Add OpenAL-Soft 1.15.1.

2026-08-19 20:20:11 +02:00 · 2013-09-25 20:36:51 +02:00
parent 4c59d385ef
commit 00c0af3b7c
85 changed files with 43429 additions and 0 deletions
@@ -0,0 +1,204 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 2009 by Chris Robinson.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ *  Boston, MA  02111-1307, USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "alMain.h"
+#include "alFilter.h"
+#include "alAuxEffectSlot.h"
+#include "alError.h"
+#include "alu.h"
+
+
+typedef struct ALmodulatorState {
+    // Must be first in all effects!
+    ALeffectState state;
+
+    enum {
+        SINUSOID,
+        SAWTOOTH,
+        SQUARE
+    } Waveform;
+
+    ALuint index;
+    ALuint step;
+
+    ALfloat Gain[MaxChannels];
+
+    FILTER iirFilter;
+    ALfloat history[1];
+} ALmodulatorState;
+
+#define WAVEFORM_FRACBITS  16
+#define WAVEFORM_FRACONE   (1<<WAVEFORM_FRACBITS)
+#define WAVEFORM_FRACMASK  (WAVEFORM_FRACONE-1)
+
+static __inline ALfloat Sin(ALuint index)
+{
+    return sinf(index * (F_PI*2.0f / WAVEFORM_FRACONE));
+}
+
+static __inline ALfloat Saw(ALuint index)
+{
+    return index*(2.0f/WAVEFORM_FRACONE) - 1.0f;
+}
+
+static __inline ALfloat Square(ALuint index)
+{
+    return ((index>>(WAVEFORM_FRACBITS-1))&1)*2.0f - 1.0f;
+}
+
+
+static __inline ALfloat hpFilter1P(FILTER *iir, ALuint offset, ALfloat input)
+{
+    ALfloat *history = &iir->history[offset];
+    ALfloat a = iir->coeff;
+    ALfloat output = input;
+
+    output = output + (history[0]-output)*a;
+    history[0] = output;
+
+    return input - output;
+}
+
+
+#define DECL_TEMPLATE(func)                                                   \
+static void Process##func(ALmodulatorState *state, ALuint SamplesToDo,        \
+  const ALfloat *RESTRICT SamplesIn,                                          \
+  ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])                                 \
+{                                                                             \
+    const ALuint step = state->step;                                          \
+    ALuint index = state->index;                                              \
+    ALfloat samp;                                                             \
+    ALuint i, k;                                                              \
+                                                                              \
+    for(i = 0;i < SamplesToDo;i++)                                            \
+    {                                                                         \
+        samp = SamplesIn[i];                                                  \
+                                                                              \
+        index += step;                                                        \
+        index &= WAVEFORM_FRACMASK;                                           \
+        samp *= func(index);                                                  \
+                                                                              \
+        samp = hpFilter1P(&state->iirFilter, 0, samp);                        \
+                                                                              \
+        for(k = 0;k < MaxChannels;k++)                                        \
+            SamplesOut[k][i] += state->Gain[k] * samp;                        \
+    }                                                                         \
+    state->index = index;                                                     \
+}
+
+DECL_TEMPLATE(Sin)
+DECL_TEMPLATE(Saw)
+DECL_TEMPLATE(Square)
+
+#undef DECL_TEMPLATE
+
+
+static ALvoid ModulatorDestroy(ALeffectState *effect)
+{
+    ALmodulatorState *state = (ALmodulatorState*)effect;
+    free(state);
+}
+
+static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
+{
+    return AL_TRUE;
+    (void)effect;
+    (void)Device;
+}
+
+static ALvoid ModulatorUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
+{
+    ALmodulatorState *state = (ALmodulatorState*)effect;
+    ALfloat gain, cw, a = 0.0f;
+    ALuint index;
+
+    if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
+        state->Waveform = SINUSOID;
+    else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
+        state->Waveform = SAWTOOTH;
+    else if(Slot->effect.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)
+        state->Waveform = SQUARE;
+
+    state->step = fastf2u(Slot->effect.Modulator.Frequency*WAVEFORM_FRACONE /
+                          Device->Frequency);
+    if(state->step == 0) state->step = 1;
+
+    cw = cosf(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
+                          Device->Frequency);
+    a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
+    state->iirFilter.coeff = a;
+
+    gain = sqrtf(1.0f/Device->NumChan);
+    gain *= Slot->Gain;
+    for(index = 0;index < MaxChannels;index++)
+        state->Gain[index] = 0.0f;
+    for(index = 0;index < Device->NumChan;index++)
+    {
+        enum Channel chan = Device->Speaker2Chan[index];
+        state->Gain[chan] = gain;
+    }
+}
+
+static ALvoid ModulatorProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
+{
+    ALmodulatorState *state = (ALmodulatorState*)effect;
+
+    switch(state->Waveform)
+    {
+        case SINUSOID:
+            ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut);
+            break;
+
+        case SAWTOOTH:
+            ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut);
+            break;
+
+        case SQUARE:
+            ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut);
+            break;
+    }
+}
+
+ALeffectState *ModulatorCreate(void)
+{
+    ALmodulatorState *state;
+
+    state = malloc(sizeof(*state));
+    if(!state)
+        return NULL;
+
+    state->state.Destroy = ModulatorDestroy;
+    state->state.DeviceUpdate = ModulatorDeviceUpdate;
+    state->state.Update = ModulatorUpdate;
+    state->state.Process = ModulatorProcess;
+
+    state->index = 0;
+    state->step = 1;
+
+    state->iirFilter.coeff = 0.0f;
+    state->iirFilter.history[0] = 0.0f;
+
+    return &state->state;
+}