mirror of
https://github.com/ZDoom/qzdoom-gpl.git
synced 2024-12-12 05:02:11 +00:00
1869 lines
58 KiB
C++
1869 lines
58 KiB
C++
/*
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** oalsound.cpp
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** System interface for sound; uses OpenAL
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**
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**---------------------------------------------------------------------------
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** Copyright 2008-2010 Chris Robinson
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** All rights reserved.
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**
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** Redistribution and use in source and binary forms, with or without
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** modification, are permitted provided that the following conditions
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** are met:
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**
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** 1. Redistributions of source code must retain the above copyright
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** notice, this list of conditions and the following disclaimer.
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** 2. Redistributions in binary form must reproduce the above copyright
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** notice, this list of conditions and the following disclaimer in the
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** documentation and/or other materials provided with the distribution.
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** 3. The name of the author may not be used to endorse or promote products
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** derived from this software without specific prior written permission.
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**
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** THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
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** IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
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** OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
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** IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
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** INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
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** NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
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** DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
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** THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
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** (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
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** THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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**---------------------------------------------------------------------------
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**
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*/
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#ifdef _WIN32
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#define WIN32_LEAN_AND_MEAN
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#include <windows.h>
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#define USE_WINDOWS_DWORD
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#endif
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#include "doomstat.h"
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#include "templates.h"
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#include "oalsound.h"
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#include "c_cvars.h"
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#include "c_dispatch.h"
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#include "i_system.h"
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#include "v_text.h"
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#include "gi.h"
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#include "actor.h"
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#include "r_state.h"
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#include "w_wad.h"
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#include "i_music.h"
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#include "i_musicinterns.h"
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#include "tempfiles.h"
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CVAR (String, snd_aldevice, "Default", CVAR_ARCHIVE|CVAR_GLOBALCONFIG)
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CVAR (Bool, snd_efx, true, CVAR_ARCHIVE|CVAR_GLOBALCONFIG)
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void I_BuildALDeviceList(FOptionValues *opt)
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{
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opt->mValues.Resize(1);
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opt->mValues[0].TextValue = "Default";
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opt->mValues[0].Text = "Default";
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#ifndef NO_OPENAL
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const ALCchar *names = (alcIsExtensionPresent(NULL, "ALC_ENUMERATE_ALL_EXT") ?
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alcGetString(NULL, ALC_ALL_DEVICES_SPECIFIER) :
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alcGetString(NULL, ALC_DEVICE_SPECIFIER));
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if(!names)
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Printf("Failed to get device list: %s\n", alcGetString(NULL, alcGetError(NULL)));
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else while(*names)
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{
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unsigned int i = opt->mValues.Reserve(1);
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opt->mValues[i].TextValue = names;
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opt->mValues[i].Text = names;
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names += strlen(names)+1;
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}
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#endif
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}
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#ifndef NO_OPENAL
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#include <algorithm>
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#include <memory>
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#include <string>
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#include <vector>
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EXTERN_CVAR (Int, snd_channels)
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EXTERN_CVAR (Int, snd_samplerate)
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EXTERN_CVAR (Bool, snd_waterreverb)
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EXTERN_CVAR (Bool, snd_pitched)
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#define foreach(type, name, vec) \
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for(std::vector<type>::iterator (name) = (vec).begin(), \
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(_end_##name) = (vec).end(); \
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(name) != (_end_##name);(name)++)
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static ALenum checkALError(const char *fn, unsigned int ln)
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{
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ALenum err = alGetError();
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if(err != AL_NO_ERROR)
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{
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if(strchr(fn, '/'))
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fn = strrchr(fn, '/')+1;
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else if(strchr(fn, '\\'))
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fn = strrchr(fn, '\\')+1;
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Printf(">>>>>>>>>>>> Received AL error %s (%#x), %s:%u\n", alGetString(err), err, fn, ln);
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}
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return err;
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}
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#define getALError() checkALError(__FILE__, __LINE__)
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static ALCenum checkALCError(ALCdevice *device, const char *fn, unsigned int ln)
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{
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ALCenum err = alcGetError(device);
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if(err != ALC_NO_ERROR)
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{
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if(strchr(fn, '/'))
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fn = strrchr(fn, '/')+1;
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else if(strchr(fn, '\\'))
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fn = strrchr(fn, '\\')+1;
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Printf(">>>>>>>>>>>> Received ALC error %s (%#x), %s:%u\n", alcGetString(device, err), err, fn, ln);
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}
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return err;
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}
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#define getALCError(d) checkALCError((d), __FILE__, __LINE__)
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class OpenALSoundStream : public SoundStream
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{
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OpenALSoundRenderer *Renderer;
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SoundStreamCallback Callback;
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void *UserData;
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std::vector<ALubyte> Data;
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ALsizei SampleRate;
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ALenum Format;
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static const int BufferCount = 4;
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ALuint Buffers[BufferCount];
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ALuint Source;
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bool Playing;
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bool Looping;
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ALfloat Volume;
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std::vector<BYTE> DecoderData;
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std::auto_ptr<SoundDecoder> Decoder;
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static bool DecoderCallback(SoundStream *_sstream, void *ptr, int length, void *user)
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{
