zdoom/raze-gles
0
0
Fork 0
mirror of https://github.com/ZDoom/raze-gles.git synced 2024-12-27 20:20:40 +00:00
Code Issues Projects Releases Packages Wiki Activity
raze-gles/source/common/sound/backend/oalsound.cpp
Christoph Oelckers 4d5755ca67 - update of the sound engine for unlimited channels per source. 
Blood needs this.
2019-12-17 00:29:38 +01:00

2370 lines
64 KiB
C++
Raw Blame History

/*
** oalsound.cpp
** System interface for sound; uses OpenAL
**
**---------------------------------------------------------------------------
** Copyright 2008-2010 Chris Robinson
** All rights reserved.
**
** Redistribution and use in source and binary forms, with or without
** modification, are permitted provided that the following conditions
** are met:
**
** 1. Redistributions of source code must retain the above copyright
** notice, this list of conditions and the following disclaimer.
** 2. Redistributions in binary form must reproduce the above copyright
** notice, this list of conditions and the following disclaimer in the
** documentation and/or other materials provided with the distribution.
** 3. The name of the author may not be used to endorse or promote products
** derived from this software without specific prior written permission.
**
** THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
** IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
** OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
** IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
** INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
** NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
** DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
** THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
** (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
** THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
**---------------------------------------------------------------------------
**
*/
#include <functional>
#include <chrono>
#include "templates.h"
#include "oalsound.h"
#include "c_dispatch.h"
#include "v_text.h"
#include "i_module.h"
#include "cmdlib.h"
#include "c_cvars.h"
#include "printf.h"
#include "zmusic/sounddecoder.h"
#include "filereadermusicinterface.h"
const char *GetSampleTypeName(SampleType type);
const char *GetChannelConfigName(ChannelConfig chan);
FModule OpenALModule{"OpenAL"};
#include "oalload.h"
CUSTOM_CVAR(Int, snd_channels, 128, CVAR_ARCHIVE | CVAR_GLOBALCONFIG) // number of channels available
{
if (self < 64) self = 64;
}
CVAR(Bool, snd_waterreverb, true, CVAR_ARCHIVE | CVAR_GLOBALCONFIG)
CVAR (String, snd_aldevice, "Default", CVAR_ARCHIVE|CVAR_GLOBALCONFIG)
CVAR (Bool, snd_efx, true, CVAR_ARCHIVE|CVAR_GLOBALCONFIG)
CVAR (String, snd_alresampler, "Default", CVAR_ARCHIVE|CVAR_GLOBALCONFIG)
#ifdef _WIN32
#define OPENALLIB "openal32.dll"
#elif defined(__OpenBSD__)
#define OPENALLIB "libopenal.so"
#else
#define OPENALLIB "libopenal.so.1"
#endif
#ifdef __APPLE__
// User's library (like OpenAL Soft installed manually or via Homebrew) has precedence
// over Apple's OpenAL framework which lacks several important features
#define OPENALLIB1 "libopenal.1.dylib"
#define OPENALLIB2 "OpenAL.framework/OpenAL"
#else // !__APPLE__
#define OPENALLIB1 NicePath("$PROGDIR/" OPENALLIB)
#define OPENALLIB2 OPENALLIB
#endif
bool IsOpenALPresent()
{
#ifdef NO_OPENAL
return false;
#elif !defined DYN_OPENAL
return true;
#else
static bool cached_result = false;
static bool done = false;
if (!done)
{
done = true;
cached_result = OpenALModule.Load({OPENALLIB1, OPENALLIB2});
}
return cached_result;
#endif
}
ReverbContainer *ForcedEnvironment;
#ifndef NO_OPENAL
EXTERN_CVAR (Int, snd_samplerate)
EXTERN_CVAR (Bool, snd_pitched)
EXTERN_CVAR (Int, snd_hrtf)
#define MAKE_PTRID(x) ((void*)(uintptr_t)(x))
#define GET_PTRID(x) ((uint32_t)(uintptr_t)(x))
static ALenum checkALError(const char *fn, unsigned int ln)
{
ALenum err = alGetError();
if(err != AL_NO_ERROR)
{
if(strchr(fn, '/'))
fn = strrchr(fn, '/')+1;
else if(strchr(fn, '\\'))
fn = strrchr(fn, '\\')+1;
Printf(">>>>>>>>>>>> Received AL error %s (%#x), %s:%u\n", alGetString(err), err, fn, ln);
}
return err;
}
#define getALError() checkALError(__FILE__, __LINE__)
static ALCenum checkALCError(ALCdevice *device, const char *fn, unsigned int ln)
{
ALCenum err = alcGetError(device);
if(err != ALC_NO_ERROR)
{
if(strchr(fn, '/'))
fn = strrchr(fn, '/')+1;
else if(strchr(fn, '\\'))
fn = strrchr(fn, '\\')+1;
Printf(">>>>>>>>>>>> Received ALC error %s (%#x), %s:%u\n", alcGetString(device, err), err, fn, ln);
}
return err;
}
#define getALCError(d) checkALCError((d), __FILE__, __LINE__)
// Fallback methods for when AL_SOFT_deferred_updates isn't available. In most
// cases these don't actually do anything, except on some Creative drivers
// where they act as appropriate fallbacks.
static ALvoid AL_APIENTRY _wrap_DeferUpdatesSOFT(void)
{
alcSuspendContext(alcGetCurrentContext());
}
static ALvoid AL_APIENTRY _wrap_ProcessUpdatesSOFT(void)
{
alcProcessContext(alcGetCurrentContext());
}
class OpenALSoundStream : public SoundStream
{
OpenALSoundRenderer *Renderer;
SoundStreamCallback Callback;
void *UserData;
TArray<ALubyte> Data;
ALsizei SampleRate;
ALenum Format;
ALsizei FrameSize;
static const int BufferCount = 4;
ALuint Buffers[BufferCount];
ALuint Source;
std::atomic<bool> Playing;
bool Looping;
ALfloat Volume;
bool SetupSource()
{
/* Get a source, killing the farthest, lowest-priority sound if needed */
if(Renderer->FreeSfx.Size() == 0)
{
FSoundChan *lowest = Renderer->FindLowestChannel();
if(lowest) Renderer->ForceStopChannel(lowest);
if(Renderer->FreeSfx.Size() == 0)
return false;
}
Renderer->FreeSfx.Pop(Source);
/* Set the default properties for localized playback */
alSource3f(Source, AL_DIRECTION, 0.f, 0.f, 0.f);
alSource3f(Source, AL_VELOCITY, 0.f, 0.f, 0.f);
alSource3f(Source, AL_POSITION, 0.f, 0.f, 0.f);
alSourcef(Source, AL_MAX_GAIN, 1.f);
alSourcef(Source, AL_GAIN, 1.f);
alSourcef(Source, AL_PITCH, 1.f);
alSourcef(Source, AL_DOPPLER_FACTOR, 0.f);
alSourcef(Source, AL_ROLLOFF_FACTOR, 0.f);
alSourcef(Source, AL_SEC_OFFSET, 0.f);
alSourcei(Source, AL_SOURCE_RELATIVE, AL_TRUE);
alSourcei(Source, AL_LOOPING, AL_FALSE);
if(Renderer->EnvSlot)
{
alSourcef(Source, AL_ROOM_ROLLOFF_FACTOR, 0.f);
alSourcef(Source, AL_AIR_ABSORPTION_FACTOR, 0.f);
alSourcei(Source, AL_DIRECT_FILTER, AL_FILTER_NULL);
alSource3i(Source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
}
if(Renderer->AL.EXT_SOURCE_RADIUS)
alSourcef(Source, AL_SOURCE_RADIUS, 0.f);
if(Renderer->AL.SOFT_source_spatialize)
alSourcei(Source, AL_SOURCE_SPATIALIZE_SOFT, AL_AUTO_SOFT);
alGenBuffers(BufferCount, Buffers);
return (getALError() == AL_NO_ERROR);
}
public:
OpenALSoundStream(OpenALSoundRenderer *renderer)
: Renderer(renderer), Source(0), Playing(false), Looping(false), Volume(1.0f)
{
memset(Buffers, 0, sizeof(Buffers));
Renderer->AddStream(this);
}
virtual ~OpenALSoundStream()
{
Renderer->RemoveStream(this);
if(Source)
{
alSourceRewind(Source);
alSourcei(Source, AL_BUFFER, 0);
Renderer->FreeSfx.Push(Source);
Source = 0;
}
if(Buffers[0])
{
alDeleteBuffers(BufferCount, &Buffers[0]);
memset(Buffers, 0, sizeof(Buffers));
}
getALError();
}
virtual bool Play(bool loop, float vol)
{
SetVolume(vol);
if(Playing.load())
return true;
/* Clear the buffer queue, then fill and queue each buffer */
alSourcei(Source, AL_BUFFER, 0);
for(int i = 0;i < BufferCount;i++)
{
if(!Callback(this, &Data[0], Data.Size(), UserData))
{
if(i == 0)
return false;
break;
}
alBufferData(Buffers[i], Format, &Data[0], Data.Size(), SampleRate);
alSourceQueueBuffers(Source, 1, &Buffers[i]);
}
if(getALError() != AL_NO_ERROR)
return false;
alSourcePlay(Source);
if(getALError() != AL_NO_ERROR)
return false;
Playing.store(true);
return true;
}
virtual void Stop()
{
if(!Playing.load())
return;
std::unique_lock<std::mutex> lock(Renderer->StreamLock);
alSourceStop(Source);
alSourcei(Source, AL_BUFFER, 0);
getALError();
Playing.store(false);
}
virtual void SetVolume(float vol)
{
Volume = vol;
UpdateVolume();
}
void UpdateVolume()
{
alSourcef(Source, AL_GAIN, Renderer->MusicVolume*Volume);
getALError();
}
virtual bool SetPaused(bool pause)
{
if(pause)
alSourcePause(Source);
else
alSourcePlay(Source);
return (getALError()==AL_NO_ERROR);
}
virtual bool IsEnded()
{
return !Playing.load();
}
virtual FString GetStats()
{
FString stats;
size_t pos = 0, len = 0;
ALfloat volume;
ALint offset;
ALint processed;
ALint queued;
ALint state;
ALenum err;
std::unique_lock<std::mutex> lock(Renderer->StreamLock);
alGetSourcef(Source, AL_GAIN, &volume);
alGetSourcei(Source, AL_SAMPLE_OFFSET, &offset);
alGetSourcei(Source, AL_BUFFERS_PROCESSED, &processed);
alGetSourcei(Source, AL_BUFFERS_QUEUED, &queued);
alGetSourcei(Source, AL_SOURCE_STATE, &state);
if((err=alGetError()) != AL_NO_ERROR)
{
lock.unlock();
stats = "Error getting stats: ";
stats += alGetString(err);
return stats;
}
lock.unlock();
stats = (state == AL_INITIAL) ? "Buffering" : (state == AL_STOPPED) ? "Underrun" :
(state == AL_PLAYING || state == AL_PAUSED) ? "Ready" : "Unknown state";
if(state == AL_PAUSED)
stats += ", paused";
if(state == AL_PLAYING)
stats += ", playing";
stats.AppendFormat(", %uHz", SampleRate);
if(!Playing)
stats += " XX";
return stats;
}
bool Process()
{
if(!Playing.load())
return false;
ALint state, processed;
alGetSourcei(Source, AL_SOURCE_STATE, &state);
alGetSourcei(Source, AL_BUFFERS_PROCESSED, &processed);
if(getALError() != AL_NO_ERROR)
{
Playing.store(false);
return false;
}
// For each processed buffer in the queue...
