/** * 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 #include #include #include #include #include #ifndef _WAVEFORMATEXTENSIBLE_ #include #include #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 *lpDS; IDirectSoundBuffer *DSpbuffer; IDirectSoundBuffer *DSsbuffer; IDirectSoundNotify *DSnotify; HANDLE hNotifyEvent; volatile int killNow; ALvoid *thread; } DSoundPlaybackData; typedef struct { // DirectSound Capture Device IDirectSoundCapture *lpDSC; IDirectSoundCaptureBuffer *DSCbuffer; DWORD dwBufferBytes; DWORD dwCursor; RingBuffer *pRing; } DSoundCaptureData; typedef struct { ALCchar *name; GUID guid; } DevMap; static const ALCchar dsDevice[] = "DirectSound Default"; 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) { char str[1024]; 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); 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) { char str[1024]; 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); 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 *pDevice = (ALCdevice*)ptr; DSoundPlaybackData *pData = (DSoundPlaybackData*)pDevice->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(pData->DSsbuffer, &DSBCaps); if(FAILED(err)) { ERR("Failed to get buffer caps: 0x%lx\n", err); aluHandleDisconnect(pDevice); return 1; } FrameSize = FrameSizeFromDevFmt(pDevice->FmtChans, pDevice->FmtType); FragSize = pDevice->UpdateSize * FrameSize; IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &LastCursor, NULL); while(!pData->killNow) { // Get current play cursor IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &PlayCursor, NULL); avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes; if(avail < FragSize) { if(!Playing) { err = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING); if(FAILED(err)) { ERR("Failed to play buffer: 0x%lx\n", err); aluHandleDisconnect(pDevice); return 1; } Playing = TRUE; } avail = WaitForSingleObjectEx(pData->hNotifyEvent, 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(pData->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(pData->DSsbuffer); if(SUCCEEDED(err)) { Playing = FALSE; LastCursor = 0; err = IDirectSoundBuffer_Lock(pData->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(pDevice, WritePtr1, WriteCnt1/FrameSize); aluMixData(pDevice, WritePtr2, WriteCnt2/FrameSize); // Unlock output buffer only when successfully locked IDirectSoundBuffer_Unlock(pData->DSsbuffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2); } else { ERR("Buffer lock error: %#lx\n", err); aluHandleDisconnect(pDevice); return 1; } // Update old write cursor location LastCursor += WriteCnt1+WriteCnt2; LastCursor %= DSBCaps.dwBufferBytes; } return 0; } static ALCenum DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName) { DSoundPlaybackData *pData = NULL; LPGUID guid = NULL; HRESULT hr; if(!deviceName) deviceName = dsDevice; else if(strcmp(deviceName, dsDevice) != 0) { ALuint i; if(!PlaybackDeviceList) { hr = DirectSoundEnumerateA(DSoundEnumPlaybackDevices, NULL); if(FAILED(hr)) ERR("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr); } 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 pData = calloc(1, sizeof(DSoundPlaybackData)); if(!pData) return ALC_OUT_OF_MEMORY; hr = DS_OK; pData->hNotifyEvent = CreateEvent(NULL, FALSE, FALSE, NULL); if(pData->hNotifyEvent == NULL) hr = E_FAIL; //DirectSound Init code if(SUCCEEDED(hr)) hr = DirectSoundCreate(guid, &pData->lpDS, NULL); if(SUCCEEDED(hr)) hr = IDirectSound_SetCooperativeLevel(pData->lpDS, GetForegroundWindow(), DSSCL_PRIORITY); if(FAILED(hr)) { if(pData->lpDS) IDirectSound_Release(pData->lpDS); if(pData->hNotifyEvent) CloseHandle(pData->hNotifyEvent); free(pData); ERR("Device init failed: 0x%08lx\n", hr); return ALC_INVALID_VALUE; } device->szDeviceName = strdup(deviceName); device->ExtraData = pData; return ALC_NO_ERROR; } static void DSoundClosePlayback(ALCdevice *device) { DSoundPlaybackData *pData = device->ExtraData; IDirectSound_Release(pData->lpDS); CloseHandle(pData->hNotifyEvent); free(pData); device->ExtraData = NULL; } static ALCboolean DSoundResetPlayback(ALCdevice *device) { DSoundPlaybackData *pData = (DSoundPlaybackData*)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 DevFmtUInt: device->FmtType = DevFmtInt; break; case DevFmtUByte: case DevFmtShort: case DevFmtInt: 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 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(pData->DSpbuffer) IDirectSoundBuffer_Release(pData->DSpbuffer); pData->DSpbuffer = NULL; } 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); } 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(pData->lpDS, &DSBDescription, &pData->DSsbuffer, NULL); if(FAILED(hr) && device->FmtType == DevFmtFloat) { device->FmtType = DevFmtShort; goto retry_open; } } if(SUCCEEDED(hr)) { hr = IDirectSoundBuffer_QueryInterface(pData->DSsbuffer, &IID_IDirectSoundNotify, (LPVOID *)&pData->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 = pData->hNotifyEvent; } if(IDirectSoundNotify_SetNotificationPositions(pData->DSnotify, device->NumUpdates, notifies) != DS_OK) hr = E_FAIL; } } if(SUCCEEDED(hr)) { ResetEvent(pData->hNotifyEvent); SetDefaultWFXChannelOrder(device); pData->thread = StartThread(DSoundPlaybackProc, device); if(pData->thread == NULL) hr = E_FAIL; } if(FAILED(hr)) { if(pData->DSnotify != NULL) IDirectSoundNotify_Release(pData->DSnotify); pData->DSnotify = NULL; if(pData->DSsbuffer != NULL) IDirectSoundBuffer_Release(pData->DSsbuffer); pData->DSsbuffer = NULL; if(pData->DSpbuffer != NULL) IDirectSoundBuffer_Release(pData->DSpbuffer); pData->DSpbuffer = NULL; return ALC_FALSE; } return ALC_TRUE; } static void DSoundStopPlayback(ALCdevice *device) { DSoundPlaybackData *pData = device->ExtraData; if(!pData->thread) return; pData->killNow = 1; StopThread(pData->thread); pData->thread = NULL; pData->killNow = 0; IDirectSoundNotify_Release(pData->DSnotify); pData->DSnotify = NULL; IDirectSoundBuffer_Release(pData->DSsbuffer); pData->DSsbuffer = NULL; if(pData->DSpbuffer != NULL) IDirectSoundBuffer_Release(pData->DSpbuffer); pData->DSpbuffer = NULL; } static ALCenum DSoundOpenCapture(ALCdevice *device, const ALCchar *deviceName) { DSoundCaptureData *pData = 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; } //Initialise requested device pData = calloc(1, sizeof(DSoundCaptureData)); if(!pData) return ALC_OUT_OF_MEMORY; hr = DS_OK; //DirectSoundCapture Init code if(SUCCEEDED(hr)) hr = DirectSoundCaptureCreate(guid, &pData->lpDSC, 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(pData->lpDSC, &DSCBDescription, &pData->DSCbuffer, NULL); } if(SUCCEEDED(hr)) { pData->pRing = CreateRingBuffer(InputType.Format.nBlockAlign, device->UpdateSize * device->NumUpdates); if(pData->pRing == NULL) hr = DSERR_OUTOFMEMORY; } if(FAILED(hr)) { ERR("Device