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OpenALSoundStream *self = static_cast<OpenALSoundStream*>(_sstream);
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if(length < 0) return false;
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size_t got = self->Decoder->read((char*)ptr, length);
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if(got < (unsigned int)length)
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{
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if(!self->Looping || !self->Decoder->seek(0))
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return false;
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got += self->Decoder->read((char*)ptr+got, length-got);
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}
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return (got == (unsigned int)length);
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}
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bool SetupSource()
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{
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/* Get a source, killing the farthest, lowest-priority sound if needed */
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if(Renderer->FreeSfx.size() == 0)
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{
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FSoundChan *lowest = Renderer->FindLowestChannel();
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if(lowest) Renderer->StopChannel(lowest);
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if(Renderer->FreeSfx.size() == 0)
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return false;
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}
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Source = Renderer->FreeSfx.back();
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Renderer->FreeSfx.pop_back();
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/* Set the default properties for localized playback */
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alSource3f(Source, AL_DIRECTION, 0.f, 0.f, 0.f);
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alSource3f(Source, AL_VELOCITY, 0.f, 0.f, 0.f);
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alSource3f(Source, AL_POSITION, 0.f, 0.f, 0.f);
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alSourcef(Source, AL_MAX_GAIN, 1.f);
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alSourcef(Source, AL_GAIN, 1.f);
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alSourcef(Source, AL_PITCH, 1.f);
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alSourcef(Source, AL_ROLLOFF_FACTOR, 0.f);
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alSourcef(Source, AL_SEC_OFFSET, 0.f);
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alSourcei(Source, AL_SOURCE_RELATIVE, AL_TRUE);
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alSourcei(Source, AL_LOOPING, AL_FALSE);
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if(Renderer->EnvSlot)
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{
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alSourcef(Source, AL_ROOM_ROLLOFF_FACTOR, 0.f);
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alSourcef(Source, AL_AIR_ABSORPTION_FACTOR, 0.f);
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alSourcei(Source, AL_DIRECT_FILTER, AL_FILTER_NULL);
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alSource3i(Source, AL_AUXILIARY_SEND_FILTER, 0, 0, AL_FILTER_NULL);
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}
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alGenBuffers(BufferCount, Buffers);
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return (getALError() == AL_NO_ERROR);
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}
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public:
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OpenALSoundStream(OpenALSoundRenderer *renderer)
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: Renderer(renderer), Source(0), Playing(false), Looping(false), Volume(1.0f)
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{
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Renderer->Streams.push_back(this);
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memset(Buffers, 0, sizeof(Buffers));
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}
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virtual ~OpenALSoundStream()
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{
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if(Source)
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{
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alSourceRewind(Source);
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alSourcei(Source, AL_BUFFER, 0);
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Renderer->FreeSfx.push_back(Source);
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Source = 0;
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}
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if(Buffers[0])
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{
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alDeleteBuffers(BufferCount, &Buffers[0]);
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memset(Buffers, 0, sizeof(Buffers));
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}
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getALError();
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Renderer->Streams.erase(std::find(Renderer->Streams.begin(),
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Renderer->Streams.end(), this));
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Renderer = NULL;
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}
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virtual bool Play(bool loop, float vol)
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{
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SetVolume(vol);
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if(Playing)
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return true;
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/* Clear the buffer queue, then fill and queue each buffer */
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alSourcei(Source, AL_BUFFER, 0);
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for(int i = 0;i < BufferCount;i++)
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{
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if(!Callback(this, &Data[0], Data.size(), UserData))
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{
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if(i == 0)
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return false;
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break;
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}
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alBufferData(Buffers[i], Format, &Data[0], Data.size(), SampleRate);
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alSourceQueueBuffers(Source, 1, &Buffers[i]);
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}
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if(getALError() != AL_NO_ERROR)
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return false;
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alSourcePlay(Source);
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Playing = (getALError()==AL_NO_ERROR);
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return Playing;
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}
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virtual void Stop()
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{
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if(!Playing)
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return;
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alSourceStop(Source);
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alSourcei(Source, AL_BUFFER, 0);
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getALError();
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Playing = false;
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}
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virtual void SetVolume(float vol)
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{
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if(vol >= 0.0f) Volume = vol;
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alSourcef(Source, AL_GAIN, Renderer->MusicVolume*Volume);
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getALError();
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}
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virtual bool SetPaused(bool pause)
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{
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if(pause)
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alSourcePause(Source);
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else
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alSourcePlay(Source);
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return (getALError()==AL_NO_ERROR);
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}
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virtual unsigned int GetPosition()
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{ return 0; }
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virtual bool IsEnded()
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{
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if(!Playing)
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return true;
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ALint state, processed;
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alGetSourcei(Source, AL_SOURCE_STATE, &state);
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alGetSourcei(Source, AL_BUFFERS_PROCESSED, &processed);
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Playing = (getALError()==AL_NO_ERROR);
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if(!Playing)
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return true;
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// For each processed buffer in the queue...
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while(processed > 0)
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{
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ALuint bufid;
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// Unqueue the oldest buffer, fill it with more data, and queue it
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// on the end
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alSourceUnqueueBuffers(Source, 1, &bufid);
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processed--;
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if(Callback(this, &Data[0], Data.size(), UserData))
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{
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alBufferData(bufid, Format, &Data[0], Data.size(), SampleRate);
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alSourceQueueBuffers(Source, 1, &bufid);
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}
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}
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// If the source is not playing or paused, and there are buffers queued,
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// then there was an underrun. Restart the source.