while(processed > 0)
{
ALuint bufid;
// Unqueue the oldest buffer, fill it with more data, and queue it
// on the end
alSourceUnqueueBuffers(Source, 1, &bufid);
processed--;
if(Callback(this, &Data[0], Data.Size(), UserData))
{
alBufferData(bufid, Format, &Data[0], Data.Size(), SampleRate);
alSourceQueueBuffers(Source, 1, &bufid);
}
}
// If the source is not playing or paused, and there are buffers queued,
// then there was an underrun. Restart the source.
bool ok = (getALError()==AL_NO_ERROR);
if(ok && state != AL_PLAYING && state != AL_PAUSED)
{
ALint queued = 0;
alGetSourcei(Source, AL_BUFFERS_QUEUED, &queued);
ok = (getALError() == AL_NO_ERROR) && (queued > 0);
if(ok)
{
alSourcePlay(Source);
ok = (getALError()==AL_NO_ERROR);
}
}
Playing.store(ok);
return ok;
}
bool Init(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata)
{
if(!SetupSource())
return false;
Callback = callback;
UserData = userdata;
SampleRate = samplerate;
Format = AL_NONE;
if((flags&Bits8)) /* Signed or unsigned? We assume unsigned 8-bit... */
{
if((flags&Mono)) Format = AL_FORMAT_MONO8;
else Format = AL_FORMAT_STEREO8;
}
else if((flags&Float))
{
if(alIsExtensionPresent("AL_EXT_FLOAT32"))
{
if((flags&Mono)) Format = AL_FORMAT_MONO_FLOAT32;
else Format = AL_FORMAT_STEREO_FLOAT32;
}
}
else if((flags&Bits32))
{
}
else
{
if((flags&Mono)) Format = AL_FORMAT_MONO16;
else Format = AL_FORMAT_STEREO16;
}
if(Format == AL_NONE)
{
Printf("Unsupported format: 0x%x\n", flags);
return false;
}
FrameSize = 1;
if((flags&Bits8))
FrameSize *= 1;
else if((flags&(Bits32|Float)))
FrameSize *= 4;
else
FrameSize *= 2;
if((flags&Mono))
FrameSize *= 1;
else
FrameSize *= 2;
buffbytes += FrameSize-1;
buffbytes -= buffbytes%FrameSize;
Data.Resize(buffbytes);
return true;
}
};
#define AREA_SOUND_RADIUS (32.f)
#define PITCH_MULT (0.7937005f) /* Approx. 4 semitones lower; what Nash suggested */
#define PITCH(pitch) (snd_pitched ? (pitch)/128.f : 1.f)
static size_t GetChannelCount(ChannelConfig chans)
{
switch(chans)
{
case ChannelConfig_Mono: return 1;
case ChannelConfig_Stereo: return 2;
}
return 0;
}
static float GetRolloff(const FRolloffInfo *rolloff, float distance)
{
return soundEngine->GetRolloff(rolloff, distance);
}
ALCdevice *OpenALSoundRenderer::InitDevice()
{
ALCdevice *device = NULL;
if (IsOpenALPresent())
{
if(strcmp(snd_aldevice, "Default") != 0)
{
device = alcOpenDevice(*snd_aldevice);
if(!device)
Printf(TEXTCOLOR_BLUE" Failed to open device " TEXTCOLOR_BOLD"%s" TEXTCOLOR_BLUE". Trying default.\n", *snd_aldevice);
}
if(!device)
{
device = alcOpenDevice(NULL);
if(!device)
{
Printf(TEXTCOLOR_RED" Could not open audio device\n");
}
}
}
else
{
Printf(TEXTCOLOR_ORANGE"Failed to load openal32.dll\n");
}
return device;
}
template<typename T>
static void LoadALFunc(const char *name, T *x)
{
*x = reinterpret_cast<T>(alGetProcAddress(name));
}
template<typename T>
static void LoadALCFunc(ALCdevice *device, const char *name, T *x)
{
*x = reinterpret_cast<T>(alcGetProcAddress(device, name));
}
#define LOAD_FUNC(x) (LoadALFunc(#x, &x))
#define LOAD_DEV_FUNC(d, x) (LoadALCFunc(d, #x, &x))
OpenALSoundRenderer::OpenALSoundRenderer()
: QuitThread(false), Device(NULL), Context(NULL), SFXPaused(0), PrevEnvironment(NULL), EnvSlot(0)
{
EnvFilters[0] = EnvFilters[1] = 0;
Printf("I_InitSound: Initializing OpenAL\n");
Device = InitDevice();
if (Device == NULL) return;
ALC.EXT_EFX = !!alcIsExtensionPresent(Device, "ALC_EXT_EFX");
ALC.EXT_disconnect = !!alcIsExtensionPresent(Device, "ALC_EXT_disconnect");
ALC.SOFT_HRTF = !!alcIsExtensionPresent(Device, "ALC_SOFT_HRTF");
ALC.SOFT_pause_device = !!alcIsExtensionPresent(Device, "ALC_SOFT_pause_device");
const ALCchar *current = NULL;
if(alcIsExtensionPresent(Device, "ALC_ENUMERATE_ALL_EXT"))
current = alcGetString(Device, ALC_ALL_DEVICES_SPECIFIER);
if(alcGetError(Device) != ALC_NO_ERROR || !current)
current = alcGetString(Device, ALC_DEVICE_SPECIFIER);
Printf(" Opened device " TEXTCOLOR_ORANGE"%s\n", current);
ALCint major=0, minor=0;
alcGetIntegerv(Device, ALC_MAJOR_VERSION, 1, &major);
alcGetIntegerv(Device, ALC_MINOR_VERSION, 1, &minor);
DPrintf(DMSG_SPAMMY, " ALC Version: " TEXTCOLOR_BLUE"%d.%d\n", major, minor);
DPrintf(DMSG_SPAMMY, " ALC Extensions: " TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_EXTENSIONS));
TArray<ALCint> attribs;
if(*snd_samplerate > 0)
{
attribs.Push(ALC_FREQUENCY);
attribs.Push(*snd_samplerate);
}
// Make sure one source is capable of stereo output with the rest doing
// mono, without running out of voices
attribs.Push(ALC_MONO_SOURCES);
attribs.Push(std::max<ALCint>(snd_channels, 2) - 1);
attribs.Push(ALC_STEREO_SOURCES);
attribs.Push(1);
if(ALC.SOFT_HRTF)
{
attribs.Push(ALC_HRTF_SOFT);
if(*snd_hrtf == 0)
attribs.Push(ALC_FALSE);
else if(*snd_hrtf > 0)
attribs.Push(ALC_TRUE);
else
attribs.Push(ALC_DONT_CARE_SOFT);
}
// Other attribs..?
attribs.Push(0);
Context = alcCreateContext(Device, &attribs[0]);
if(!Context || alcMakeContextCurrent(Context) == ALC_FALSE)
{
Printf(TEXTCOLOR_RED" Failed to setup context: %s\n", alcGetString(Device, alcGetError(Device)));
if(Context)
alcDestroyContext(Context);
Context = NULL;
alcCloseDevice(Device);
Device = NULL;
return;
}
attribs.Clear();
const ALchar *const version = alGetString(AL_VERSION);
if (strstr(version, "ALSOFT") == nullptr)
{
Printf(TEXTCOLOR_RED " You are using an unsupported OpenAL implementation\n"
" Install OpenAL Soft library for a better experience\n");
}
DPrintf(DMSG_SPAMMY, " Vendor: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VENDOR));
DPrintf(DMSG_SPAMMY, " Renderer: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_RENDERER));
DPrintf(DMSG_SPAMMY, " Version: " TEXTCOLOR_ORANGE"%s\n", version);
DPrintf(DMSG_SPAMMY, " Extensions: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_EXTENSIONS));
AL.EXT_source_distance_model = !!alIsExtensionPresent("AL_EXT_source_distance_model");
AL.EXT_SOURCE_RADIUS = !!alIsExtensionPresent("AL_EXT_SOURCE_RADIUS");
AL.SOFT_deferred_updates = !!alIsExtensionPresent("AL_SOFT_deferred_updates");
AL.SOFT_loop_points = !!alIsExtensionPresent("AL_SOFT_loop_points");
AL.SOFT_source_resampler = !!alIsExtensionPresent("AL_SOFT_source_resampler");
AL.SOFT_source_spatialize = !!alIsExtensionPresent("AL_SOFT_source_spatialize");
// Speed of sound is in units per second. Presuming we want to simulate a
// typical speed of sound of 343.3 meters per second, multiply it by the
// units per meter scale (1), and set the meters per unit to the scale's
// reciprocal. It's important to set these correctly for both doppler
// effects and reverb.
