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AuroraOpenALSoft/Alc/dsound.c
Chris Robinson b444dea63b Fix compilation for Windows
2009-08-28 06:33:59 -07:00

552 lines
16 KiB
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
*/
#include "config.h"
#define _WIN32_WINNT 0x0500
#define INITGUID
#include <stdlib.h>
#include <stdio.h>
#include <memory.h>
#include <dsound.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, LPDIRECTSOUND *ppDS, LPUNKNOWN pUnkOuter);
static HRESULT (WINAPI *pDirectSoundEnumerateA)(LPDSENUMCALLBACKA pDSEnumCallback, LPVOID pContext);
// Since DSound doesn't report the fragment size, emulate it
static int num_frags;
typedef struct {
// DirectSound Playback Device
LPDIRECTSOUND lpDS;
LPDIRECTSOUNDBUFFER DSpbuffer;
LPDIRECTSOUNDBUFFER DSsbuffer;
volatile int killNow;
ALvoid *thread;
} DSoundData;
typedef struct {
ALCchar *name;
GUID guid;
} DevMap;
static const ALCchar dsDevice[] = "DirectSound Software";
static DevMap *DeviceList;
static ALuint NumDevices;
static ALuint DSoundProc(ALvoid *ptr)
{
ALCdevice *pDevice = (ALCdevice*)ptr;
DSoundData *pData = (DSoundData*)pDevice->ExtraData;
DSBCAPS DSBCaps;
DWORD LastCursor = 0;
DWORD PlayCursor;
VOID *WritePtr1, *WritePtr2;
DWORD WriteCnt1, WriteCnt2;
DWORD FragSize;
DWORD avail;
HRESULT err;
memset(&DSBCaps, 0, sizeof(DSBCaps));
DSBCaps.dwSize = sizeof(DSBCaps);
err = IDirectSoundBuffer_GetCaps(pData->DSsbuffer, &DSBCaps);
if(FAILED(err))
{
AL_PRINT("Failed to get buffer caps: 0x%lx\n", err);
return 1;
}
FragSize = DSBCaps.dwBufferBytes / num_frags;
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;
if(avail < FragSize)
{
Sleep(1);
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, 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 %= DSBCaps.dwBufferBytes;
}
return 0;
}
static ALCboolean DSoundOpenPlayback(ALCdevice *device, const ALCchar *deviceName)
{
DSoundData *pData = NULL;
LPGUID guid = NULL;
HRESULT hr;
if(ds_handle == NULL)
return ALC_FALSE;
if(!deviceName)
deviceName = dsDevice;
else if(strcmp(deviceName, dsDevice) != 0)
{
ALuint i;
for(i = 0;i < NumDevices;i++)
{
if(strcmp(deviceName, DeviceList[i].name) == 0)
{
guid = &DeviceList[i].guid;
break;
}
}
if(i == NumDevices)
return ALC_FALSE;
}
//Initialise requested device
pData = calloc(1, sizeof(DSoundData));
if(!pData)
{
SetALCError(ALC_OUT_OF_MEMORY);
return ALC_FALSE;
}
//DirectSound Init code
hr = pDirectSoundCreate(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);
free(pData);
return ALC_FALSE;
}
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 DSoundStartContext(ALCdevice *device, ALCcontext *context)
{
DSoundData *pData = (DSoundData*)device->ExtraData;
DSBUFFERDESC DSBDescription;
WAVEFORMATEXTENSIBLE OutputType;
DWORD frameSize = 0;
ALenum format = 0;
DWORD speakers;
HRESULT hr;
(void)context;
memset(&OutputType, 0, sizeof(OutputType));
hr = IDirectSound_GetSpeakerConfig(pData->lpDS, &speakers);
if(SUCCEEDED(hr) && *(GetConfigValue(NULL, "format", "")) != 0)
{
if(device->Format == AL_FORMAT_MONO8 || device->Format == AL_FORMAT_MONO16)