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Playing = (getALError()==AL_NO_ERROR);
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if(Playing && state != AL_PLAYING && state != AL_PAUSED)
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{
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ALint queued = 0;
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alGetSourcei(Source, AL_BUFFERS_QUEUED, &queued);
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Playing = (getALError() == AL_NO_ERROR) && (queued > 0);
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if(Playing)
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{
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alSourcePlay(Source);
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Playing = (getALError()==AL_NO_ERROR);
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}
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}
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return !Playing;
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}
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FString GetStats()
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{
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FString stats;
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ALfloat volume;
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ALint processed;
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ALint queued;
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ALint state;
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ALenum err;
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alGetSourcef(Source, AL_GAIN, &volume);
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alGetSourcei(Source, AL_SOURCE_STATE, &state);
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alGetSourcei(Source, AL_BUFFERS_QUEUED, &queued);
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alGetSourcei(Source, AL_BUFFERS_PROCESSED, &processed);
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if((err=alGetError()) != AL_NO_ERROR)
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{
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stats = "Error getting stats: ";
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stats += alGetString(err);
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return stats;
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}
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stats = (state == AL_INITIAL) ? "Buffering" : (state == AL_STOPPED) ? "Underrun" :
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(state == AL_PLAYING || state == AL_PAUSED) ? "Ready" : "Unknown state";
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stats.AppendFormat(",%3d%% buffered", (queued ? 100-(processed*100/queued) : 0));
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stats.AppendFormat(", %d%%", int(volume * 100));
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if(state == AL_PAUSED)
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stats += ", paused";
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if(state == AL_PLAYING)
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stats += ", playing";
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stats.AppendFormat(", %uHz", SampleRate);
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if(!Playing)
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stats += " XX";
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return stats;
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}
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bool Init(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata)
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{
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if(!SetupSource())
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return false;
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Callback = callback;
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UserData = userdata;
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SampleRate = samplerate;
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Format = AL_NONE;
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if((flags&Bits8)) /* Signed or unsigned? We assume unsigned 8-bit... */
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{
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if((flags&Mono)) Format = AL_FORMAT_MONO8;
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else Format = AL_FORMAT_STEREO8;
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}
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else if(!(flags&(Bits32|Float)))
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{
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if((flags&Mono)) Format = AL_FORMAT_MONO16;
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else Format = AL_FORMAT_STEREO16;
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}
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else if((flags&Float) && alIsExtensionPresent("AL_EXT_FLOAT32"))
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{
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if((flags&Mono)) Format = AL_FORMAT_MONO_FLOAT32;
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else Format = AL_FORMAT_STEREO_FLOAT32;
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}
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if(Format == AL_NONE)
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{
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Printf("Unsupported format: 0x%x\n", flags);
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return false;
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}
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int smpsize = 1;
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if((flags&Bits8))
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smpsize *= 1;
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else if((flags&(Bits32|Float)))
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smpsize *= 4;
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else
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smpsize *= 2;
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if((flags&Mono))
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smpsize *= 1;
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else
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smpsize *= 2;
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buffbytes += smpsize-1;
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buffbytes -= buffbytes%smpsize;
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Data.resize(buffbytes);
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return true;
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}
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bool Init(const char *fname, int offset, int length, bool loop)
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{
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if(!SetupSource())
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return false;
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Decoder.reset(Renderer->CreateDecoder(fname, offset, length));
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if(!Decoder.get()) return false;
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Callback = DecoderCallback;
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UserData = NULL;
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Format = AL_NONE;
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ChannelConfig chans;
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SampleType type;
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int srate;
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Decoder->getInfo(&srate, &chans, &type);
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if(chans == ChannelConfig_Mono)
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{
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if(type == SampleType_UInt8) Format = AL_FORMAT_MONO8;
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if(type == SampleType_Int16) Format = AL_FORMAT_MONO16;
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}
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if(chans == ChannelConfig_Stereo)
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{
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if(type == SampleType_UInt8) Format = AL_FORMAT_STEREO8;
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if(type == SampleType_Int16) Format = AL_FORMAT_STEREO16;
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}
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if(Format == AL_NONE)
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{
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Printf("Unsupported audio format (0x%x / 0x%x)\n", chans, type);
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return false;
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}
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SampleRate = srate;
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Looping = loop;
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Data.resize((size_t)(0.2 * SampleRate) * 4);
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return true;
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}
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bool Init(const BYTE *data, int length, bool loop)
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{
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if(!SetupSource())
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return false;
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DecoderData.insert(DecoderData.end(), data, data+length);
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Decoder.reset(Renderer->CreateDecoder(&DecoderData[0], DecoderData.size()));
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if(!Decoder.get()) return false;
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Callback = DecoderCallback;
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UserData = NULL;
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Format = AL_NONE;
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ChannelConfig chans;
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SampleType type;
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int srate;
|
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|
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Decoder->getInfo(&srate, &chans, &type);
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if(chans == ChannelConfig_Mono)
|
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{
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if(type == SampleType_UInt8) Format = AL_FORMAT_MONO8;
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if(type == SampleType_Int16) Format = AL_FORMAT_MONO16;
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}
|
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if(chans == ChannelConfig_Stereo)
|
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{
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if(type == SampleType_UInt8) Format = AL_FORMAT_STEREO8;
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if(type == SampleType_Int16) Format = AL_FORMAT_STEREO16;
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}
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|
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if(Format == AL_NONE)
|
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{
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Printf("Unsupported audio format (0x%x / 0x%x)\n", chans, type);
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return false;
|
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}
|
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SampleRate = srate;
|
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Looping = loop;
|
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Data.resize((size_t)(0.2 * SampleRate) * 4);
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|
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return true;
|
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}
|
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};
|
|
|
|
|
|
extern ReverbContainer *ForcedEnvironment;
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|
|
#define PITCH_MULT (0.7937005f) /* Approx. 4 semitones lower; what Nash suggested */
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|
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#define PITCH(pitch) (snd_pitched ? (pitch)/128.f : 1.f)
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|
|
|
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static float GetRolloff(const FRolloffInfo *rolloff, float distance)
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{
|
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if(distance <= rolloff->MinDistance)
|
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return 1.f;
|
|
// Logarithmic rolloff has no max distance where it goes silent.
