Mercurial > fife-parpg
view ext/openal-soft/Alc/dsound.c @ 121:ae3b8139c7c7
* Applying settings patch by greyghost
* See: #274
author | mvbarracuda@33b003aa-7bff-0310-803a-e67f0ece8222 |
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date | Tue, 05 Aug 2008 14:44:15 +0000 |
parents | 4a0efb7baf70 |
children |
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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" #define INITGUID #include <stdlib.h> #include <stdio.h> #include <memory.h> #include <dsound.h> #include <mmreg.h> #include "alMain.h" #include "AL/al.h" #include "AL/alc.h" #ifndef DSSPEAKER_7POINT1 #define DSSPEAKER_7POINT1 7 #endif DEFINE_GUID(KSDATAFORMAT_SUBTYPE_PCM, 0x00000001, 0x0000, 0x0010, 0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71); // Since DSound doesn't report the fragment size, just assume 4 fragments #define DS_FRAGS 4 typedef struct { // DirectSound Playback Device LPDIRECTSOUND lpDS; LPDIRECTSOUNDBUFFER DSpbuffer; LPDIRECTSOUNDBUFFER DSsbuffer; int killNow; ALvoid *thread; } DSoundData; typedef struct { ALCchar *name; GUID guid; } DevMap; static DevMap DeviceList[16]; static ALuint DSoundProc(ALvoid *ptr) { ALCdevice *pDevice = (ALCdevice*)ptr; DSoundData *pData = (DSoundData*)pDevice->ExtraData; DWORD LastCursor = 0; DWORD PlayCursor; VOID *WritePtr1, *WritePtr2; DWORD WriteCnt1, WriteCnt2; DWORD BufferSize; DWORD avail; HRESULT err; BufferSize = pDevice->UpdateSize * DS_FRAGS * aluBytesFromFormat(pDevice->Format) * aluChannelsFromFormat(pDevice->Format); while(!pData->killNow) { // Get current play and write cursors IDirectSoundBuffer_GetCurrentPosition(pData->DSsbuffer, &PlayCursor, NULL); avail = (PlayCursor-LastCursor+BufferSize) % BufferSize; if(avail == 0) { Sleep(1); continue; } // 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, play and lock if(err == DSERR_BUFFERLOST) { err = IDirectSoundBuffer_Restore(pData->DSsbuffer); if(SUCCEEDED(err)) err = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING); if(SUCCEEDED(err)) err = IDirectSoundBuffer_Lock(pData->DSsbuffer, LastCursor, avail, &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 SuspendContext(NULL); aluMixData(pDevice->Context, WritePtr1, WriteCnt1, pDevice->Format); aluMixData(pDevice->Context, WritePtr2, WriteCnt2, pDevice->Format); ProcessContext(NULL); // Unlock output buffer only when successfully locked IDirectSoundBuffer_Unlock(pData->DSsbuffer, WritePtr1, WriteCnt1, WritePtr2, WriteCnt2); } else AL_PRINT("Buffer lock error: %#lx\n", err); // Update old write cursor location LastCursor += WriteCnt1+WriteCnt2; LastCursor %= BufferSize; } return 0; } static ALCboolean DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName) { DSBUFFERDESC DSBDescription; DSoundData *pData = NULL; WAVEFORMATEXTENSIBLE OutputType; DWORD frameSize = 0; LPGUID guid = NULL; DWORD speakers; HRESULT hr; if(deviceName) { int i; for(i = 0;DeviceList[i].name;i++) { if(strcmp(deviceName, DeviceList[i].name) == 0) { device->szDeviceName = DeviceList[i].name; if(i > 0) guid = &DeviceList[i].guid; break; } } if(!DeviceList[i].name) return ALC_FALSE; } else device->szDeviceName = DeviceList[0].name; memset(&OutputType, 0, sizeof(OutputType)); //Initialise requested device pData = calloc(1, sizeof(DSoundData)); if(!pData) { SetALCError(ALC_OUT_OF_MEMORY); return ALC_FALSE; } //DirectSound Init code hr = DirectSoundCreate(guid, &pData->lpDS, NULL); if(SUCCEEDED(hr)) hr = IDirectSound_SetCooperativeLevel(pData->lpDS, GetForegroundWindow(), DSSCL_PRIORITY); if(SUCCEEDED(hr)) hr = IDirectSound_GetSpeakerConfig(pData->lpDS, &speakers); if(SUCCEEDED(hr)) { speakers = DSSPEAKER_CONFIG(speakers); if(speakers == DSSPEAKER_MONO) { if(aluBytesFromFormat(device->Format) == 1) device->Format = AL_FORMAT_MONO8; else device->Format = AL_FORMAT_MONO16; } else if(speakers == DSSPEAKER_STEREO) { if(aluBytesFromFormat(device->Format) == 1) device->Format = AL_FORMAT_STEREO8; else device->Format = AL_FORMAT_STEREO16; } else if(speakers == DSSPEAKER_QUAD) { if(aluBytesFromFormat(device->Format) == 1) device->Format = AL_FORMAT_QUAD8; else device->Format = AL_FORMAT_QUAD16; OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; } else if(speakers == DSSPEAKER_5POINT1) { if(aluBytesFromFormat(device->Format) == 1) device->Format = AL_FORMAT_51CHN8; else device->Format = AL_FORMAT_51CHN16; OutputType.