/** * 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 "alu.h" #include "threads.h" #include "compat.h" #include "alstring.h" #ifndef DSSPEAKER_5POINT1 # define DSSPEAKER_5POINT1 0x00000006 #endif #ifndef DSSPEAKER_7POINT1 # define DSSPEAKER_7POINT1 0x00000007 #endif #ifndef DSSPEAKER_7POINT1_SURROUND # define DSSPEAKER_7POINT1_SURROUND 0x00000008 #endif #ifndef DSSPEAKER_5POINT1_SURROUND # define DSSPEAKER_5POINT1_SURROUND 0x00000009 #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 *pDirectSoundEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext); static HRESULT (WINAPI *pDirectSoundCaptureCreate)(LPCGUID pcGuidDevice, IDirectSoundCapture **ppDSC, IUnknown *pUnkOuter); static HRESULT (WINAPI *pDirectSoundCaptureEnumerateW)(LPDSENUMCALLBACKW pDSEnumCallback, void *pContext); #define DirectSoundCreate pDirectSoundCreate #define DirectSoundEnumerateW pDirectSoundEnumerateW #define DirectSoundCaptureCreate pDirectSoundCaptureCreate #define DirectSoundCaptureEnumerateW pDirectSoundCaptureEnumerateW typedef struct { // DirectSound Playback Device IDirectSound *DS; IDirectSoundBuffer *PrimaryBuffer; IDirectSoundBuffer *Buffer; IDirectSoundNotify *Notifies; HANDLE NotifyEvent; volatile int killNow; althread_t thread; } DSoundPlaybackData; typedef struct { // DirectSound Capture Device IDirectSoundCapture *DSC; IDirectSoundCaptureBuffer *DSCbuffer; DWORD BufferBytes; DWORD Cursor; RingBuffer *Ring; } DSoundCaptureData; typedef struct { al_string 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(DirectSoundEnumerateW); LOAD_FUNC(DirectSoundCaptureCreate); LOAD_FUNC(DirectSoundCaptureEnumerateW); #undef LOAD_FUNC } return ALC_TRUE; } static BOOL CALLBACK DSoundEnumPlaybackDevices(LPGUID guid, LPCWSTR desc, LPCWSTR UNUSED(drvname), LPVOID UNUSED(data)) { LPOLESTR guidstr = NULL; al_string dname; HRESULT hr; void *temp; int count; ALuint i; if(!guid) return TRUE; AL_STRING_INIT(dname); count = 0; do { al_string_copy_wcstr(&dname, desc); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); al_string_append_cstr(&dname, str); } count++; for(i = 0;i < NumPlaybackDevices;i++) { if(al_string_cmp(dname, PlaybackDeviceList[i].name) == 0) break; } } while(i != NumPlaybackDevices); hr = StringFromCLSID(guid, &guidstr); if(SUCCEEDED(hr)) { TRACE("Got device \"%s\", GUID \"%ls\"\n", al_string_get_cstr(dname), guidstr); CoTaskMemFree(guidstr); } temp = realloc(PlaybackDeviceList, sizeof(DevMap) * (NumPlaybackDevices+1)); if(!temp) AL_STRING_DEINIT(dname); else { PlaybackDeviceList = temp; PlaybackDeviceList[NumPlaybackDevices].name = dname; PlaybackDeviceList[NumPlaybackDevices].guid = *guid; NumPlaybackDevices++; } return TRUE; } static BOOL CALLBACK DSoundEnumCaptureDevices(LPGUID guid, LPCWSTR desc, LPCWSTR UNUSED(drvname), LPVOID UNUSED(data)) { LPOLESTR guidstr = NULL; al_string dname; HRESULT hr; void *temp; int count; ALuint i; if(!guid) return TRUE; AL_STRING_INIT(dname); count = 0; do { al_string_copy_wcstr(&dname, desc); if(count != 0) { char str[64]; snprintf(str, sizeof(str), " #%d", count+1); al_string_append_cstr(&dname, str); } count++; for(i = 0;i < NumCaptureDevices;i++) { if(al_string_cmp(dname, CaptureDeviceList[i].name) == 0) break; } } while(i != NumCaptureDevices); hr = StringFromCLSID(guid, &guidstr); if(SUCCEEDED(hr)) { TRACE("Got device \"%s\", GUID \"%ls\"\n", al_string_get_cstr(dname), guidstr); CoTaskMemFree(guidstr); } temp = realloc(CaptureDeviceList, sizeof(DevMap) * (NumCaptureDevices+1)); if(!temp) AL_STRING_DEINIT(dname); else { CaptureDeviceList = temp; CaptureDeviceList[NumCaptureDevices].name = dname; CaptureDeviceList[NumCaptureDevices].guid = *guid; NumCaptureDevices++; } return TRUE; } FORCE_ALIGN 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(); SetThreadName(MIXER_THREAD_NAME); memset(&DSBCaps, 0, sizeof(DSBCaps)); DSBCaps.dwSize = sizeof(DSBCaps); err = IDirectSoundBuffer_GetCaps(data->Buffer, &DSBCaps); if(FAILED(err)) { ERR("Failed to get buffer caps: 0x%lx\n", err); ALCdevice_Lock(Device); aluHandleDisconnect(Device); ALCdevice_Unlock(Device); return 1; } FrameSize = FrameSizeFromDevFmt(Device->FmtChans, Device->FmtType); FragSize = Device->UpdateSize * FrameSize; IDirectSoundBuffer_GetCurrentPosition(data->Buffer, &LastCursor, NULL); while(!data->killNow) { // Get current play cursor IDirectSoundBuffer_GetCurrentPosition(data->Buffer, &PlayCursor, NULL); avail = (PlayCursor-LastCursor+DSBCaps.dwBufferBytes) % DSBCaps.dwBufferBytes; if(avail < FragSize) { if(!Playing) { err = IDirectSoundBuffer_Play(data->Buffer, 0, 0, DSBPLAY_LOOPING); if(FAILED(err)) { ERR("Failed to play buffer: 0x%lx\n", err); ALCdevice_Lock(Device); aluHandleDisconnect(Device); ALCdevice_Unlock(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->Buffer, 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->Buffer); if(SUCCEEDED(err)) { Playing = FALSE; LastCursor = 0; err = IDirectSoundBuffer_Lock(data->Buffer, 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->Buffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2); } else { ERR("Buffer lock error: %#lx\n", err); ALCdevice_Lock(Device); aluHandleDisconnect(Device); ALCdevice_Unlock(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, hrcom; if(!PlaybackDeviceList) { /* Initialize COM to prevent name truncation */ hrcom = CoInitialize(NULL); hr = DirectSoundEnumerateW(DSoundEnumPlaybackDevices, NULL); if(FAILED(hr)) ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr); if(SUCCEEDED(hrcom)) CoUninitialize(); } if(!deviceName && NumPlaybackDevices > 0) { deviceName = al_string_get_cstr(PlaybackDeviceList[0].name); guid = &PlaybackDeviceList[0].guid; } else { ALuint i; for(i = 0;i < NumPlaybackDevices;i++) { if(al_string_cmp_cstr(PlaybackDeviceList[i].name, deviceName) == 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; } al_string_copy_cstr(&device->DeviceName, deviceName); device->ExtraData = data; return ALC_NO_ERROR; } static void DSoundClosePlayback(ALCdevice *device) { DSoundPlaybackData *data = device->ExtraData; if(data->Notifies) IDirectSoundNotify_Release(data->Notifies); data->Notifies = NULL; if(data->Buffer) IDirectSoundBuffer_Release(data->Buffer); data->Buffer = NULL; if(data->PrimaryBuffer != NULL) IDirectSoundBuffer_Release(data->PrimaryBuffer); data->PrimaryBuffer = 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->Notifies) IDirectSoundNotify_Release(data->Notifies); data->Notifies = NULL; if(data->Buffer) IDirectSoundBuffer_Release(data->Buffer); data->Buffer = NULL; if(data->PrimaryBuffer != NULL) IDirectSoundBuffer_Release(data->PrimaryBuffer); data->PrimaryBuffer = 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 || speakers == DSSPEAKER_5POINT1_SURROUND) device->FmtChans = DevFmtX51; else if(speakers == DSSPEAKER_7POINT1 || speakers == DSSPEAKER_7POINT1_SURROUND) 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->PrimaryBuffer) IDirectSoundBuffer_Release(data->PrimaryBuffer); data->PrimaryBuffer = NULL; } else { if(SUCCEEDED(hr) && !data->PrimaryBuffer) { memset(&DSBDescription,0,sizeof(DSBUFFERDESC)); DSBDescription.dwSize=sizeof(DSBUFFERDESC); DSBDescription.dwFlags=DSBCAPS_PRIMARYBUFFER; hr = IDirectSound_CreateSoundBuffer(data->DS, &DSBDescription, &data->PrimaryBuffer, NULL); } if(SUCCEEDED(hr)) hr = IDirectSoundBuffer_SetFormat(data->PrimaryBuffer,&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->Buffer, NULL); if(FAILED(hr) && device->FmtType == DevFmtFloat) { device->FmtType = DevFmtShort; goto retry_open; } } if(SUCCEEDED(hr)) { hr = IDirectSoundBuffer_QueryInterface(data->Buffer, &IID_IDirectSoundNotify, (LPVOID *)&data->Notifies); 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->Notifies, device->NumUpdates, notifies) != DS_OK) hr = E_FAIL; } } if(FAILED(hr)) { if(data->Notifies != NULL) IDirectSoundNotify_Release(data->Notifies); data->Notifies = NULL; if(data->Buffer != NULL) IDirectSoundBuffer_Release(data->Buffer); data->Buffer = NULL; if(data->PrimaryBuffer != NULL) IDirectSoundBuffer_Release(data->PrimaryBuffer); data->PrimaryBuffer = NULL; return ALC_FALSE; } ResetEvent(data->NotifyEvent); SetDefaultWFXChannelOrder(device); return ALC_TRUE; } static ALCboolean DSoundStartPlayback(ALCdevice *device) { DSoundPlaybackData *data = (DSoundPlaybackData*)device->ExtraData; if(!StartThread(&data->thread, DSoundPlaybackProc, device)) 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->Buffer); } 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 = DirectSoundCaptureEnumerateW(DSoundEnumCaptureDevices, NULL); if(FAILED(hr)) ERR("Error enumerating DirectSound devices (0x%lx)!\n", hr); if(SUCCEEDED(hrcom)) CoUninitialize(); } if(!deviceName && NumCaptureDevices > 0) { deviceName = al_string_get_cstr(CaptureDeviceList[0].name); guid = &CaptureDeviceList[0].guid; } else { ALuint i; for(i = 0;i < NumCaptureDevices;i++) { if(al_string_cmp_cstr(CaptureDeviceList[i].name, deviceName) == 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); al_string_copy_cstr(&device->DeviceName, 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_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) AL_STRING_DEINIT(PlaybackDeviceList[i].name); free(PlaybackDeviceList); PlaybackDeviceList = NULL; NumPlaybackDevices = 0; for(i = 0;i < NumCaptureDevices;++i) AL_STRING_DEINIT(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; /* Initialize COM to prevent name truncation */ hrcom = CoInitialize(NULL); switch(type) { case ALL_DEVICE_PROBE: for(i = 0;i < NumPlaybackDevices;++i) AL_STRING_DEINIT(PlaybackDeviceList[i].name); free(PlaybackDeviceList); PlaybackDeviceList = NULL; NumPlaybackDevices = 0; hr = DirectSoundEnumerateW(DSoundEnumPlaybackDevices, NULL); if(FAILED(hr)) ERR("Error enumerating DirectSound playback devices (0x%lx)!\n", hr); else { for(i = 0;i < NumPlaybackDevices;i++) AppendAllDevicesList(al_string_get_cstr(PlaybackDeviceList[i].name)); } break; case CAPTURE_DEVICE_PROBE: for(i = 0;i < NumCaptureDevices;++i) AL_STRING_DEINIT(CaptureDeviceList[i].name); free(CaptureDeviceList); CaptureDeviceList = NULL; NumCaptureDevices = 0; hr = DirectSoundCaptureEnumerateW(DSoundEnumCaptureDevices, NULL); if(FAILED(hr)) ERR("Error enumerating DirectSound capture devices (0x%lx)!\n", hr); else { for(i = 0;i < NumCaptureDevices;i++) AppendCaptureDeviceList(al_string_get_cstr(CaptureDeviceList[i].name)); } break; } if(SUCCEEDED(hrcom)) CoUninitialize(); }