|
|
if(rolloff->RolloffType == ROLLOFF_Log)
|
|
return rolloff->MinDistance /
|
|
(rolloff->MinDistance + rolloff->RolloffFactor*(distance-rolloff->MinDistance));
|
|
if(distance >= rolloff->MaxDistance)
|
|
return 0.f;
|
|
|
|
float volume = (rolloff->MaxDistance - distance) / (rolloff->MaxDistance - rolloff->MinDistance);
|
|
if(rolloff->RolloffType == ROLLOFF_Linear)
|
|
return volume;
|
|
|
|
if(rolloff->RolloffType == ROLLOFF_Custom && S_SoundCurve != NULL)
|
|
return S_SoundCurve[int(S_SoundCurveSize * (1.f - volume))] / 127.f;
|
|
return (powf(10.f, volume) - 1.f) / 9.f;
|
|
}
|
|
|
|
|
|
static ALenum FormatFromDesc(int bits, int channels)
|
|
{
|
|
if(bits == 8)
|
|
{
|
|
if(channels == 1) return AL_FORMAT_MONO8;
|
|
if(channels == 2) return AL_FORMAT_STEREO8;
|
|
if(channels == 4) return AL_FORMAT_QUAD8;
|
|
if(channels == 6) return AL_FORMAT_51CHN8;
|
|
if(channels == 7) return AL_FORMAT_61CHN8;
|
|
if(channels == 8) return AL_FORMAT_71CHN8;
|
|
}
|
|
if(bits == 16)
|
|
{
|
|
if(channels == 1) return AL_FORMAT_MONO16;
|
|
if(channels == 2) return AL_FORMAT_STEREO16;
|
|
if(channels == 4) return AL_FORMAT_QUAD16;
|
|
if(channels == 6) return AL_FORMAT_51CHN16;
|
|
if(channels == 7) return AL_FORMAT_61CHN16;
|
|
if(channels == 8) return AL_FORMAT_71CHN16;
|
|
}
|
|
if(bits == 32)
|
|
{
|
|
if(channels == 1) return AL_FORMAT_MONO_FLOAT32;
|
|
if(channels == 2) return AL_FORMAT_STEREO_FLOAT32;
|
|
if(channels == 4) return AL_FORMAT_QUAD32;
|
|
if(channels == 6) return AL_FORMAT_51CHN32;
|
|
if(channels == 7) return AL_FORMAT_61CHN32;
|
|
if(channels == 8) return AL_FORMAT_71CHN32;
|
|
}
|
|
return AL_NONE;
|
|
}
|
|
|
|
|
|
template<typename T>
|
|
static void LoadALFunc(const char *name, T *x)
|
|
{ *x = reinterpret_cast<T>(alGetProcAddress(name)); }
|
|
|
|
OpenALSoundRenderer::OpenALSoundRenderer()
|
|
: Device(NULL), Context(NULL), SrcDistanceModel(false), SFXPaused(0),
|
|
PrevEnvironment(NULL), EnvSlot(0)
|
|
{
|
|
EnvFilters[0] = EnvFilters[1] = 0;
|
|
|
|
Printf("I_InitSound: Initializing OpenAL\n");
|
|
|
|
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");
|
|
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);
|
|
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(" ALC Version: "TEXTCOLOR_BLUE"%d.%d\n", major, minor);
|
|
DPrintf(" ALC Extensions: "TEXTCOLOR_ORANGE"%s\n", alcGetString(Device, ALC_EXTENSIONS));
|
|
|
|
DisconnectNotify = alcIsExtensionPresent(Device, "ALC_EXT_disconnect");
|
|
|
|
std::vector<ALCint> attribs;
|
|
if(*snd_samplerate > 0)
|
|
{
|
|
attribs.push_back(ALC_FREQUENCY);
|
|
attribs.push_back(*snd_samplerate);
|
|
}
|
|
// Make sure one source is capable of stereo output with the rest doing
|
|
// mono, without running out of voices
|
|
attribs.push_back(ALC_MONO_SOURCES);
|
|
attribs.push_back(std::max<ALCint>(*snd_channels, 2) - 1);
|
|
attribs.push_back(ALC_STEREO_SOURCES);
|
|
attribs.push_back(1);
|
|
// Other attribs..?
|
|
attribs.push_back(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();
|
|
|
|
DPrintf(" Vendor: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VENDOR));
|
|
DPrintf(" Renderer: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_RENDERER));
|
|
DPrintf(" Version: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_VERSION));
|
|
DPrintf(" Extensions: "TEXTCOLOR_ORANGE"%s\n", alGetString(AL_EXTENSIONS));
|
|
|
|
SrcDistanceModel = alIsExtensionPresent("AL_EXT_source_distance_model");
|
|
LoopPoints = alIsExtensionPresent("AL_SOFT_loop_points");
|
|
|
|
alDopplerFactor(0.5f);
|
|
alSpeedOfSound(343.3f * 96.0f);
|
|
alDistanceModel(AL_INVERSE_DISTANCE);
|
|
if(SrcDistanceModel)
|
|
alEnable(AL_SOURCE_DISTANCE_MODEL);
|
|
|
|
ALenum err = getALError();
|
|
if(err != AL_NO_ERROR)
|
|
{
|
|
alcMakeContextCurrent(NULL);
|
|
alcDestroyContext(Context);
|
|
Context = NULL;
|
|
alcCloseDevice(Device);
|
|
Device = NULL;
|
|
return;
|
|
}
|
|
|
|
ALCint numMono=0, numStereo=0;
|
|
alcGetIntegerv(Device, ALC_MONO_SOURCES, 1, &numMono);
|
|
alcGetIntegerv(Device, ALC_STEREO_SOURCES, 1, &numStereo);
|
|
|
|
Sources.resize(std::min<size_t>(std::max<ALCint>(*snd_channels, 2),
|
|
numMono+numStereo));
|
|
for(size_t i = 0;i < Sources.size();i++)
|
|
{
|
|
alGenSources(1, &Sources[i]);
|
|
if(getALError() != AL_NO_ERROR)
|
|
{
|
|
Sources.resize(i);
|
|
break;
|
|
}
|
|
FreeSfx.push_back(Sources[i]);
|
|
}
|
|
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;
|
|
}
|
|
DPrintf(" Allocated "TEXTCOLOR_BLUE"%zu"TEXTCOLOR_NORMAL" sources\n", Sources.size());
|
|
|