alSpeedOfSound(343.3f);
if(ALC.EXT_EFX)
alListenerf(AL_METERS_PER_UNIT, 1.0f);
alDistanceModel(AL_INVERSE_DISTANCE);
if(AL.EXT_source_distance_model)
alEnable(AL_SOURCE_DISTANCE_MODEL);
if(AL.SOFT_deferred_updates)
{
LOAD_FUNC(alDeferUpdatesSOFT);
LOAD_FUNC(alProcessUpdatesSOFT);
}
else
{
alDeferUpdatesSOFT = _wrap_DeferUpdatesSOFT;
alProcessUpdatesSOFT = _wrap_ProcessUpdatesSOFT;
}
if(AL.SOFT_source_resampler)
LOAD_FUNC(alGetStringiSOFT);
if(ALC.SOFT_pause_device)
{
LOAD_DEV_FUNC(Device, alcDevicePauseSOFT);
LOAD_DEV_FUNC(Device, alcDeviceResumeSOFT);
}
ALenum err = getALError();
if(err != AL_NO_ERROR)
{
alcMakeContextCurrent(NULL);
alcDestroyContext(Context);
Context = NULL;
alcCloseDevice(Device);
Device = NULL;
return;
}
ALCint refresh=0;
alcGetIntegerv(Device, ALC_REFRESH, 1, &refresh);
if(refresh > 0)
{
// Round up instead of down
UpdateTimeMS = (1000+refresh-1) / refresh;
}
ALCint numMono=0, numStereo=0;
alcGetIntegerv(Device, ALC_MONO_SOURCES, 1, &numMono);
alcGetIntegerv(Device, ALC_STEREO_SOURCES, 1, &numStereo);
// OpenAL specification doesn't require alcGetIntegerv() to return
// meaningful values for ALC_MONO_SOURCES and ALC_MONO_SOURCES.
// At least Apple's OpenAL implementation returns zeroes,
// although it can generate reasonable number of sources.
const int numChannels = std::max<int>(snd_channels, 2);
int numSources = numMono + numStereo;
if (0 == numSources)
{
numSources = numChannels;
}
Sources.Resize(std::min<int>(numChannels, numSources));
for(unsigned i = 0;i < Sources.Size();i++)
{
alGenSources(1, &Sources[i]);
if(getALError() != AL_NO_ERROR)
{
Sources.Resize(i);
Sources.ShrinkToFit();
break;
}
}
if(Sources.Size() == 0)
{
Printf(TEXTCOLOR_RED" Error: could not generate any sound sources!\n");
alcMakeContextCurrent(NULL);
alcDestroyContext(Context);
Context = NULL;
alcCloseDevice(Device);
Device = NULL;
return;
}
FreeSfx = Sources;
DPrintf(DMSG_NOTIFY, " Allocated " TEXTCOLOR_BLUE"%u" TEXTCOLOR_NORMAL" sources\n", Sources.Size());
WasInWater = false;
if(*snd_efx && ALC.EXT_EFX)
{
// EFX function pointers
LOAD_FUNC(alGenEffects);
LOAD_FUNC(alDeleteEffects);
LOAD_FUNC(alIsEffect);
LOAD_FUNC(alEffecti);
LOAD_FUNC(alEffectiv);
LOAD_FUNC(alEffectf);
LOAD_FUNC(alEffectfv);
LOAD_FUNC(alGetEffecti);
LOAD_FUNC(alGetEffectiv);
LOAD_FUNC(alGetEffectf);
LOAD_FUNC(alGetEffectfv);
LOAD_FUNC(alGenFilters);
LOAD_FUNC(alDeleteFilters);
LOAD_FUNC(alIsFilter);
LOAD_FUNC(alFilteri);
LOAD_FUNC(alFilteriv);
LOAD_FUNC(alFilterf);
LOAD_FUNC(alFilterfv);
LOAD_FUNC(alGetFilteri);
LOAD_FUNC(alGetFilteriv);
LOAD_FUNC(alGetFilterf);
LOAD_FUNC(alGetFilterfv);
LOAD_FUNC(alGenAuxiliaryEffectSlots);
LOAD_FUNC(alDeleteAuxiliaryEffectSlots);
LOAD_FUNC(alIsAuxiliaryEffectSlot);
LOAD_FUNC(alAuxiliaryEffectSloti);
LOAD_FUNC(alAuxiliaryEffectSlotiv);
LOAD_FUNC(alAuxiliaryEffectSlotf);
LOAD_FUNC(alAuxiliaryEffectSlotfv);
LOAD_FUNC(alGetAuxiliaryEffectSloti);
LOAD_FUNC(alGetAuxiliaryEffectSlotiv);
LOAD_FUNC(alGetAuxiliaryEffectSlotf);
LOAD_FUNC(alGetAuxiliaryEffectSlotfv);
if(getALError() == AL_NO_ERROR)
{
ALuint envReverb;
alGenEffects(1, &envReverb);
if(getALError() == AL_NO_ERROR)
{
alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
if(alGetError() == AL_NO_ERROR)
DPrintf(DMSG_SPAMMY, " EAX Reverb found\n");
alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_REVERB);
if(alGetError() == AL_NO_ERROR)
DPrintf(DMSG_SPAMMY, " Standard Reverb found\n");
alDeleteEffects(1, &envReverb);
getALError();
}
alGenAuxiliaryEffectSlots(1, &EnvSlot);
alGenFilters(2, EnvFilters);
if(getALError() == AL_NO_ERROR)
{
alFilteri(EnvFilters[0], AL_FILTER_TYPE, AL_FILTER_LOWPASS);
alFilteri(EnvFilters[1], AL_FILTER_TYPE, AL_FILTER_LOWPASS);
if(getALError() == AL_NO_ERROR)
DPrintf(DMSG_SPAMMY, " Lowpass found\n");
else
{
alDeleteFilters(2, EnvFilters);
EnvFilters[0] = EnvFilters[1] = 0;
alDeleteAuxiliaryEffectSlots(1, &EnvSlot);
EnvSlot = 0;
getALError();
}
}
else
{
alDeleteFilters(2, EnvFilters);
alDeleteAuxiliaryEffectSlots(1, &EnvSlot);
EnvFilters[0] = EnvFilters[1] = 0;
EnvSlot = 0;
getALError();
}
}
}
if(EnvSlot)
Printf(" EFX enabled\n");
if(AL.SOFT_source_resampler && strcmp(*snd_alresampler, "Default") != 0)
{
const ALint num_resamplers = alGetInteger(AL_NUM_RESAMPLERS_SOFT);
ALint ridx = alGetInteger(AL_DEFAULT_RESAMPLER_SOFT);
ALint i;
for(i = 0;i < num_resamplers;i++)
{
if(strcmp(alGetStringiSOFT(AL_RESAMPLER_NAME_SOFT, i), *snd_alresampler) == 0)
{
ridx = i;
break;
}
}
if(i == num_resamplers)
Printf(TEXTCOLOR_RED" Failed to find resampler " TEXTCOLOR_ORANGE"%s\n", *snd_alresampler);
else for(ALint src : Sources)
alSourcei(src, AL_SOURCE_RESAMPLER_SOFT, ridx);
}
}
#undef LOAD_DEV_FUNC
#undef LOAD_FUNC
OpenALSoundRenderer::~OpenALSoundRenderer()
{
if(!Device)
return;
if(StreamThread.joinable())
{
std::unique_lock<std::mutex> lock(StreamLock);
QuitThread.store(true);
lock.unlock();
StreamWake.notify_all();
StreamThread.join();
}
while(Streams.Size() > 0)
delete Streams[0];
alDeleteSources(Sources.Size(), &Sources[0]);
Sources.Clear();
FreeSfx.Clear();
SfxGroup.Clear();
PausableSfx.Clear();
ReverbSfx.Clear();
if(EnvEffects.CountUsed() > 0)
{
EffectMapIter iter(EnvEffects);
EffectMap::Pair *pair;
while(iter.NextPair(pair))
alDeleteEffects(1, &(pair->Value));
}
EnvEffects.Clear();
if(EnvSlot)
{
alDeleteAuxiliaryEffectSlots(1, &EnvSlot);
alDeleteFilters(2, EnvFilters);
}
EnvSlot = 0;
EnvFilters[0] = EnvFilters[1] = 0;
alcMakeContextCurrent(NULL);
alcDestroyContext(Context);
Context = NULL;
alcCloseDevice(Device);
Device = NULL;
}
void OpenALSoundRenderer::BackgroundProc()
{
std::unique_lock<std::mutex> lock(StreamLock);
while(!QuitThread.load())
{
if(Streams.Size() == 0)
{
// If there's nothing to play, wait indefinitely.
