEAKER_LOW_FREQUENCY |
SPEAKER_SIDE_LEFT |
SPEAKER_SIDE_RIGHT;
break;
case DevFmtX61:
InputType.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:
InputType.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;
}
InputType.Format.wFormatTag = WAVE_FORMAT_PCM;
InputType.Format.nChannels = ChannelsFromDevFmt(device->FmtChans);
InputType.Format.wBitsPerSample = BytesFromDevFmt(device->FmtType) * 8;
InputType.Format.nBlockAlign = InputType.Format.nChannels*InputType.Format.wBitsPerSample/8;
InputType.Format.nSamplesPerSec = device->Frequency;
InputType.Format.nAvgBytesPerSec = InputType.Format.nSamplesPerSec*InputType.Format.nBlockAlign;
InputType.Format.cbSize = 0;
if(InputType.Format.nChannels > 2 || device->FmtType == DevFmtFloat)
{
InputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
InputType.Format.cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
InputType.Samples.wValidBitsPerSample = InputType.Format.wBitsPerSample;
if(device->FmtType == DevFmtFloat)
InputType.SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
else
InputType.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
}
samples = device->UpdateSize * device->NumUpdates;
samples = maxu(samples, 100 * device->Frequency / 1000);
memset(&DSCBDescription, 0, sizeof(DSCBUFFERDESC));
DSCBDescription.dwSize = sizeof(DSCBUFFERDESC);
DSCBDescription.dwFlags = 0;
DSCBDescription.dwBufferBytes = samples * InputType.Format.nBlockAlign;
DSCBDescription.lpwfxFormat = &InputType.Format;
hr = IDirectSoundCapture_CreateCaptureBuffer(data->DSC, &DSCBDescription, &data->DSCbuffer, NULL);
}
if(SUCCEEDED(hr))
{
data->Ring = CreateRingBuffer(InputType.Format.nBlockAlign, device->UpdateSize * device->NumUpdates);
if(data->Ring == NULL)
hr = DSERR_OUTOFMEMORY;
}
if(FAILED(hr))
{
ERR("Device init failed: 0x%08lx\n", hr);
DestroyRingBuffer(data->Ring);
data->Ring = NULL;
if(data->DSCbuffer != NULL)
IDirectSoundCaptureBuffer_Release(data->DSCbuffer);
data->DSCbuffer = NULL;
if(data->DSC)
IDirectSoundCapture_Release(data->DSC);
data->DSC = NULL;
free(data);
return ALC_INVALID_VALUE;
}
data->BufferBytes = DSCBDescription.dwBufferBytes;
SetDefaultWFXChannelOrder(device);
device->DeviceName = strdup(deviceName);
device->ExtraData = data;
return ALC_NO_ERROR;
}
static void DSoundCloseCapture(ALCdevice *device)
{
DSoundCaptureData *data = device->ExtraData;
DestroyRingBuffer(data->Ring);
data->Ring = NULL;
if(data->DSCbuffer != NULL)
{
IDirectSoundCaptureBuffer_Stop(data->DSCbuffer);
IDirectSoundCaptureBuffer_Release(data->DSCbuffer);
data->DSCbuffer = NULL;
}
IDirectSoundCapture_Release(data->DSC);
data->DSC = NULL;
free(data);
device->ExtraData = NULL;
}
static void DSoundStartCapture(ALCdevice *device)
{
DSoundCaptureData *data = device->ExtraData;
HRESULT hr;
hr = IDirectSoundCaptureBuffer_Start(data->DSCbuffer, DSCBSTART_LOOPING);
if(FAILED(hr))
{
ERR("start failed: 0x%08lx\n", hr);
aluHandleDisconnect(device);
}
}
static void DSoundStopCapture(ALCdevice *device)
{
DSoundCaptureData *data = device->ExtraData;
HRESULT hr;
hr = IDirectSoundCaptureBuffer_Stop(data->DSCbuffer);
if(FAILED(hr))
{
ERR("stop failed: 0x%08lx\n", hr);
aluHandleDisconnect(device);
}
}
static ALCenum DSoundCaptureSamples(ALCdevice *Device, ALCvoid *pBuffer, ALCuint lSamples)
{