init failed: 0x%08lx\n", hr); DestroyRingBuffer(pData->pRing); pData->pRing = NULL; if(pData->DSCbuffer != NULL) IDirectSoundCaptureBuffer_Release(pData->DSCbuffer); pData->DSCbuffer = NULL; if(pData->lpDSC) IDirectSoundCapture_Release(pData->lpDSC); pData->lpDSC = NULL; free(pData); return ALC_INVALID_VALUE; } pData->dwBufferBytes = DSCBDescription.dwBufferBytes; SetDefaultWFXChannelOrder(device); device->szDeviceName = strdup(deviceName); device->ExtraData = pData; return ALC_NO_ERROR; } static void DSoundCloseCapture(ALCdevice *device) { DSoundCaptureData *pData = device->ExtraData; DestroyRingBuffer(pData->pRing); pData->pRing = NULL; if(pData->DSCbuffer != NULL) { IDirectSoundCaptureBuffer_Stop(pData->DSCbuffer); IDirectSoundCaptureBuffer_Release(pData->DSCbuffer); pData->DSCbuffer = NULL; } IDirectSoundCapture_Release(pData->lpDSC); pData->lpDSC = NULL; free(pData); device->ExtraData = NULL; } static void DSoundStartCapture(ALCdevice *device) { DSoundCaptureData *pData = device->ExtraData; HRESULT hr; hr = IDirectSoundCaptureBuffer_Start(pData->DSCbuffer, DSCBSTART_LOOPING); if(FAILED(hr)) { ERR("start failed: 0x%08lx\n", hr); aluHandleDisconnect(device); } } static void DSoundStopCapture(ALCdevice *device) { DSoundCaptureData *pData = device->ExtraData; HRESULT hr; hr = IDirectSoundCaptureBuffer_Stop(pData->DSCbuffer); if(FAILED(hr)) { ERR("stop failed: 0x%08lx\n", hr); aluHandleDisconnect(device); } } static ALCenum DSoundCaptureSamples(ALCdevice *pDevice, ALCvoid *pBuffer, ALCuint lSamples) { DSoundCaptureData *pData = pDevice->ExtraData; ReadRingBuffer(pData->pRing, pBuffer, lSamples); return ALC_NO_ERROR; } static ALCuint DSoundAvailableSamples(ALCdevice *pDevice) { DSoundCaptureData *pData = pDevice->ExtraData; DWORD dwRead, dwCursor, dwBufferBytes, dwNumBytes; void *pvAudio1, *pvAudio2; DWORD dwAudioBytes1, dwAudioBytes2; DWORD FrameSize; HRESULT hr; if(!pDevice->Connected) goto done; FrameSize = FrameSizeFromDevFmt(pDevice->FmtChans, pDevice->FmtType); dwBufferBytes = pData->dwBufferBytes; dwCursor = pData->dwCursor; hr = IDirectSoundCaptureBuffer_GetCurrentPosition(pData->DSCbuffer, NULL, &dwRead); if(SUCCEEDED(hr)) { dwNumBytes = (dwBufferBytes + dwRead - dwCursor) % dwBufferBytes; if(dwNumBytes == 0) goto done; hr = IDirectSoundCaptureBuffer_Lock(pData->DSCbuffer, dwCursor, dwNumBytes, &pvAudio1, &dwAudioBytes1, &pvAudio2, &dwAudioBytes2, 0); } if(SUCCEEDED(hr)) { WriteRingBuffer(pData->pRing, pvAudio1, dwAudioBytes1/FrameSize); if(pvAudio2 != NULL) WriteRingBuffer(pData->pRing, pvAudio2, dwAudioBytes2/FrameSize); hr = IDirectSoundCaptureBuffer_Unlock(pData->DSCbuffer, pvAudio1, dwAudioBytes1, pvAudio2, dwAudioBytes2); pData->dwCursor = (dwCursor + dwAudioBytes1 + dwAudioBytes2) % dwBufferBytes; } if(FAILED(hr)) { ERR("update failed: 0x%08lx\n", hr); aluHandleDisconnect(pDevice); } done: return RingBufferSize(pData->pRing); } static const BackendFuncs DSoundFuncs = { DSoundOpenPlayback, DSoundClosePlayback, DSoundResetPlayback, DSoundStopPlayback, DSoundOpenCapture, DSoundCloseCapture, DSoundStartCapture, DSoundStopCapture, DSoundCaptureSamples, DSoundAvailableSamples }; 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 DEVICE_PROBE: AppendDeviceList(dsDevice); break; 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++) AppendAllDeviceList(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; } }