StreamWake.wait(lock);
}
else
{
// Else, process all active streams and sleep for 100ms
for(size_t i = 0;i < Streams.Size();i++)
Streams[i]->Process();
StreamWake.wait_for(lock, std::chrono::milliseconds(100));
}
}
}
void OpenALSoundRenderer::AddStream(OpenALSoundStream *stream)
{
std::unique_lock<std::mutex> lock(StreamLock);
Streams.Push(stream);
lock.unlock();
// There's a stream to play, make sure the background thread is aware
StreamWake.notify_all();
}
void OpenALSoundRenderer::RemoveStream(OpenALSoundStream *stream)
{
std::unique_lock<std::mutex> lock(StreamLock);
unsigned int idx = Streams.Find(stream);
if(idx < Streams.Size())
Streams.Delete(idx);
}
void OpenALSoundRenderer::SetSfxVolume(float volume)
{
SfxVolume = volume;
if (!soundEngine) return;
FSoundChan *schan = soundEngine->GetChannels();
while(schan)
{
if(schan->SysChannel != NULL)
{
ALuint source = GET_PTRID(schan->SysChannel);
volume = SfxVolume;
alDeferUpdatesSOFT();
alSourcef(source, AL_MAX_GAIN, volume);
alSourcef(source, AL_GAIN, volume * schan->Volume);
}
schan = schan->NextChan;
}
alProcessUpdatesSOFT();
getALError();
}
void OpenALSoundRenderer::SetMusicVolume(float volume)
{
MusicVolume = volume;
for(uint32_t i = 0;i < Streams.Size();++i)
Streams[i]->UpdateVolume();
}
unsigned int OpenALSoundRenderer::GetMSLength(SoundHandle sfx)
{
if(sfx.data)
{
ALuint buffer = GET_PTRID(sfx.data);
if(alIsBuffer(buffer))
{
ALint bits, channels, freq, size;
alGetBufferi(buffer, AL_BITS, &bits);
alGetBufferi(buffer, AL_CHANNELS, &channels);
alGetBufferi(buffer, AL_FREQUENCY, &freq);
alGetBufferi(buffer, AL_SIZE, &size);
if(getALError() == AL_NO_ERROR)
return (unsigned int)(size / (channels*bits/8) * 1000. / freq);
}
}
return 0;
}
unsigned int OpenALSoundRenderer::GetSampleLength(SoundHandle sfx)
{
if(sfx.data)
{
ALuint buffer = GET_PTRID(sfx.data);
ALint bits, channels, size;
alGetBufferi(buffer, AL_BITS, &bits);
alGetBufferi(buffer, AL_CHANNELS, &channels);
alGetBufferi(buffer, AL_SIZE, &size);
if(getALError() == AL_NO_ERROR)
return (ALsizei)(size / (channels * bits / 8));
}
return 0;
}
float OpenALSoundRenderer::GetOutputRate()
{
ALCint rate = 44100; // Default, just in case
alcGetIntegerv(Device, ALC_FREQUENCY, 1, &rate);
return (float)rate;
}
std::pair<SoundHandle,bool> OpenALSoundRenderer::LoadSoundRaw(uint8_t *sfxdata, int length, int frequency, int channels, int bits, int loopstart, int loopend, bool monoize)
{
SoundHandle retval = { NULL };
if(length == 0) return std::make_pair(retval, true);
/* Only downmix to mono if we can't spatialize multi-channel sounds. */
monoize = monoize && !AL.SOFT_source_spatialize;
if(bits == -8)
{
// Simple signed->unsigned conversion
for(int i = 0;i < length;i++)
sfxdata[i] ^= 0x80;
bits = -bits;
}
if(channels > 1 && monoize)
{
size_t frames = length / channels * 8 / bits;
if(bits == 16)
{
for(size_t i = 0;i < frames;i++)
{
int sum = 0;
for(int c = 0;c < channels;c++)
sum += ((short*)sfxdata)[i*channels + c];
((short*)sfxdata)[i] = sum / channels;
}
}
else if(bits == 8)
{
for(size_t i = 0;i < frames;i++)
{
int sum = 0;
for(int c = 0;c < channels;c++)
sum += sfxdata[i*channels + c] - 128;
sfxdata[i] = (sum / channels) + 128;
}
}
length /= channels;
channels = 1;
}
ALenum format = AL_NONE;
if(bits == 16)
{
if(channels == 1) format = AL_FORMAT_MONO16;
if(channels == 2) format = AL_FORMAT_STEREO16;
}
else if(bits == 8)
{
if(channels == 1) format = AL_FORMAT_MONO8;
if(channels == 2) format = AL_FORMAT_STEREO8;
}
if(format == AL_NONE || frequency <= 0)
{
Printf("Unhandled format: %d bit, %d channel, %d hz\n", bits, channels, frequency);
return std::make_pair(retval, true);
}
length -= length%(channels*bits/8);
ALenum err;
ALuint buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, format, sfxdata, length, frequency);
if((err=getALError()) != AL_NO_ERROR)
{
Printf("Failed to buffer data: %s\n", alGetString(err));
alDeleteBuffers(1, &buffer);
getALError();
return std::make_pair(retval, true);
}
if((loopstart > 0 || loopend > 0) && AL.SOFT_loop_points)
{
if(loopstart < 0)
loopstart = 0;
if(loopend < loopstart)
loopend = length / (channels*bits/8);
ALint loops[2] = { loopstart, loopend };
DPrintf(DMSG_NOTIFY, "Setting loop points %d -> %d\n", loops[0], loops[1]);
alBufferiv(buffer, AL_LOOP_POINTS_SOFT, loops);
getALError();
}
else if(loopstart > 0 || loopend > 0)
{
static bool warned = false;
if(!warned)
Printf(DMSG_WARNING, "Loop points not supported!\n");
warned = true;
}
retval.data = MAKE_PTRID(buffer);
return std::make_pair(retval, AL.SOFT_source_spatialize || channels==1);
}
std::pair<SoundHandle,bool> OpenALSoundRenderer::LoadSound(uint8_t *sfxdata, int length, bool monoize, FSoundLoadBuffer *pBuffer)
{
SoundHandle retval = { NULL };
ALenum format = AL_NONE;
ChannelConfig chans;
SampleType type;
int srate;
uint32_t loop_start = 0, loop_end = ~0u;
bool startass = false, endass = false;
/* Only downmix to mono if we can't spatialize multi-channel sounds. */
monoize = monoize && !AL.SOFT_source_spatialize;
auto mreader = new MusicIO::MemoryReader(sfxdata, length);
FindLoopTags(mreader, &loop_start, &startass, &loop_end, &endass);
mreader->seek(0, SEEK_SET);
std::unique_ptr<SoundDecoder> decoder(SoundDecoder::CreateDecoder(mreader));
if (!decoder)
{
delete mreader;
return std::make_pair(retval, true);
}
// the decode will take ownership of the reader here.
decoder->getInfo(&srate, &chans, &type);
int samplesize = 1;
if (chans == ChannelConfig_Mono || monoize)
{
if (type == SampleType_UInt8) format = AL_FORMAT_MONO8, samplesize = 1;
if (type == SampleType_Int16) format = AL_FORMAT_MONO16, samplesize = 2;
}
else if (chans == ChannelConfig_Stereo)
{
if (type == SampleType_UInt8) format = AL_FORMAT_STEREO8, samplesize = 2;
if (type == SampleType_Int16) format = AL_FORMAT_STEREO16, samplesize = 4;
}
if (format == AL_NONE)
{
Printf("Unsupported audio format: %s, %s\n", GetChannelConfigName(chans),
GetSampleTypeName(type));
return std::make_pair(retval, true);
}
auto data = decoder->readAll();
if(chans != ChannelConfig_Mono && monoize)
{
size_t chancount = GetChannelCount(chans);
size_t frames = data.size() / chancount /
(type == SampleType_Int16 ? 2 : 1);
if(type == SampleType_Int16)
{
short *sfxdata = (short*)&data[0];
for(size_t i = 0;i < frames;i++)
{
int sum = 0;
for(size_t c = 0;c < chancount;c++)
sum += sfxdata[i*chancount + c];
sfxdata[i] = short(sum / chancount);
}
}
else if(type == SampleType_UInt8)
{
uint8_t *sfxdata = (uint8_t*)&data[0];
for(size_t i = 0;i < frames;i++)
{
int sum = 0;
for(size_t c = 0;c < chancount;c++)
sum += sfxdata[i*chancount + c] - 128;
sfxdata[i] = uint8_t((sum / chancount) + 128);
}
}
data.resize((data.size()/chancount));
}
ALenum err;
ALuint buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, format, &data[0], (ALsizei)data.size(), srate);
if((err=getALError()) != AL_NO_ERROR)
{
Printf("Failed to buffer data: %s\n", alGetString(err));
alDeleteBuffers(1, &buffer);
getALError();
return std::make_pair(retval, true);
}
if (!startass) loop_start = uint32_t(int64_t(loop_start) * srate / 1000);
if (!endass && loop_end != ~0u) loop_end = uint32_t(int64_t(loop_end) * srate / 1000);
const uint32_t samples = (uint32_t)data.size() / samplesize;
if (loop_start > samples) loop_start = 0;
if (loop_end > samples) loop_end = samples;
if ((loop_start > 0 || loop_end > 0) && loop_end > loop_start && AL.SOFT_loop_points)
{
ALint loops[2] = { static_cast<ALint>(loop_start), static_cast<ALint>(loop_end) };
DPrintf(DMSG_NOTIFY, "Setting loop points %d -> %d\n", loops[0], loops[1]);
alBufferiv(buffer, AL_LOOP_POINTS_SOFT, loops);
// no console messages here, please!
}
retval.data = MAKE_PTRID(buffer);
if (pBuffer != nullptr)
{
pBuffer->mBuffer = std::move(data);
pBuffer->loop_start = loop_start;
pBuffer->loop_end = loop_end;
pBuffer->chans = chans;
pBuffer->type = type;
pBuffer->srate = srate;
}
return std::make_pair(retval, AL.SOFT_source_spatialize || chans == ChannelConfig_Mono || monoize);
}
std::pair<SoundHandle, bool> OpenALSoundRenderer::LoadSoundBuffered(FSoundLoadBuffer *pBuffer, bool monoize)
{
SoundHandle retval = { NULL };
ALenum format = AL_NONE;
int srate = pBuffer->srate;
auto type = pBuffer->type;
auto chans = pBuffer->chans;
uint32_t loop_start = pBuffer->loop_start, loop_end = pBuffer->loop_end;
/* Only downmix to mono if we can't spatialize multi-channel sounds. */
monoize = monoize && !AL.SOFT_source_spatialize;
if (chans == ChannelConfig_Mono || monoize)
{
if (type == SampleType_UInt8) format = AL_FORMAT_MONO8;
if (type == SampleType_Int16) format = AL_FORMAT_MONO16;
}
else if (chans == ChannelConfig_Stereo)
{
if (type == SampleType_UInt8) format = AL_FORMAT_STEREO8;
if (type == SampleType_Int16) format = AL_FORMAT_STEREO16;
}
if (format == AL_NONE)
{
Printf("Unsupported audio format: %s, %s\n", GetChannelConfigName(chans),
GetSampleTypeName(type));
return std::make_pair(retval, true);
}
auto &data = pBuffer->mBuffer;
if (pBuffer->chans == ChannelConfig_Stereo && monoize)
{
size_t chancount = GetChannelCount(chans);
size_t frames = data.size() / chancount /
(type == SampleType_Int16 ? 2 : 1);
if (type == SampleType_Int16)
{
short *sfxdata = (short*)&data[0];
for (size_t i = 0; i < frames; i++)
{
int sum = 0;
for (size_t c = 0; c < chancount; c++)
sum += sfxdata[i*chancount + c];
sfxdata[i] = short(sum / chancount);
}
}
else if (type == SampleType_UInt8)
{
uint8_t *sfxdata = (uint8_t*)&data[0];
for (size_t i = 0; i < frames; i++)
{
int sum = 0;
for (size_t c = 0; c < chancount; c++)
sum += sfxdata[i*chancount + c] - 128;
sfxdata[i] = uint8_t((sum / chancount) + 128);
}
}
data.resize(data.size() / chancount);
}
ALenum err;
ALuint buffer = 0;
alGenBuffers(1, &buffer);
alBufferData(buffer, format, &data[0], (ALsizei)data.size(), srate);
if ((err = getALError()) != AL_NO_ERROR)
{
Printf("Failed to buffer data: %s\n", alGetString(err));
alDeleteBuffers(1, &buffer);
getALError();
return std::make_pair(retval, true);
}
// the loop points were already validated by the previous load.
if ((loop_start > 0 || loop_end > 0) && loop_end > loop_start && AL.SOFT_loop_points)
{
ALint loops[2] = { static_cast<ALint>(loop_start), static_cast<ALint>(loop_end) };
DPrintf(DMSG_NOTIFY, "Setting loop points %d -> %d\n", loops[0], loops[1]);
alBufferiv(buffer, AL_LOOP_POINTS_SOFT, loops);
// no console messages here, please!