DSoundCaptureData *data = Device->ExtraData;
ReadRingBuffer(data->Ring, pBuffer, lSamples);
return ALC_NO_ERROR;
}
static ALCuint DSoundAvailableSamples(ALCdevice *Device)
{
DSoundCaptureData *data = Device->ExtraData;
DWORD ReadCursor, LastCursor, BufferBytes, NumBytes;
VOID *ReadPtr1, *ReadPtr2;
DWORD ReadCnt1, ReadCnt2;
DWORD FrameSize;
HRESULT hr;
if(!Device->Connected)
goto done;
FrameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType);
BufferBytes = data->BufferBytes;
LastCursor = data->Cursor;
hr = IDirectSoundCaptureBuffer_GetCurrentPosition(data->DSCbuffer, NULL, &ReadCursor);
if(SUCCEEDED(hr))
{
NumBytes = (ReadCursor-LastCursor + BufferBytes) % BufferBytes;
if(NumBytes == 0)
goto done;
hr = IDirectSoundCaptureBuffer_Lock(data->DSCbuffer, LastCursor, NumBytes,
&ReadPtr1, &ReadCnt1,
&ReadPtr2, &ReadCnt2, 0);
}
if(SUCCEEDED(hr))
{
WriteRingBuffer(data->Ring, ReadPtr1, ReadCnt1/FrameSize);
if(ReadPtr2 != NULL)
WriteRingBuffer(data->Ring, ReadPtr2, ReadCnt2/FrameSize);
hr = IDirectSoundCaptureBuffer_Unlock(data->DSCbuffer,
ReadPtr1, ReadCnt1,
ReadPtr2, ReadCnt2);
data->Cursor = (LastCursor+ReadCnt1+ReadCnt2) % BufferBytes;
}
if(FAILED(hr))
{
ERR("update failed: 0x%08lx\n", hr);
aluHandleDisconnect(Device);
}
done:
return RingBufferSize(data->Ring);
}
static const BackendFuncs DSoundFuncs = {
DSoundOpenPlayback,
DSoundClosePlayback,
DSoundResetPlayback,
DSoundStartPlayback,
DSoundStopPlayback,
DSoundOpenCapture,
DSoundCloseCapture,
DSoundStartCapture,
DSoundStopCapture,
DSoundCaptureSamples,
DSoundAvailableSamples,
ALCdevice_LockDefault,
ALCdevice_UnlockDefault,
ALCdevice_GetLatencyDefault
};
ALCboolean alcDSoundInit(BackendFuncs *FuncList)
{
if(!DSoundLoad())
return ALC_FALSE;
*FuncList = DSoundFuncs;
return ALC_TRUE;
}
void alcDSoundDeinit(void)
{
ALuint i;
for(i = 0;i < NumPlaybackDevices;++i)
free(PlaybackDeviceList[i].name);
free(PlaybackDeviceList);
PlaybackDeviceList = NULL;
NumPlaybackDevices = 0;
for(i = 0;i < NumCaptureDevices;++i)
free(CaptureDeviceList[i].name);
free(CaptureDeviceList);
CaptureDeviceList = NULL;
NumCaptureDevices = 0;
if(ds_handle)
CloseLib(ds_handle);
ds_handle = NULL;
}
void alcDSoundProbe(enum DevProbe type)
{
HRESULT hr, hrcom;
ALuint i;
switch(type)
{
case ALL_DEVICE_PROBE:
for(i = 0;i < NumPlaybackDevices;++i)
free(PlaybackDeviceList[i].name);
free(PlaybackDeviceList);
PlaybackDeviceList = NULL;
NumPlaybackDevices = 0;
hr = DirectSoundEnumerateA(DSoundEnumPlaybackDevices, NULL);
if(FAILED(hr))
ERR("Error enumerating DirectSound playback devices (%#x)!\n", (unsigned int)hr);
else
{
for(i = 0;i < NumPlaybackDevices;i++)
AppendAllDevicesList(PlaybackDeviceList[i].name);
}
break;
case CAPTURE_DEVICE_PROBE:
for(i = 0;i < NumCaptureDevices;++i)
free(CaptureDeviceList[i].name);
free(CaptureDeviceList);
CaptureDeviceList = NULL;
NumCaptureDevices = 0;
/* Initialize COM to prevent name truncation */
hrcom = CoInitialize(NULL);
hr = DirectSoundCaptureEnumerateA(DSoundEnumCaptureDevices, NULL);
if(FAILED(hr))
ERR("Error enumerating DirectSound capture devices (%#x)!\n", (unsigned int)hr);
else
{
for(i = 0;i < NumCaptureDevices;i++)
AppendCaptureDeviceList(CaptureDeviceList[i].name);
}
if(SUCCEEDED(hrcom))
CoUninitialize();
break;
}
}
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