|
WasInWater = false;
|
|
if(*snd_efx && alcIsExtensionPresent(Device, "ALC_EXT_EFX"))
|
|
{
|
|
// EFX function pointers
|
|
#define LOAD_FUNC(x) (LoadALFunc(#x, &x))
|
|
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);
|
|
#undef LOAD_FUNC
|
|
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(" EAX Reverb found\n");
|
|
alEffecti(envReverb, AL_EFFECT_TYPE, AL_EFFECT_REVERB);
|
|
if(alGetError() == AL_NO_ERROR)
|
|
DPrintf(" 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(" 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");
|
|
}
|
|
|
|
OpenALSoundRenderer::~OpenALSoundRenderer()
|
|
{
|
|
if(!Device)
|
|
return;
|
|
|
|
while(Streams.size() > 0)
|
|
delete Streams[0];
|
|
|
|
alDeleteSources(Sources.size(), &Sources[0]);
|
|
Sources.clear();
|
|
FreeSfx.clear();
|
|
SfxGroup.clear();
|
|
PausableSfx.clear();
|
|
ReverbSfx.clear();
|
|
|
|
for(EffectMap::iterator i = EnvEffects.begin();i != EnvEffects.end();i++)
|
|
{
|
|
if(i->second)
|
|
alDeleteEffects(1, &(i->second));
|
|
}
|
|
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::SetSfxVolume(float volume)
|
|
{
|
|
SfxVolume = volume;
|
|
|
|
FSoundChan *schan = Channels;
|
|
while(schan)
|
|
{
|
|
if(schan->SysChannel != NULL)
|
|
{
|
|
ALuint source = *((ALuint*)schan->SysChannel);
|
|
volume = SfxVolume;
|
|
|
|
alcSuspendContext(Context);
|
|
alSourcef(source, AL_MAX_GAIN, volume);
|
|
if(schan->ManualGain)
|
|
volume *= GetRolloff(&schan->Rolloff, sqrt(schan->DistanceSqr));
|
|
alSourcef(source, AL_GAIN, volume * schan->Volume);
|
|
}
|
|
schan = schan->NextChan;
|
|
}
|
|
|
|
getALError();
|
|
}
|
|
|
|
void OpenALSoundRenderer::SetMusicVolume(float volume)
|
|
{
|
|
MusicVolume = volume;
|
|
foreach(SoundStream*, i, Streams)
|
|
(*i)->SetVolume(-1.f);
|
|
}
|
|
|
|
unsigned int OpenALSoundRenderer::GetMSLength(SoundHandle sfx)
|
|
{
|
|
if(sfx.data)
|
|
{
|
|
ALuint buffer = *((ALuint*)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 = *((ALuint*)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;
|
|
}
|
|
|
|
|
|
SoundHandle OpenALSoundRenderer::LoadSoundRaw(BYTE *sfxdata, int length, int frequency, int channels, int bits, int loopstart, int loopend)
|
|
{
|
|
SoundHandle retval = { NULL };
|
|
|
|
if(length == 0) return retval;
|
|
|
|
if(bits == -8)
|
|
{
|
|
// Simple signed->unsigned conversion
|
|
for(int i = 0;i < length;i++)
|
|
sfxdata[i] ^= 0x80;
|
|
bits = -bits;
|
|
}
|
|
|
|
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 retval;
|
|
}
|
|
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 retval;
|
|
}
|
|
|
|
if((loopstart > 0 || loopend > 0) && LoopPoints)
|
|
{
|
|
if(loopstart < 0)
|
|
loopstart = 0;
|
|
if(loopend < loopstart)
|
|
loopend = length / (channels*bits/8);
|
|
|
|
ALint loops[2] = { loopstart, loopend };
|
|
DPrintf("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("Loop points not supported!\n");
|
|
warned = true;
|
|
}
|
|
|
|
retval.data = new ALuint(buffer);
|
|
return retval;
|
|
}
|
|
|
|
SoundHandle OpenALSoundRenderer::LoadSound(BYTE *sfxdata, int length)
|
|
{
|
|
SoundHandle retval = { NULL };
|
|
ALenum format = AL_NONE;
|
|
ChannelConfig chans;
|
|
SampleType type;
|
|
int srate;
|
|
|
|
std::auto_ptr<SoundDecoder> decoder(CreateDecoder(sfxdata, length));
|
|
if(!decoder.get()) return retval;
|
|
|
|
decoder->getInfo(&srate, &chans, &type);
|
|
if(chans == ChannelConfig_Mono)
|
|
{
|
|
if(type == SampleType_UInt8) format = AL_FORMAT_MONO8;
|
|
if(type == SampleType_Int16) format = AL_FORMAT_MONO16;
|
|
}
|
|
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 (0x%x / 0x%x)\n", chans, type);
|
|
return retval;
|
|
}
|
|
|
|
std::vector<char> data = decoder->readAll();
|
|
|
|
ALuint buffer = 0;
|
|
alGenBuffers(1, &buffer);
|
|
alBufferData(buffer, format, &data[0], data.size(), srate);
|
|
|
|
ALenum err;
|
|
if((err=getALError()) != AL_NO_ERROR)
|
|
{
|
|
Printf("Failed to buffer data: %s\n", alGetString(err));
|
|
alDeleteBuffers(1, &buffer);
|
|
getALError();
|
|
return retval;
|
|
}
|
|
|
|
retval.data = new ALuint(buffer);
|
|
return retval;
|
|
}
|
|
|
|
void OpenALSoundRenderer::UnloadSound(SoundHandle sfx)
|
|
{
|
|
if(!sfx.data)
|
|
return;
|
|
|
|
FSoundChan *schan = Channels;
|
|
while(schan)
|
|
{
|
|
if(schan->SysChannel)
|
|
{
|
|
ALint bufID = 0;
|
|
alGetSourcei(*((ALuint*)schan->SysChannel), AL_BUFFER, &bufID);
|
|
if(bufID == *((ALint*)sfx.data))
|
|
{
|
|
FSoundChan *next = schan->NextChan;
|
|
StopChannel(schan);
|
|
schan = next;
|
|
continue;
|
|
}
|
|
}
|
|