}
retval.data = MAKE_PTRID(buffer);
return std::make_pair(retval, AL.SOFT_source_spatialize || chans == ChannelConfig_Mono || monoize);
}
void OpenALSoundRenderer::UnloadSound(SoundHandle sfx)
{
if(!sfx.data)
return;
ALuint buffer = GET_PTRID(sfx.data);
FSoundChan *schan = soundEngine->GetChannels();
while(schan)
{
if(schan->SysChannel)
{
ALint bufID = 0;
alGetSourcei(GET_PTRID(schan->SysChannel), AL_BUFFER, &bufID);
if((ALuint)bufID == buffer)
{
FSoundChan *next = schan->NextChan;
ForceStopChannel(schan);
schan = next;
continue;
}
}
schan = schan->NextChan;
}
// Make sure to kill any currently fading sounds too
for(auto iter = FadingSources.begin();iter != FadingSources.end();)
{
ALint bufID = 0;
alGetSourcei(iter->first, AL_BUFFER, &bufID);
if(static_cast<ALuint>(bufID) == buffer)
{
FreeSource(iter->first);
iter = FadingSources.erase(iter);
}
else
++iter;
}
alDeleteBuffers(1, &buffer);
getALError();
}
SoundStream *OpenALSoundRenderer::CreateStream(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata)
{
if(StreamThread.get_id() == std::thread::id())
StreamThread = std::thread(std::mem_fn(&OpenALSoundRenderer::BackgroundProc), this);
OpenALSoundStream *stream = new OpenALSoundStream(this);
if (!stream->Init(callback, buffbytes, flags, samplerate, userdata))
{
delete stream;
return NULL;
}
return stream;
}
FISoundChannel *OpenALSoundRenderer::StartSound(SoundHandle sfx, float vol, int pitch, int chanflags, FISoundChannel *reuse_chan)
{
if(FreeSfx.Size() == 0)
{
FSoundChan *lowest = FindLowestChannel();
if(lowest) ForceStopChannel(lowest);
if(FreeSfx.Size() == 0)
return NULL;
}
ALuint buffer = GET_PTRID(sfx.data);
ALuint source = FreeSfx.Last();
alSource3f(source, AL_POSITION, 0.f, 0.f, 0.f);
alSource3f(source, AL_VELOCITY, 0.f, 0.f, 0.f);
alSource3f(source, AL_DIRECTION, 0.f, 0.f, 0.f);
alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
alSourcei(source, AL_LOOPING, (chanflags&SNDF_LOOP) ? AL_TRUE : AL_FALSE);
alSourcef(source, AL_REFERENCE_DISTANCE, 1.f);
alSourcef(source, AL_MAX_DISTANCE, 1000.f);
alSourcef(source, AL_DOPPLER_FACTOR, 0.f);
alSourcef(source, AL_ROLLOFF_FACTOR, 0.f);
alSourcef(source, AL_MAX_GAIN, SfxVolume);
alSourcef(source, AL_GAIN, SfxVolume*vol);
if(AL.EXT_SOURCE_RADIUS)
alSourcef(source, AL_SOURCE_RADIUS, 0.f);
if(AL.SOFT_source_spatialize)
alSourcei(source, AL_SOURCE_SPATIALIZE_SOFT, AL_AUTO_SOFT);
if(EnvSlot)
{
if(!(chanflags&SNDF_NOREVERB))
{
alSourcei(source, AL_DIRECT_FILTER, EnvFilters[0]);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]);
}
else
{
alSourcei(source, AL_DIRECT_FILTER, AL_FILTER_NULL);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
}
alSourcef(source, AL_ROOM_ROLLOFF_FACTOR, 0.f);
}
if(WasInWater && !(chanflags&SNDF_NOREVERB))
alSourcef(source, AL_PITCH, PITCH(pitch)*PITCH_MULT);
else
alSourcef(source, AL_PITCH, PITCH(pitch));
if(!reuse_chan || reuse_chan->StartTime == 0)
alSourcef(source, AL_SEC_OFFSET, 0.f);
else
{
if((chanflags&SNDF_ABSTIME))
alSourcei(source, AL_SAMPLE_OFFSET, ALint(reuse_chan->StartTime));
else
{
float offset = std::chrono::duration_cast<std::chrono::duration<float>>(
std::chrono::steady_clock::now().time_since_epoch() -
std::chrono::steady_clock::time_point::duration(reuse_chan->StartTime)
).count();
if(offset > 0.f) alSourcef(source, AL_SEC_OFFSET, offset);
}
}
if(getALError() != AL_NO_ERROR)
return NULL;
alSourcei(source, AL_BUFFER, buffer);
if((chanflags&SNDF_NOPAUSE) || !SFXPaused)
alSourcePlay(source);
if(getALError() != AL_NO_ERROR)
{
alSourcei(source, AL_BUFFER, 0);
getALError();
return NULL;
}
if(!(chanflags&SNDF_NOREVERB))
ReverbSfx.Push(source);
if(!(chanflags&SNDF_NOPAUSE))
PausableSfx.Push(source);
SfxGroup.Push(source);
FreeSfx.Pop();
FISoundChannel *chan = reuse_chan;
if(!chan) chan = soundEngine->GetChannel(MAKE_PTRID(source));
else chan->SysChannel = MAKE_PTRID(source);
chan->Rolloff.RolloffType = ROLLOFF_Log;
chan->Rolloff.RolloffFactor = 0.f;
chan->Rolloff.MinDistance = 1.f;
chan->DistanceSqr = 0.f;
chan->ManualRolloff = false;
return chan;
}
FISoundChannel *OpenALSoundRenderer::StartSound3D(SoundHandle sfx, SoundListener *listener, float vol,
FRolloffInfo *rolloff, float distscale, int pitch, int priority, const FVector3 &pos, const FVector3 &vel,
int channum, int chanflags, FISoundChannel *reuse_chan)
{
float dist_sqr = (float)(pos - listener->position).LengthSquared();
if(FreeSfx.Size() == 0)
{
FSoundChan *lowest = FindLowestChannel();
if(lowest)
{
if(lowest->Priority < priority || (lowest->Priority == priority &&
lowest->DistanceSqr > dist_sqr))
ForceStopChannel(lowest);
}
if(FreeSfx.Size() == 0)
return NULL;
}
bool manualRolloff = true;
ALuint buffer = GET_PTRID(sfx.data);
ALuint source = FreeSfx.Last();
if(rolloff->RolloffType == ROLLOFF_Log)
{
if(AL.EXT_source_distance_model)
alSourcei(source, AL_DISTANCE_MODEL, AL_INVERSE_DISTANCE);
alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale);
alSourcef(source, AL_MAX_DISTANCE, (1000.f+rolloff->MinDistance)/distscale);
alSourcef(source, AL_ROLLOFF_FACTOR, rolloff->RolloffFactor);
manualRolloff = false;
}
else if(rolloff->RolloffType == ROLLOFF_Linear && AL.EXT_source_distance_model)
{
alSourcei(source, AL_DISTANCE_MODEL, AL_LINEAR_DISTANCE);
alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale);
alSourcef(source, AL_MAX_DISTANCE, rolloff->MaxDistance/distscale);
alSourcef(source, AL_ROLLOFF_FACTOR, 1.f);
manualRolloff = false;
}
if(manualRolloff)
{
// How manual rolloff works:
//
// If a sound is using Custom or Doom style rolloff, or Linear style
// when AL_EXT_source_distance_model is not supported, we have to play
// around a bit to get appropriate distance attenation. What we do is
// calculate the attenuation that should be applied, then given an
// Inverse Distance rolloff model with OpenAL, reverse the calculation
// to get the distance needed for that much attenuation. The Inverse
// Distance calculation is:
//
// Gain = MinDist / (MinDist + RolloffFactor*(Distance - MinDist))
//
// Thus, the reverse is:
//
// Distance = (MinDist/Gain - MinDist)/RolloffFactor + MinDist
//
// This can be simplified by using a MinDist and RolloffFactor of 1,
// which makes it:
//
// Distance = 1.0f/Gain;
//
// The source position is then set that many units away from the
// listener position, and OpenAL takes care of the rest.
if(AL.EXT_source_distance_model)
alSourcei(source, AL_DISTANCE_MODEL, AL_INVERSE_DISTANCE);
alSourcef(source, AL_REFERENCE_DISTANCE, 1.f);
alSourcef(source, AL_MAX_DISTANCE, 100000.f);
alSourcef(source, AL_ROLLOFF_FACTOR, 1.f);
FVector3 dir = pos - listener->position;
if(dir.DoesNotApproximatelyEqual(FVector3(0.f, 0.f, 0.f)))
{
float gain = GetRolloff(rolloff, sqrtf(dist_sqr) * distscale);
dir.MakeResize((gain > 0.00001f) ? 1.f/gain : 100000.f);
}
if(AL.EXT_SOURCE_RADIUS)
{
/* Since the OpenAL distance is decoupled from the sound's distance, get the OpenAL
* distance that corresponds to the area radius. */
alSourcef(source, AL_SOURCE_RADIUS, (chanflags&SNDF_AREA) ?