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT | SPEAKER_FRONT_CENTER | SPEAKER_LOW_FREQUENCY | SPEAKER_BACK_LEFT | SPEAKER_BACK_RIGHT; } else if(speakers == DSSPEAKER_7POINT1) { if(aluBytesFromFormat(device->Format) == 1) device->Format = AL_FORMAT_71CHN8; else device->Format = AL_FORMAT_71CHN16; 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; } frameSize = aluBytesFromFormat(device->Format) * aluChannelsFromFormat(device->Format); OutputType.Format.wFormatTag = WAVE_FORMAT_PCM; OutputType.Format.nChannels = aluChannelsFromFormat(device->Format); OutputType.Format.wBitsPerSample = aluBytesFromFormat(device->Format) * 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; device->UpdateSize /= DS_FRAGS; } if(OutputType.Format.nChannels > 2) { OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE; OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample; OutputType.Format.cbSize = 22; 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); } if(SUCCEEDED(hr)) { memset(&DSBDescription,0,sizeof(DSBUFFERDESC)); DSBDescription.dwSize=sizeof(DSBUFFERDESC); DSBDescription.dwFlags=DSBCAPS_GLOBALFOCUS|DSBCAPS_GETCURRENTPOSITION2; DSBDescription.dwBufferBytes=device->UpdateSize * DS_FRAGS * frameSize; DSBDescription.lpwfxFormat=&OutputType.Format; hr = IDirectSound_CreateSoundBuffer(pData->lpDS, &DSBDescription, &pData->DSsbuffer, NULL); } if(SUCCEEDED(hr)) hr = IDirectSoundBuffer_Play(pData->DSsbuffer, 0, 0, DSBPLAY_LOOPING); device->ExtraData = pData; pData->thread = StartThread(DSoundProc, device); if(!pData->thread) hr = E_FAIL; if(FAILED(hr)) { if (pData->DSsbuffer) IDirectSoundBuffer_Release(pData->DSsbuffer); if (pData->DSpbuffer) IDirectSoundBuffer_Release(pData->DSpbuffer); if (pData->lpDS) IDirectSound_Release(pData->lpDS); free(pData); return ALC_FALSE; } return ALC_TRUE; } static void DSoundClosePlayback(ALCdevice *device) { DSoundData *pData = device->ExtraData; pData->killNow = 1; StopThread(pData->thread); IDirectSoundBuffer_Release(pData->DSsbuffer); if (pData->DSpbuffer) IDirectSoundBuffer_Release(pData->DSpbuffer); IDirectSound_Release(pData->lpDS); free(pData); device->ExtraData = NULL; } static ALCboolean DSoundOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName, ALCuint frequency, ALCenum format, ALCsizei SampleSize) { (void)pDevice; (void)deviceName; (void)frequency; (void)format; (void)SampleSize; 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; } BackendFuncs DSoundFuncs = { DSoundOpenPlayback, DSoundClosePlayback, DSoundOpenCapture, DSoundCloseCapture, DSoundStartCapture, DSoundStopCapture, DSoundCaptureSamples, DSoundAvailableSamples }; static BOOL CALLBACK DSoundEnumDevices(LPGUID guid, LPCSTR desc, LPCSTR drvname, LPVOID data) { size_t *iter = data; (void)drvname; if(guid) { char str[128]; snprintf(str, sizeof(str), "DirectSound Software on %s", desc); DeviceList[*iter].name = AppendAllDeviceList(str); DeviceList[*iter].guid = *guid; (*iter)++; } else DeviceList[0].name = AppendDeviceList("DirectSound Software"); return TRUE; } void alcDSoundInit(BackendFuncs *FuncList) { size_t iter = 1; HRESULT hr; *FuncList = DSoundFuncs; hr = DirectSoundEnumerate(DSoundEnumDevices, &iter); if(FAILED(hr)) AL_PRINT("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr); }