schan = schan->NextChan;
|
|
}
|
|
|
|
alDeleteBuffers(1, ((ALuint*)sfx.data));
|
|
getALError();
|
|
delete ((ALuint*)sfx.data);
|
|
}
|
|
|
|
short *OpenALSoundRenderer::DecodeSample(int outlen, const void *coded, int sizebytes, ECodecType ctype)
|
|
{
|
|
char *samples = (char*)calloc(1, outlen);
|
|
ChannelConfig chans;
|
|
SampleType type;
|
|
int srate;
|
|
|
|
std::auto_ptr<SoundDecoder> decoder(CreateDecoder((const BYTE*)coded, sizebytes));
|
|
if(!decoder.get()) return (short*)samples;
|
|
|
|
decoder->getInfo(&srate, &chans, &type);
|
|
if(chans != ChannelConfig_Mono || type != SampleType_Int16)
|
|
{
|
|
DPrintf("Sample is not 16-bit mono\n");
|
|
return (short*)samples;
|
|
}
|
|
|
|
decoder->read(samples, outlen);
|
|
return (short*)samples;
|
|
}
|
|
|
|
SoundStream *OpenALSoundRenderer::CreateStream(SoundStreamCallback callback, int buffbytes, int flags, int samplerate, void *userdata)
|
|
{
|
|
std::auto_ptr<OpenALSoundStream> stream(new OpenALSoundStream(this));
|
|
if(!stream->Init(callback, buffbytes, flags, samplerate, userdata))
|
|
return NULL;
|
|
return stream.release();
|
|
}
|
|
|
|
SoundStream *OpenALSoundRenderer::OpenStream(const char *filename, int flags, int offset, int length)
|
|
{
|
|
std::auto_ptr<OpenALSoundStream> stream(new OpenALSoundStream(this));
|
|
|
|
bool loop = (flags&SoundStream::Loop);
|
|
bool ok = ((offset == -1) ? stream->Init((const BYTE*)filename, length, loop) :
|
|
stream->Init(filename, offset, length, loop));
|
|
if(ok == false)
|
|
return NULL;
|
|
|
|
return stream.release();
|
|
}
|
|
|
|
FISoundChannel *OpenALSoundRenderer::StartSound(SoundHandle sfx, float vol, int pitch, int chanflags, FISoundChannel *reuse_chan)
|
|
{
|
|
if(FreeSfx.size() == 0)
|
|
{
|
|
FSoundChan *lowest = FindLowestChannel();
|
|
if(lowest) StopChannel(lowest);
|
|
|
|
if(FreeSfx.size() == 0)
|
|
return NULL;
|
|
}
|
|
|
|
ALuint buffer = *((ALuint*)sfx.data);
|
|
ALuint &source = *std::find(Sources.begin(), Sources.end(), FreeSfx.back());
|
|
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_ROLLOFF_FACTOR, 0.f);
|
|
alSourcef(source, AL_MAX_GAIN, SfxVolume);
|
|
alSourcef(source, AL_GAIN, SfxVolume*vol);
|
|
|
|
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);
|
|
alSourcef(source, AL_PITCH, PITCH(pitch));
|
|
}
|
|
else if(WasInWater && !(chanflags&SNDF_NOREVERB))
|
|
alSourcef(source, AL_PITCH, PITCH(pitch)*PITCH_MULT);
|
|
else
|
|
alSourcef(source, AL_PITCH, PITCH(pitch));
|
|
|
|
if(!reuse_chan)
|
|
alSourcef(source, AL_SEC_OFFSET, 0.f);
|
|
else
|
|
{
|
|
if((chanflags&SNDF_ABSTIME))
|
|
alSourcef(source, AL_SEC_OFFSET, reuse_chan->StartTime.Lo/1000.f);
|
|
else
|
|
{
|
|
// FIXME: set offset based on the current time and the StartTime
|
|
alSourcef(source, AL_SEC_OFFSET, 0.f);
|
|
}
|
|
}
|
|
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_back(source);
|
|
if(!(chanflags&SNDF_NOPAUSE))
|
|
PausableSfx.push_back(source);
|
|
SfxGroup.push_back(source);
|
|
FreeSfx.pop_back();
|
|
|
|
FISoundChannel *chan = reuse_chan;
|
|
if(!chan) chan = S_GetChannel(&source);
|
|
else chan->SysChannel = &source;
|
|
|
|
chan->Rolloff.RolloffType = ROLLOFF_Linear;
|
|
chan->Rolloff.MaxDistance = 1000.f;
|
|
chan->Rolloff.MinDistance = 1.f;
|
|
chan->DistanceScale = 1.f;
|
|
chan->DistanceSqr = 1.f;
|
|
chan->ManualGain = 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 = (pos - listener->position).LengthSquared() *
|
|
distscale*distscale;
|
|
|
|
if(FreeSfx.size() == 0)
|
|
{
|
|
FSoundChan *lowest = FindLowestChannel();
|
|
if(lowest)
|
|
{
|
|
if(lowest->Priority < priority || (lowest->Priority == priority &&
|
|
lowest->DistanceSqr > dist_sqr))
|
|
StopChannel(lowest);
|
|
}
|
|
if(FreeSfx.size() == 0)
|
|
return NULL;
|
|
}
|
|
|
|
float rolloffFactor, gain;
|
|
bool manualGain = true;
|
|
|
|
ALuint buffer = *((ALuint*)sfx.data);
|
|
ALint channels = 1;
|
|
alGetBufferi(buffer, AL_CHANNELS, &channels);
|
|
|
|
ALuint &source = *std::find(Sources.begin(), Sources.end(), FreeSfx.back());
|
|
alSource3f(source, AL_POSITION, pos[0], pos[1], -pos[2]);
|
|
alSource3f(source, AL_VELOCITY, vel[0], vel[1], -vel[2]);
|
|
alSource3f(source, AL_DIRECTION, 0.f, 0.f, 0.f);
|
|
alSourcei(source, AL_SOURCE_RELATIVE, AL_FALSE);
|
|
|
|
alSourcei(source, AL_LOOPING, (chanflags&SNDF_LOOP) ? AL_TRUE : AL_FALSE);
|
|
|
|
// Multi-channel sources won't attenuate in OpenAL, and "area sounds" have
|
|
// special rolloff properties (they have a panning radius of 32 units, but
|
|
// start attenuating at MinDistance).