// Clamp in case the max distance is <= the area radius
1.f/std::max<float>(GetRolloff(rolloff, AREA_SOUND_RADIUS), 0.00001f) : 0.f
);
}
else if((chanflags&SNDF_AREA) && dist_sqr < AREA_SOUND_RADIUS*AREA_SOUND_RADIUS)
{
FVector3 amb(0.f, !(dir.Y>=0.f) ? -1.f : 1.f, 0.f);
float a = sqrtf(dist_sqr) / AREA_SOUND_RADIUS;
dir = amb + (dir-amb)*a;
}
dir += listener->position;
if(dist_sqr < (0.0004f*0.0004f))
{
// Head relative
alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
alSource3f(source, AL_POSITION, 0.f, 0.f, 0.f);
}
else
{
alSourcei(source, AL_SOURCE_RELATIVE, AL_FALSE);
alSource3f(source, AL_POSITION, dir[0], dir[1], -dir[2]);
}
}
else
{
FVector3 dir = pos;
if(AL.EXT_SOURCE_RADIUS)
alSourcef(source, AL_SOURCE_RADIUS, (chanflags&SNDF_AREA) ? AREA_SOUND_RADIUS : 0.f);
else if((chanflags&SNDF_AREA) && dist_sqr < AREA_SOUND_RADIUS*AREA_SOUND_RADIUS)
{
dir -= listener->position;
float mindist = rolloff->MinDistance/distscale;
FVector3 amb(0.f, !(dir.Y>=0.f) ? -mindist : mindist, 0.f);
float a = sqrtf(dist_sqr) / AREA_SOUND_RADIUS;
dir = amb + (dir-amb)*a;
dir += listener->position;
}
if(dist_sqr < (0.0004f*0.0004f))
{
// Head relative
alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
alSource3f(source, AL_POSITION, 0.f, 0.f, 0.f);
}
else
{
alSourcei(source, AL_SOURCE_RELATIVE, AL_FALSE);
alSource3f(source, AL_POSITION, dir[0], dir[1], -dir[2]);
}
}
alSource3f(source, AL_VELOCITY, vel[0], vel[1], -vel[2]);
alSource3f(source, AL_DIRECTION, 0.f, 0.f, 0.f);
alSourcef(source, AL_DOPPLER_FACTOR, 0.f);
alSourcei(source, AL_LOOPING, (chanflags&SNDF_LOOP) ? AL_TRUE : AL_FALSE);
alSourcef(source, AL_MAX_GAIN, SfxVolume);
alSourcef(source, AL_GAIN, SfxVolume*vol);
if(AL.SOFT_source_spatialize)
alSourcei(source, AL_SOURCE_SPATIALIZE_SOFT, AL_TRUE);
if(EnvSlot)
{
if(!(chanflags&SNDF_NOREVERB))
{
alSourcei(source, AL_DIRECT_FILTER, EnvFilters[0]);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]);
}
else
{
alSourcei(source, AL_DIRECT_FILTER, AL_FILTER_NULL);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
}
alSourcef(source, AL_ROOM_ROLLOFF_FACTOR, 0.f);
}
if(WasInWater && !(chanflags&SNDF_NOREVERB))
alSourcef(source, AL_PITCH, PITCH(pitch)*PITCH_MULT);
else
alSourcef(source, AL_PITCH, PITCH(pitch));
if(!reuse_chan || reuse_chan->StartTime == 0)
alSourcef(source, AL_SEC_OFFSET, 0.f);
else
{
if((chanflags&SNDF_ABSTIME))
alSourcei(source, AL_SAMPLE_OFFSET, ALint(reuse_chan->StartTime));
else
{
float offset = std::chrono::duration_cast<std::chrono::duration<float>>(
std::chrono::steady_clock::now().time_since_epoch() -
std::chrono::steady_clock::time_point::duration(reuse_chan->StartTime)
).count();
if(offset > 0.f) alSourcef(source, AL_SEC_OFFSET, offset);
}
}
if(getALError() != AL_NO_ERROR)
return NULL;
alSourcei(source, AL_BUFFER, buffer);
if((chanflags&SNDF_NOPAUSE) || !SFXPaused)
alSourcePlay(source);
if(getALError() != AL_NO_ERROR)
{
alSourcei(source, AL_BUFFER, 0);
getALError();
return NULL;
}
if(!(chanflags&SNDF_NOREVERB))
ReverbSfx.Push(source);
if(!(chanflags&SNDF_NOPAUSE))
PausableSfx.Push(source);
SfxGroup.Push(source);
FreeSfx.Pop();
FISoundChannel *chan = reuse_chan;
if(!chan) chan = soundEngine->GetChannel(MAKE_PTRID(source));
else chan->SysChannel = MAKE_PTRID(source);
chan->Rolloff = *rolloff;
chan->DistanceSqr = dist_sqr;
chan->ManualRolloff = manualRolloff;
return chan;
}
void OpenALSoundRenderer::ChannelVolume(FISoundChannel *chan, float volume)
{
if(chan == NULL || chan->SysChannel == NULL)
return;
alDeferUpdatesSOFT();
ALuint source = GET_PTRID(chan->SysChannel);
alSourcef(source, AL_GAIN, SfxVolume * volume);
}
void OpenALSoundRenderer::ChannelPitch(FISoundChannel *chan, float pitch)
{
if (chan == NULL || chan->SysChannel == NULL)
return;
alDeferUpdatesSOFT();
ALuint source = GET_PTRID(chan->SysChannel);
if (WasInWater && !(chan->ChanFlags & CHANF_UI))
alSourcef(source, AL_PITCH, std::max(pitch, 0.0001f)*PITCH_MULT);
else
alSourcef(source, AL_PITCH, std::max(pitch, 0.0001f));
}
void OpenALSoundRenderer::FreeSource(ALuint source)
{
alSourceRewind(source);
alSourcei(source, AL_BUFFER, 0);
getALError();
uint32_t i;
if((i=PausableSfx.Find(source)) < PausableSfx.Size())
PausableSfx.Delete(i);
if((i=ReverbSfx.Find(source)) < ReverbSfx.Size())
ReverbSfx.Delete(i);
if((i=SfxGroup.Find(source)) < SfxGroup.Size())
SfxGroup.Delete(i);
FreeSfx.Push(source);
}
void OpenALSoundRenderer::StopChannel(FISoundChannel *chan)
{
if(chan == NULL || chan->SysChannel == NULL)
return;
ALuint source = GET_PTRID(chan->SysChannel);
// Release first, so it can be properly marked as evicted if it's being killed
soundEngine->ChannelEnded(chan);
ALint state = AL_INITIAL;
alGetSourcei(source, AL_SOURCE_STATE, &state);
if(state != AL_PLAYING)
FreeSource(source);
else
{
// The sound is being killed while playing, so set its gain to 0 and track it
// as it fades.
alSourcef(source, AL_GAIN, 0.f);
getALError();
FadingSources.insert(std::make_pair(
source, std::chrono::steady_clock::now().time_since_epoch().count()
));
}
}
void OpenALSoundRenderer::ForceStopChannel(FISoundChannel *chan)
{
ALuint source = GET_PTRID(chan->SysChannel);
if(!source) return;
soundEngine->ChannelEnded(chan);
FreeSource(source);
}
unsigned int OpenALSoundRenderer::GetPosition(FISoundChannel *chan)
{
if(chan == NULL || chan->SysChannel == NULL)
return 0;
ALint pos;
alGetSourcei(GET_PTRID(chan->SysChannel), AL_SAMPLE_OFFSET, &pos);
if(getALError() == AL_NO_ERROR)
return pos;
return 0;
}
void OpenALSoundRenderer::SetSfxPaused(bool paused, int slot)
{
int oldslots = SFXPaused;
if(paused)
{
SFXPaused |= 1 << slot;
if(oldslots == 0 && PausableSfx.Size() > 0)
{
alSourcePausev(PausableSfx.Size(), &PausableSfx[0]);
getALError();
PurgeStoppedSources();
}
}
else
{
SFXPaused &= ~(1 << slot);
if(SFXPaused == 0 && oldslots != 0 && PausableSfx.Size() > 0)
{
alSourcePlayv(PausableSfx.Size(), &PausableSfx[0]);
getALError();
}
}
}
void OpenALSoundRenderer::SetInactive(SoundRenderer::EInactiveState state)
{
switch(state)
{
case SoundRenderer::INACTIVE_Active:
alListenerf(AL_GAIN, 1.0f);
if(ALC.SOFT_pause_device)
{
alcDeviceResumeSOFT(Device);
getALCError(Device);
}
break;
case SoundRenderer::INACTIVE_Complete:
if(ALC.SOFT_pause_device)
{
alcDevicePauseSOFT(Device);
getALCError(Device);
}
/* fall-through */
case SoundRenderer::INACTIVE_Mute:
alListenerf(AL_GAIN, 0.0f);
break;
}
}
void OpenALSoundRenderer::Sync(bool sync)
{
if(sync)
{
if(SfxGroup.Size() > 0)
{
alSourcePausev(SfxGroup.Size(), &SfxGroup[0]);
getALError();
PurgeStoppedSources();
}
}
else
{
// Might already be something to handle this; basically, get a vector
// of all values in SfxGroup that are not also in PausableSfx (when
// SFXPaused is non-0).