speakers = DSSPEAKER_COMBINED(DSSPEAKER_MONO, 0);
else if(device->Format == AL_FORMAT_STEREO8 || device->Format == AL_FORMAT_STEREO16)
speakers = DSSPEAKER_COMBINED(DSSPEAKER_STEREO, 0);
else if(device->Format == AL_FORMAT_QUAD8 || device->Format == AL_FORMAT_QUAD16)
speakers = DSSPEAKER_COMBINED(DSSPEAKER_QUAD, 0);
else if(device->Format == AL_FORMAT_51CHN8 || device->Format == AL_FORMAT_51CHN16)
speakers = DSSPEAKER_COMBINED(DSSPEAKER_5POINT1, 0);
else if(device->Format == AL_FORMAT_71CHN8 || device->Format == AL_FORMAT_71CHN16)
speakers = DSSPEAKER_COMBINED(DSSPEAKER_7POINT1, 0);
}
if(SUCCEEDED(hr))
{
speakers = DSSPEAKER_CONFIG(speakers);
if(speakers == DSSPEAKER_MONO)
{
if(aluBytesFromFormat(device->Format) == 1)
format = AL_FORMAT_MONO8;
else if(aluBytesFromFormat(device->Format) == 2)
format = AL_FORMAT_MONO16;
else if(aluBytesFromFormat(device->Format) == 4)
format = AL_FORMAT_MONO_FLOAT32;
}
else if(speakers == DSSPEAKER_STEREO)
{
if(aluBytesFromFormat(device->Format) == 1)
format = AL_FORMAT_STEREO8;
else if(aluBytesFromFormat(device->Format) == 2)
format = AL_FORMAT_STEREO16;
else if(aluBytesFromFormat(device->Format) == 4)
format = AL_FORMAT_STEREO_FLOAT32;
}
else if(speakers == DSSPEAKER_QUAD)
{
if(aluBytesFromFormat(device->Format) == 1)
format = AL_FORMAT_QUAD8;
else if(aluBytesFromFormat(device->Format) == 2)
format = AL_FORMAT_QUAD16;
else if(aluBytesFromFormat(device->Format) == 4)
format = AL_FORMAT_QUAD32;
OutputType.dwChannelMask = SPEAKER_FRONT_LEFT |
SPEAKER_FRONT_RIGHT |
SPEAKER_BACK_LEFT |
SPEAKER_BACK_RIGHT;
}
else if(speakers == DSSPEAKER_5POINT1)
{
if(aluBytesFromFormat(device->Format) == 1)
format = AL_FORMAT_51CHN8;
else if(aluBytesFromFormat(device->Format) == 2)
format = AL_FORMAT_51CHN16;
else if(aluBytesFromFormat(device->Format) == 4)
format = AL_FORMAT_51CHN32;
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)
format = AL_FORMAT_71CHN8;
else if(aluBytesFromFormat(device->Format) == 2)
format = AL_FORMAT_71CHN16;
else if(aluBytesFromFormat(device->Format) == 4)
format = AL_FORMAT_71CHN32;
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;
}
else
format = device->Format;
frameSize = aluBytesFromFormat(format) * aluChannelsFromFormat(format);
OutputType.Format.wFormatTag = WAVE_FORMAT_PCM;
OutputType.Format.nChannels = aluChannelsFromFormat(format);
OutputType.Format.wBitsPerSample = aluBytesFromFormat(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;
}
if(OutputType.Format.nChannels > 2 || OutputType.Format.wBitsPerSample > 16)
{
OutputType.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
OutputType.Samples.wValidBitsPerSample = OutputType.Format.wBitsPerSample;
OutputType.Format.cbSize = 22;
if(OutputType.Format.wBitsPerSample == 32)
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);
}
if(SUCCEEDED(hr))
{
memset(&DSBDescription,0,sizeof(DSBUFFERDESC));
DSBDescription.dwSize=sizeof(DSBUFFERDESC);
DSBDescription.dwFlags=DSBCAPS_GLOBALFOCUS|DSBCAPS_GETCURRENTPOSITION2;
DSBDescription.dwBufferBytes=(device->BufferSize/num_frags) * num_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);