|
|
if(channels == 1 && !(chanflags&SNDF_AREA))
|
|
{
|
|
if(rolloff->RolloffType == ROLLOFF_Log)
|
|
{
|
|
if(SrcDistanceModel)
|
|
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);
|
|
rolloffFactor = rolloff->RolloffFactor;
|
|
manualGain = false;
|
|
gain = 1.f;
|
|
}
|
|
else if(rolloff->RolloffType == ROLLOFF_Linear && SrcDistanceModel)
|
|
{
|
|
alSourcei(source, AL_DISTANCE_MODEL, AL_LINEAR_DISTANCE);
|
|
alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale);
|
|
alSourcef(source, AL_MAX_DISTANCE, rolloff->MaxDistance/distscale);
|
|
rolloffFactor = 1.f;
|
|
manualGain = false;
|
|
gain = 1.f;
|
|
}
|
|
}
|
|
if(manualGain)
|
|
{
|
|
if(SrcDistanceModel)
|
|
alSourcei(source, AL_DISTANCE_MODEL, AL_NONE);
|
|
if((chanflags&SNDF_AREA) && rolloff->MinDistance < 32.f)
|
|
alSourcef(source, AL_REFERENCE_DISTANCE, 32.f/distscale);
|
|
else
|
|
alSourcef(source, AL_REFERENCE_DISTANCE, rolloff->MinDistance/distscale);
|
|
alSourcef(source, AL_MAX_DISTANCE, (1000.f+rolloff->MinDistance)/distscale);
|
|
rolloffFactor = 0.f;
|
|
gain = GetRolloff(rolloff, sqrt(dist_sqr));
|
|
}
|
|
alSourcef(source, AL_ROLLOFF_FACTOR, rolloffFactor);
|
|
alSourcef(source, AL_MAX_GAIN, SfxVolume);
|
|
alSourcef(source, AL_GAIN, SfxVolume * gain);
|
|
|
|
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, rolloffFactor);
|
|
alSourcef(source, AL_PITCH, PITCH(pitch));
|
|
}
|
|
else if(WasInWater && !(chanflags&SNDF_NOREVERB))
|
|
alSourcef(source, AL_PITCH, PITCH(pitch)*PITCH_MULT);
|
|
else
|
|
alSourcef(source, AL_PITCH, PITCH(pitch));
|
|
|
|
if(!reuse_chan)
|
|
alSourcef(source, AL_SEC_OFFSET, 0.f);
|
|
else
|
|
{
|
|
if((chanflags&SNDF_ABSTIME))
|
|
alSourcef(source, AL_SEC_OFFSET, reuse_chan->StartTime.Lo/1000.f);
|
|
else
|
|
{
|
|
// FIXME: set offset based on the current time and the StartTime
|
|
alSourcef(source, AL_SAMPLE_OFFSET, 0.f);
|
|
}
|
|
}
|
|
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_back(source);
|
|
if(!(chanflags&SNDF_NOPAUSE))
|
|
PausableSfx.push_back(source);
|
|
SfxGroup.push_back(source);
|
|
FreeSfx.pop_back();
|
|
|
|
FISoundChannel *chan = reuse_chan;
|
|
if(!chan) chan = S_GetChannel(&source);
|
|
else chan->SysChannel = &source;
|
|
|
|
chan->Rolloff = *rolloff;
|
|
chan->DistanceScale = distscale;
|
|
chan->DistanceSqr = dist_sqr;
|
|
chan->ManualGain = manualGain;
|
|
|
|
return chan;
|
|
}
|
|
|
|
void OpenALSoundRenderer::ChannelVolume(FISoundChannel *chan, float volume)
|
|
{
|
|
if(chan == NULL || chan->SysChannel == NULL)
|
|
return;
|
|
|
|
alcSuspendContext(Context);
|
|
|
|
ALuint source = *((ALuint*)chan->SysChannel);
|
|
|
|
if(chan->ManualGain)
|
|
volume *= GetRolloff(&chan->Rolloff, sqrt(chan->DistanceSqr));
|
|
alSourcef(source, AL_GAIN, SfxVolume * volume);
|
|
}
|
|
|
|
void OpenALSoundRenderer::StopChannel(FISoundChannel *chan)
|
|
{
|
|
if(chan == NULL || chan->SysChannel == NULL)
|
|
return;
|
|
|
|
ALuint source = *((ALuint*)chan->SysChannel);
|
|
// Release first, so it can be properly marked as evicted if it's being
|
|
// forcefully killed
|
|
S_ChannelEnded(chan);
|
|
|
|
alSourceRewind(source);
|
|
alSourcei(source, AL_BUFFER, 0);
|
|
getALError();
|
|
|
|
std::vector<ALuint>::iterator i;
|
|
|
|
i = std::find(PausableSfx.begin(), PausableSfx.end(), source);
|
|
if(i != PausableSfx.end()) PausableSfx.erase(i);
|
|
i = std::find(ReverbSfx.begin(), ReverbSfx.end(), source);
|
|
if(i != ReverbSfx.end()) ReverbSfx.erase(i);
|
|
|
|
SfxGroup.erase(std::find(SfxGroup.begin(), SfxGroup.end(), source));
|
|
FreeSfx.push_back(source);
|
|
}
|
|
|
|
unsigned int OpenALSoundRenderer::GetPosition(FISoundChannel *chan)
|
|
{
|
|
if(chan == NULL || chan->SysChannel == NULL)
|
|
return 0;
|
|
|
|
ALint pos;
|
|
alGetSourcei(*((ALuint*)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)
|
|
{
|
|
}
|
|
|
|
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).
|
|
std::vector<ALuint> toplay = SfxGroup;
|
|
if(SFXPaused)
|
|
{
|
|
std::vector<ALuint>::iterator i = toplay.begin();
|
|
while(i != toplay.end())
|
|
{
|
|
if(std::find(PausableSfx.begin(), PausableSfx.end(), *i) != PausableSfx.end())
|
|
i = toplay.erase(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;
|
|
|
|
alcSuspendContext(Context);
|
|
|
|
ALuint source = *((ALuint*)chan->SysChannel);
|
|
alSource3f(source, AL_POSITION, pos[0], pos[1], -pos[2]);
|
|
alSource3f(source, AL_VELOCITY, vel[0], vel[1], -vel[2]);
|
|
|
|
chan->DistanceSqr = (pos - listener->position).LengthSquared() *
|
|
chan->DistanceScale*chan->DistanceScale;
|
|
// Not all sources can use the distance models provided by OpenAL.