TArray<ALuint> toplay = SfxGroup;
if(SFXPaused)
{
uint32_t i = 0;
while(i < toplay.Size())
{
uint32_t p = PausableSfx.Find(toplay[i]);
if(p < PausableSfx.Size())
toplay.Delete(i);
else
i++;
}
}
if(toplay.Size() > 0)
{
alSourcePlayv(toplay.Size(), &toplay[0]);
getALError();
}
}
}
void OpenALSoundRenderer::UpdateSoundParams3D(SoundListener *listener, FISoundChannel *chan, bool areasound, const FVector3 &pos, const FVector3 &vel)
{
if(chan == NULL || chan->SysChannel == NULL)
return;
FVector3 dir = pos - listener->position;
chan->DistanceSqr = (float)dir.LengthSquared();
if(chan->ManualRolloff)
{
if(!AL.EXT_SOURCE_RADIUS && areasound &&
chan->DistanceSqr < AREA_SOUND_RADIUS*AREA_SOUND_RADIUS)
{
FVector3 amb(0.f, !(dir.Y>=0.f) ? -1.f : 1.f, 0.f);
float a = sqrtf(chan->DistanceSqr) / AREA_SOUND_RADIUS;
dir = amb + (dir-amb)*a;
}
if(dir.DoesNotApproximatelyEqual(FVector3(0.f, 0.f, 0.f)))
{
float gain = GetRolloff(&chan->Rolloff, sqrtf(chan->DistanceSqr)*chan->DistanceScale);
dir.MakeResize((gain > 0.00001f) ? 1.f/gain : 100000.f);
}
}
else if(!AL.EXT_SOURCE_RADIUS && areasound &&
chan->DistanceSqr < AREA_SOUND_RADIUS*AREA_SOUND_RADIUS)
{
float mindist = chan->Rolloff.MinDistance / chan->DistanceScale;
FVector3 amb(0.f, !(dir.Y>=0.f) ? -mindist : mindist, 0.f);
float a = sqrtf(chan->DistanceSqr) / AREA_SOUND_RADIUS;
dir = amb + (dir-amb)*a;
}
dir += listener->position;
alDeferUpdatesSOFT();
ALuint source = GET_PTRID(chan->SysChannel);
if(chan->DistanceSqr < (0.0004f*0.0004f))
{
alSourcei(source, AL_SOURCE_RELATIVE, AL_TRUE);
alSource3f(source, AL_POSITION, 0.f, 0.f, 0.f);
}
else
{
alSourcei(source, AL_SOURCE_RELATIVE, AL_FALSE);
alSource3f(source, AL_POSITION, dir[0], dir[1], -dir[2]);
}
alSource3f(source, AL_VELOCITY, vel[0], vel[1], -vel[2]);
getALError();
}
void OpenALSoundRenderer::UpdateListener(SoundListener *listener)
{
if(!listener->valid)
return;
alDeferUpdatesSOFT();
float angle = listener->angle;
ALfloat orient[6];
// forward
orient[0] = cosf(angle);
orient[1] = 0.f;
orient[2] = -sinf(angle);
// up
orient[3] = 0.f;
orient[4] = 1.f;
orient[5] = 0.f;
alListenerfv(AL_ORIENTATION, orient);
alListener3f(AL_POSITION, listener->position.X,
listener->position.Y,
-listener->position.Z);
alListener3f(AL_VELOCITY, listener->velocity.X,
listener->velocity.Y,
-listener->velocity.Z);
getALError();
const ReverbContainer *env = ForcedEnvironment;
if(!env)
{
env = listener->Environment;
if(!env)
env = DefaultEnvironments[0];
}
if(env != PrevEnvironment || env->Modified)
{
PrevEnvironment = env;
DPrintf(DMSG_NOTIFY, "Reverb Environment %s\n", env->Name);
if(EnvSlot != 0)
LoadReverb(env);
const_cast<ReverbContainer*>(env)->Modified = false;
}
// NOTE: Moving into and out of water will undo pitch variations on sounds.
if(listener->underwater || env->SoftwareWater)
{
if(!WasInWater)
{
WasInWater = true;
if(EnvSlot != 0 && *snd_waterreverb)
{
// Find the "Underwater" reverb environment
env = S_FindEnvironment(0x1600);
LoadReverb(env ? env : DefaultEnvironments[0]);
alFilterf(EnvFilters[0], AL_LOWPASS_GAIN, 1.f);
alFilterf(EnvFilters[0], AL_LOWPASS_GAINHF, 0.125f);
alFilterf(EnvFilters[1], AL_LOWPASS_GAIN, 1.f);
alFilterf(EnvFilters[1], AL_LOWPASS_GAINHF, 1.f);
// Apply the updated filters on the sources
FSoundChan *schan = soundEngine->GetChannels();
while (schan)
{
ALuint source = GET_PTRID(schan->SysChannel);
if (source && !(schan->ChanFlags & CHANF_UI))
{
alSourcei(source, AL_DIRECT_FILTER, EnvFilters[0]);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]);
}
schan = schan->NextChan;
}
}
FSoundChan *schan = soundEngine->GetChannels();
while (schan)
{
ALuint source = GET_PTRID(schan->SysChannel);
if (source && !(schan->ChanFlags & CHANF_UI))
alSourcef(source, AL_PITCH, schan->Pitch / 128.0f * PITCH_MULT);
schan = schan->NextChan;
}
getALError();
}
}
else if(WasInWater)
{
WasInWater = false;
if(EnvSlot != 0)
{
LoadReverb(env);
alFilterf(EnvFilters[0], AL_LOWPASS_GAIN, 1.f);
alFilterf(EnvFilters[0], AL_LOWPASS_GAINHF, 1.f);
alFilterf(EnvFilters[1], AL_LOWPASS_GAIN, 1.f);
alFilterf(EnvFilters[1], AL_LOWPASS_GAINHF, 1.f);
FSoundChan *schan = soundEngine->GetChannels();
while (schan)
{
ALuint source = GET_PTRID(schan->SysChannel);
if (source && !(schan->ChanFlags & CHANF_UI))
{
alSourcei(source, AL_DIRECT_FILTER, EnvFilters[0]);
alSource3i(source, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]);
}
schan = schan->NextChan;
}
}
FSoundChan *schan = soundEngine->GetChannels();
while (schan)
{
ALuint source = GET_PTRID(schan->SysChannel);
if (source && !(schan->ChanFlags & CHANF_UI))
alSourcef(source, AL_PITCH, schan->Pitch / 128.0f);
schan = schan->NextChan;
}
getALError();
}
}
void OpenALSoundRenderer::UpdateSounds()
{
alProcessUpdatesSOFT();
if(!FadingSources.empty())
{
auto cur_time = std::chrono::steady_clock::now().time_since_epoch();
for(auto iter = FadingSources.begin();iter != FadingSources.end();)
{
auto time_diff = std::chrono::duration_cast<std::chrono::milliseconds>(cur_time -
std::chrono::steady_clock::time_point::duration(iter->second));
if(time_diff.count() >= UpdateTimeMS)
{
FreeSource(iter->first);
iter = FadingSources.erase(iter);
}
else
++iter;
}
}
if(ALC.EXT_disconnect)
{
ALCint connected = ALC_TRUE;
alcGetIntegerv(Device, ALC_CONNECTED, 1, &connected);
if(connected == ALC_FALSE)
{
Printf("Sound device disconnected; restarting...\n");
soundEngine->Reset();
return;
}
}
PurgeStoppedSources();
}
bool OpenALSoundRenderer::IsValid()
{
return Device != NULL;
}
void OpenALSoundRenderer::MarkStartTime(FISoundChannel *chan)
{
// FIXME: Get current time (preferably from the audio clock, but the system
// time will have to do)
chan->StartTime = std::chrono::steady_clock::now().time_since_epoch().count();
}
float OpenALSoundRenderer::GetAudibility(FISoundChannel *chan)
{
if(chan == NULL || chan->SysChannel == NULL)
return 0.f;
ALuint source = GET_PTRID(chan->SysChannel);
ALfloat volume = 0.f;
alGetSourcef(source, AL_GAIN, &volume);
getALError();
volume *= GetRolloff(&chan->Rolloff, sqrtf(chan->DistanceSqr) * chan->DistanceScale);
return volume;
}
void OpenALSoundRenderer::PrintStatus()
{
Printf("Output device: " TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_DEVICE_SPECIFIER));
getALCError(Device);
ALCint frequency, major, minor, mono, stereo;
alcGetIntegerv(Device, ALC_FREQUENCY, 1, &frequency);
alcGetIntegerv(Device, ALC_MAJOR_VERSION, 1, &major);
alcGetIntegerv(Device, ALC_MINOR_VERSION, 1, &minor);
alcGetIntegerv(Device, ALC_MONO_SOURCES, 1, &mono);
alcGetIntegerv(Device, ALC_STEREO_SOURCES, 1, &stereo);
if(getALCError(Device) == AL_NO_ERROR)
{
Printf("Device sample rate: " TEXTCOLOR_BLUE"%d" TEXTCOLOR_NORMAL"hz\n", frequency);
Printf("ALC Version: " TEXTCOLOR_BLUE"%d.%d\n", major, minor);
Printf("ALC Extensions: " TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_EXTENSIONS));
Printf("Available sources: " TEXTCOLOR_BLUE"%d" TEXTCOLOR_NORMAL" (" TEXTCOLOR_BLUE"%d" TEXTCOLOR_NORMAL" mono, " TEXTCOLOR_BLUE"%d" TEXTCOLOR_NORMAL" stereo)\n", mono+stereo, mono, stereo);
}
if(!alcIsExtensionPresent(Device, "ALC_EXT_EFX"))
Printf("EFX not found\n");
else
{
ALCint sends;
alcGetIntegerv(Device, ALC_EFX_MAJOR_VERSION, 1, &major);
alcGetIntegerv(Device, ALC_EFX_MINOR_VERSION, 1, &minor);
alcGetIntegerv(Device, ALC_MAX_AUXILIARY_SENDS, 1, &sends);
if(getALCError(Device) == AL_NO_ERROR)
{
Printf("EFX Version: " TEXTCOLOR_BLUE"%d.%d\n", major, minor);
Printf("Auxiliary sends: " TEXTCOLOR_BLUE"%d\n", sends);
}
}
Printf("Vendor: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VENDOR));
Printf("Renderer: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_RENDERER));
Printf("Version: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VERSION));
Printf("Extensions: " TEXTCOLOR_ORANGE"%s\n", alGetString(AL_EXTENSIONS));
getALError();
}
FString OpenALSoundRenderer::GatherStats()
{
FString out;
ALCint refresh = 1;
alcGetIntegerv(Device, ALC_REFRESH, 1, &refresh);
getALCError(Device);
uint32_t total = Sources.Size();
uint32_t used = SfxGroup.Size()+Streams.Size();
uint32_t unused = FreeSfx.Size();
out.Format("%u sources (" TEXTCOLOR_YELLOW"%u" TEXTCOLOR_NORMAL" active, " TEXTCOLOR_YELLOW"%u" TEXTCOLOR_NORMAL" free), Update interval: " TEXTCOLOR_YELLOW"%.1f" TEXTCOLOR_NORMAL"ms",
total, used, unused, 1000.f/static_cast<float>(refresh));
return out;
}
void OpenALSoundRenderer::PrintDriversList()
{
const ALCchar *drivers = (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") ?
alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER) :
alcGetString(NULL, ALC_DEVICE_SPECIFIER));
if(drivers == NULL)
{
Printf(TEXTCOLOR_YELLOW"Failed to retrieve device list: %s\n", alcGetString(NULL, alcGetError(NULL)));
return;
}
const ALCchar *current = NULL;
if(alcIsExtensionPresent(Device, "ALC_ENUMERATE_ALL_EXT"))
current = alcGetString(Device, ALC_ALL_DEVICES_SPECIFIER);
if(alcGetError(Device) != ALC_NO_ERROR || !current)
current = alcGetString(Device, ALC_DEVICE_SPECIFIER);
if(current == NULL)
{
Printf(TEXTCOLOR_YELLOW"Failed to retrieve device name: %s\n", alcGetString(Device, alcGetError(Device)));
return;
}
Printf("%c%s%2d. %s\n", ' ', ((strcmp(snd_aldevice, "Default") == 0) ? TEXTCOLOR_BOLD : ""), 0,
"Default");
for(int i = 1;*drivers;i++)
{
Printf("%c%s%2d. %s\n", ((strcmp(current, drivers)==0) ? '*' : ' '),
((strcmp(*snd_aldevice, drivers)==0) ? TEXTCOLOR_BOLD : ""), i,
drivers);
drivers += strlen(drivers)+1;
}
}
void OpenALSoundRenderer::PurgeStoppedSources()
{
// Release channels that are stopped
for(uint32_t i = 0;i < SfxGroup.Size();++i)
{
ALuint src = SfxGroup[i];
ALint state = AL_INITIAL;
alGetSourcei(src, AL_SOURCE_STATE, &state);
if(state == AL_INITIAL || state == AL_PLAYING || state == AL_PAUSED)
continue;
FSoundChan *schan = soundEngine->GetChannels();
while(schan)
{
if(schan->SysChannel != NULL && src == GET_PTRID(schan->SysChannel))
{
ForceStopChannel(schan);
break;
}
schan = schan->NextChan;
}
}
getALError();
}
void OpenALSoundRenderer::LoadReverb(const ReverbContainer *env)
{
ALuint *envReverb = EnvEffects.CheckKey(env->ID);
bool doLoad = (env->Modified || !envReverb);
if(!envReverb)
{
bool ok = false;
envReverb = &EnvEffects.Insert(env->ID, 0);
alGenEffects(1, envReverb);
if(getALError() == AL_NO_ERROR)
{
alEffecti(*envReverb, AL_EFFECT_TYPE, AL_EFFECT_EAXREVERB);
ok = (alGetError() == AL_NO_ERROR);
if(!ok)
{
alEffecti(*envReverb, AL_EFFECT_TYPE, AL_EFFECT_REVERB);
ok = (alGetError() == AL_NO_ERROR);
}
if(!ok)
{
alEffecti(*envReverb, AL_EFFECT_TYPE, AL_EFFECT_NULL);
ok = (alGetError() == AL_NO_ERROR);
}
if(!ok)
{
alDeleteEffects(1, envReverb);
getALError();
}
}
if(!ok)
{
*envReverb = 0;
doLoad = false;
}
}
if(doLoad)
{
const REVERB_PROPERTIES &props = env->Properties;
ALint type = AL_EFFECT_NULL;
alGetEffecti(*envReverb, AL_EFFECT_TYPE, &type);
#define mB2Gain(x) ((float)pow(10., (x)/2000.))
if(type == AL_EFFECT_EAXREVERB)
{
ALfloat reflectpan[3] = { props.ReflectionsPan0,
props.ReflectionsPan1,
props.ReflectionsPan2 };
ALfloat latepan[3] = { props.ReverbPan0, props.ReverbPan1,
props.ReverbPan2 };
#undef SETPARAM
#define SETPARAM(e,t,v) alEffectf((e), AL_EAXREVERB_##t, clamp((v), AL_EAXREVERB_MIN_##t, AL_EAXREVERB_MAX_##t))
SETPARAM(*envReverb, DIFFUSION, props.EnvDiffusion);
SETPARAM(*envReverb, DENSITY, powf(props.EnvSize, 3.0f) * 0.0625f);
SETPARAM(*envReverb, GAIN, mB2Gain(props.Room));
SETPARAM(*envReverb, GAINHF, mB2Gain(props.RoomHF));
SETPARAM(*envReverb, GAINLF, mB2Gain(props.RoomLF));
SETPARAM(*envReverb, DECAY_TIME, props.DecayTime);
SETPARAM(*envReverb, DECAY_HFRATIO, props.DecayHFRatio);
SETPARAM(*envReverb, DECAY_LFRATIO, props.DecayLFRatio);
SETPARAM(*envReverb, REFLECTIONS_GAIN, mB2Gain(props.Reflections));
SETPARAM(*envReverb, REFLECTIONS_DELAY, props.ReflectionsDelay);
alEffectfv(*envReverb, AL_EAXREVERB_REFLECTIONS_PAN, reflectpan);
SETPARAM(*envReverb, LATE_REVERB_GAIN, mB2Gain(props.Reverb));
SETPARAM(*envReverb, LATE_REVERB_DELAY, props.ReverbDelay);
alEffectfv(*envReverb, AL_EAXREVERB_LATE_REVERB_PAN, latepan);
SETPARAM(*envReverb, ECHO_TIME, props.EchoTime);
SETPARAM(*envReverb, ECHO_DEPTH, props.EchoDepth);
SETPARAM(*envReverb, MODULATION_TIME, props.ModulationTime);
SETPARAM(*envReverb, MODULATION_DEPTH, props.ModulationDepth);
SETPARAM(*envReverb, AIR_ABSORPTION_GAINHF, mB2Gain(props.AirAbsorptionHF));
SETPARAM(*envReverb, HFREFERENCE, props.HFReference);
SETPARAM(*envReverb, LFREFERENCE, props.LFReference);
SETPARAM(*envReverb, ROOM_ROLLOFF_FACTOR, props.RoomRolloffFactor);
alEffecti(*envReverb, AL_EAXREVERB_DECAY_HFLIMIT,
(props.Flags&REVERB_FLAGS_DECAYHFLIMIT)?AL_TRUE:AL_FALSE);
#undef SETPARAM
}
else if(type == AL_EFFECT_REVERB)
{
#define SETPARAM(e,t,v) alEffectf((e), AL_REVERB_##t, clamp((v), AL_REVERB_MIN_##t, AL_REVERB_MAX_##t))
SETPARAM(*envReverb, DIFFUSION, props.EnvDiffusion);
SETPARAM(*envReverb, DENSITY, powf(props.EnvSize, 3.0f) * 0.0625f);
SETPARAM(*envReverb, GAIN, mB2Gain(props.Room));
SETPARAM(*envReverb, GAINHF, mB2Gain(props.RoomHF));
SETPARAM(*envReverb, DECAY_TIME, props.DecayTime);
SETPARAM(*envReverb, DECAY_HFRATIO, props.DecayHFRatio);
SETPARAM(*envReverb, REFLECTIONS_GAIN, mB2Gain(props.Reflections));
SETPARAM(*envReverb, REFLECTIONS_DELAY, props.ReflectionsDelay);
SETPARAM(*envReverb, LATE_REVERB_GAIN, mB2Gain(props.Reverb));
SETPARAM(*envReverb, LATE_REVERB_DELAY, props.ReverbDelay);
SETPARAM(*envReverb, AIR_ABSORPTION_GAINHF, mB2Gain(props.AirAbsorptionHF));
SETPARAM(*envReverb, ROOM_ROLLOFF_FACTOR, props.RoomRolloffFactor);
alEffecti(*envReverb, AL_REVERB_DECAY_HFLIMIT,
(props.Flags&REVERB_FLAGS_DECAYHFLIMIT)?AL_TRUE:AL_FALSE);
#undef SETPARAM
}
#undef mB2Gain
}
alAuxiliaryEffectSloti(EnvSlot, AL_EFFECTSLOT_EFFECT, *envReverb);
getALError();
}
FSoundChan *OpenALSoundRenderer::FindLowestChannel()
{
FSoundChan *schan = soundEngine->GetChannels();
FSoundChan *lowest = NULL;
while(schan)
{
if(schan->SysChannel != NULL)
{
if(!lowest || schan->Priority < lowest->Priority ||
(schan->Priority == lowest->Priority &&
schan->DistanceSqr > lowest->DistanceSqr))
lowest = schan;
}
schan = schan->NextChan;
}
return lowest;
}
#include "menu/menu.h"
void I_BuildALDeviceList(FOptionValues* opt)
{
opt->mValues.Resize(1);
opt->mValues[0].TextValue = "Default";
opt->mValues[0].Text = "Default";
#ifndef NO_OPENAL
if (IsOpenALPresent())
{
const ALCchar* names = (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") ?
alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER) :
alcGetString(NULL, ALC_DEVICE_SPECIFIER));
if (!names)
Printf("Failed to get device list: %s\n", alcGetString(NULL, alcGetError(NULL)));
else while (*names)
{
unsigned int i = opt->mValues.Reserve(1);
opt->mValues[i].TextValue = names;
opt->mValues[i].Text = names;
names += strlen(names) + 1;
}
}
#endif
}
void I_BuildALResamplersList(FOptionValues* opt)
{
opt->mValues.Resize(1);
opt->mValues[0].TextValue = "Default";
opt->mValues[0].Text = "Default";
#ifndef NO_OPENAL
if (!IsOpenALPresent())
return;
if (!alcGetCurrentContext() || !alIsExtensionPresent("AL_SOFT_source_resampler"))
return;
LPALGETSTRINGISOFT alGetStringiSOFT = reinterpret_cast<LPALGETSTRINGISOFT>(alGetProcAddress("alGetStringiSOFT"));
ALint num_resamplers = alGetInteger(AL_NUM_RESAMPLERS_SOFT);
unsigned int idx = opt->mValues.Reserve(num_resamplers);
for (ALint i = 0; i < num_resamplers; ++i)
{
const ALchar* name = alGetStringiSOFT(AL_RESAMPLER_NAME_SOFT, i);
opt->mValues[idx].TextValue = name;
opt->mValues[idx].Text = name;
++idx;
}
#endif
}
#endif // NO_OPENAL
Reference in a new issue View git blame Copy permalink