if(SUCCEEDED(hr))
{
device->ExtraData = pData;
pData->thread = StartThread(DSoundProc, device);
if(!pData->thread)
hr = E_FAIL;
}
if(FAILED(hr))
{
if (pData->DSsbuffer)
IDirectSoundBuffer_Release(pData->DSsbuffer);
pData->DSsbuffer = NULL;
if (pData->DSpbuffer)
IDirectSoundBuffer_Release(pData->DSpbuffer);
pData->DSpbuffer = NULL;
return ALC_FALSE;
}
device->Format = format;
device->UpdateSize = device->BufferSize/num_frags;
return ALC_TRUE;
}
static void DSoundStopContext(ALCdevice *device, ALCcontext *context)
{
DSoundData *pData = device->ExtraData;
(void)context;
if(!pData->thread)
return;
pData->killNow = 1;
StopThread(pData->thread);
pData->thread = NULL;
IDirectSoundBuffer_Release(pData->DSsbuffer);
pData->DSsbuffer = NULL;
if (pData->DSpbuffer)
IDirectSoundBuffer_Release(pData->DSpbuffer);
pData->DSpbuffer = NULL;
}
static ALCboolean DSoundOpenCapture(ALCdevice *pDevice, const ALCchar *deviceName)
{
(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;
}
BackendFuncs DSoundFuncs = {
DSoundOpenPlayback,
DSoundClosePlayback,
DSoundStartContext,
DSoundStopContext,
DSoundOpenCapture,
DSoundCloseCapture,
DSoundStartCapture,
DSoundStopCapture,
DSoundCaptureSamples,
DSoundAvailableSamples
};
static BOOL CALLBACK DSoundEnumDevices(LPGUID guid, LPCSTR desc, LPCSTR drvname, LPVOID data)
{
(void)data;
(void)drvname;
if(guid)
{
char str[128];
void *temp;
temp = realloc(DeviceList, sizeof(DevMap) * (NumDevices+1));
if(temp)
{
DeviceList = temp;
snprintf(str, sizeof(str), "DirectSound Software on %s", desc);
AppendAllDeviceList(str);
DeviceList[NumDevices].name = strdup(str);
DeviceList[NumDevices].guid = *guid;
NumDevices++;
}
}
return TRUE;
}
void alcDSoundInit(BackendFuncs *FuncList)
{
*FuncList = DSoundFuncs;
#ifdef _WIN32
ds_handle = LoadLibraryA("dsound.dll");
if(ds_handle == NULL)
{
AL_PRINT("Failed to load dsound.dll\n");
return;
}
#define LOAD_FUNC(f) do { \
p##f = (void*)GetProcAddress((HMODULE)ds_handle, #f); \
if(p##f == NULL) \
{ \
FreeLibrary(ds_handle); \
ds_handle = NULL; \
AL_PRINT("Could not load %s from dsound.dll\n", #f); \
return; \
} \
} while(0)
#else
ds_handle = (void*)0xDEADBEEF;
#define LOAD_FUNC(f) p##f = f
#endif
LOAD_FUNC(DirectSoundCreate);
LOAD_FUNC(DirectSoundEnumerateA);
#undef LOAD_FUNC
num_frags = GetConfigValueInt("dsound", "periods", 4);
if(num_frags < 2) num_frags = 2;
}
void alcDSoundDeinit(void)
{
ALuint i;
for(i = 0;i < NumDevices;++i)
free(DeviceList[i].name);
free(DeviceList);
DeviceList = NULL;
NumDevices = 0;
#ifdef _WIN32
if(ds_handle)
FreeLibrary(ds_handle);
ds_handle = NULL;
#endif
}
void alcDSoundProbe(int type)
{
if(!ds_handle)
return;
if(type == DEVICE_PROBE)
AppendDeviceList(dsDevice);
else if(type == ALL_DEVICE_PROBE)
{
HRESULT hr;
ALuint i;
for(i = 0;i < NumDevices;++i)
free(DeviceList[i].name);
free(DeviceList);
DeviceList = NULL;
NumDevices = 0;
hr = pDirectSoundEnumerateA(DSoundEnumDevices, NULL);
if(FAILED(hr))
AL_PRINT("Error enumerating DirectSound devices (%#x)!\n", (unsigned int)hr);
}
}
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