|
|
// For the ones that can't, apply the calculated attenuation as the
|
|
// source gain. Positions still handle the panning,
|
|
if(chan->ManualGain)
|
|
{
|
|
float gain = GetRolloff(&chan->Rolloff, sqrt(chan->DistanceSqr));
|
|
alSourcef(source, AL_GAIN, SfxVolume*gain*((FSoundChan*)chan)->Volume);
|
|
}
|
|
|
|
getALError();
|
|
}
|
|
|
|
void OpenALSoundRenderer::UpdateListener(SoundListener *listener)
|
|
{
|
|
if(!listener->valid)
|
|
return;
|
|
|
|
alcSuspendContext(Context);
|
|
|
|
float angle = listener->angle;
|
|
ALfloat orient[6];
|
|
// forward
|
|
orient[0] = cos(angle);
|
|
orient[1] = 0.f;
|
|
orient[2] = -sin(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("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 either snd_waterreverb or EFX are disabled.
|
|
if(listener->underwater || env->SoftwareWater)
|
|
{
|
|
if(!WasInWater)
|
|
{
|
|
WasInWater = true;
|
|
|
|
if(EnvSlot != 0 && *snd_waterreverb)
|
|
{
|
|
// Find the "Underwater" reverb environment
|
|
env = Environments;
|
|
while(env && env->ID != 0x1600)
|
|
env = env->Next;
|
|
LoadReverb(env ? env : DefaultEnvironments[0]);
|
|
|
|
alFilterf(EnvFilters[0], AL_LOWPASS_GAIN, 0.1f);
|
|
alFilterf(EnvFilters[0], AL_LOWPASS_GAINHF, 1.f);
|
|
alFilterf(EnvFilters[1], AL_LOWPASS_GAIN, 1.f);
|
|
alFilterf(EnvFilters[1], AL_LOWPASS_GAINHF, 1.f);
|
|
|
|
// Apply the updated filters on the sources
|
|
foreach(ALuint, i, ReverbSfx)
|
|
{
|
|
alSourcei(*i, AL_DIRECT_FILTER, EnvFilters[0]);
|
|
alSource3i(*i, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
foreach(ALuint, i, ReverbSfx)
|
|
alSourcef(*i, AL_PITCH, PITCH_MULT);
|
|
}
|
|
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);
|
|
foreach(ALuint, i, ReverbSfx)
|
|
{
|
|
alSourcei(*i, AL_DIRECT_FILTER, EnvFilters[0]);
|
|
alSource3i(*i, AL_AUXILIARY_SEND_FILTER, EnvSlot, 0, EnvFilters[1]);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
foreach(ALuint, i, ReverbSfx)
|
|
alSourcef(*i, AL_PITCH, 1.f);
|
|
}
|
|
getALError();
|
|
}
|
|
}
|
|
|
|
void OpenALSoundRenderer::UpdateSounds()
|
|
{
|
|
alcProcessContext(Context);
|
|
|
|
// For some reason this isn't being called?
|
|
foreach(SoundStream*, stream, Streams)
|
|
(*stream)->IsEnded();
|
|
|
|
if(DisconnectNotify)
|
|
{
|
|
ALCint connected = ALC_TRUE;
|
|
alcGetIntegerv(Device, ALC_CONNECTED, 1, &connected);
|
|
if(connected == ALC_FALSE)
|
|
{
|
|
Printf("Sound device disconnected; restarting...\n");
|
|
static char snd_reset[] = "snd_reset";
|
|
AddCommandString(snd_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.AsOne = 0;
|
|
}
|
|
|
|
float OpenALSoundRenderer::GetAudibility(FISoundChannel *chan)
|
|
{
|
|
if(chan == NULL || chan->SysChannel == NULL)
|
|
return 0.f;
|
|
|
|
ALuint source = *((ALuint*)chan->SysChannel);
|
|
ALfloat volume = 0.f;
|
|
|
|
if(!chan->ManualGain)
|
|
volume = SfxVolume * ((FSoundChan*)chan)->Volume *
|
|
GetRolloff(&chan->Rolloff, sqrt(chan->DistanceSqr));
|
|
else
|
|
{
|
|
alGetSourcef(source, AL_GAIN, &volume);
|
|
getALError();
|
|
}
|
|
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()
|
|
{
|
|
ALCint updates = 1;
|
|
alcGetIntegerv(Device, ALC_REFRESH, 1, &updates);
|
|
getALCError(Device);
|
|
|
|
ALuint total = Sources.size();
|
|
ALuint used = SfxGroup.size()+Streams.size();
|
|
ALuint unused = FreeSfx.size();
|
|
FString out;
|
|
out.Format("%u sources ("TEXTCOLOR_YELLOW"%u"TEXTCOLOR_NORMAL" active, "TEXTCOLOR_YELLOW"%u"TEXTCOLOR_NORMAL" free), Update interval: "TEXTCOLOR_YELLOW"%d"TEXTCOLOR_NORMAL"ms",
|
|
total, used, unused, 1000/updates);
|
|
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
|
|
foreach(ALuint, i, SfxGroup)
|
|
{
|
|
ALint state = AL_PLAYING;
|
|
alGetSourcei(*i, AL_SOURCE_STATE, &state);
|
|
if(state == AL_PLAYING || state == AL_PAUSED)
|
|
continue;
|
|
|
|
FSoundChan *schan = Channels;
|
|
while(schan)
|
|
{
|
|
if(schan->SysChannel != NULL && *i == *((ALuint*)schan->SysChannel))
|
|
{
|
|
StopChannel(schan);
|
|
break;
|
|
}
|
|
schan = schan->NextChan;
|
|
}
|
|
}
|
|
getALError();
|
|
}
|
|
|
|
void OpenALSoundRenderer::LoadReverb(const ReverbContainer *env)
|
|
{
|
|
ALuint &envReverb = EnvEffects[env->ID];
|
|
bool doLoad = (env->Modified || !envReverb);
|
|
|
|
if(!envReverb)
|
|
{
|
|
bool ok = false;
|
|
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, DENSITY, props.Density/100.f);
|
|
SETPARAM(envReverb, DIFFUSION, props.Diffusion/100.f);
|
|
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, DENSITY, props.Density/100.f);
|
|
SETPARAM(envReverb, DIFFUSION, props.Diffusion/100.f);
|
|
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 = Channels;
|
|
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;
|
|
}
|
|
|
|
#endif // NO_OPENAL
|