Remove unused reverb modulation code

Still unsure how to handle reverb modulation without some kind of reference
output, so remove the related functions to not waste CPU time. It'll remain in
the Git history should it ever need to be resurrected.

1 files changed, 2 insertions, 107 deletions

diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c
index 392c2258..caa02328 100644
--- a/Alc/effects/reverb.c
+++ b/Alc/effects/reverb.c
@@ -214,13 +214,6 @@ static const ALfloat LATE_LINE_LENGTHS[NUM_LINES] =
     1.9419362e-3f, 2.4466860e-3f, 3.3791220e-3f, 3.8838720e-3f
 };
 
-/* This coefficient is used to define the delay scale from the sinus, according
- * to the modulation depth property. This value must be below the shortest late
- * line length, otherwise with certain parameters (high mod time, low density)
- * the downswing can sample before the input.
- */
-static const ALfloat MODULATION_DEPTH_COEFF = 0.001f;
-
 
 typedef struct DelayLineI {
     /* The delay lines use interleaved samples, with the lengths being powers
@@ -268,18 +261,6 @@ typedef struct EarlyReflections {
     ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS];
 } EarlyReflections;
 
-typedef struct Modulator {
-    /* The vibrato time is tracked with an index over a modulus-wrapped range
-     * (in samples).
-     */
-    ALsizei Index;
-    ALsizei Range;
-    ALfloat IdxScale;
-
-    /* The LFO delay scale (in samples scaled by FRACTIONONE). */
-    ALfloat Depth[2];
-} Modulator;
-
 typedef struct LateReverb {
     /* Attenuation to compensate for the modal density and decay rate of the
      * late lines.
@@ -336,8 +317,6 @@ typedef struct ALreverbState {
 
     EarlyReflections Early;
 
-    Modulator Mod;
-
     LateReverb Late;
 
     /* Indicates the cross-fade point for delay line reads [0,FADE_SAMPLES]. */
@@ -347,7 +326,6 @@ typedef struct ALreverbState {
     ALsizei Offset;
 
     /* Temporary storage used when processing. */
-    alignas(16) ALsizei ModulationDelays[NUM_LINES][MAX_UPDATE_SAMPLES][2];
     alignas(16) ALfloat AFormatSamples[NUM_LINES][MAX_UPDATE_SAMPLES];
     alignas(16) ALfloat ReverbSamples[NUM_LINES][MAX_UPDATE_SAMPLES];
     alignas(16) ALfloat EarlySamples[NUM_LINES][MAX_UPDATE_SAMPLES];
@@ -412,12 +390,6 @@ static void ALreverbState_Construct(ALreverbState *state)
         state->Early.Coeff[i] = 0.0f;
     }
 
-    state->Mod.Index = 0;
-    state->Mod.Range = 1;
-    state->Mod.IdxScale = 0.0f;
-    state->Mod.Depth[0] = 0.0f;
-    state->Mod.Depth[1] = 0.0f;
-
     state->Late.DensityGain = 0.0f;
 
     state->Late.Delay.Mask = 0;
@@ -554,12 +526,9 @@ static ALboolean AllocLines(const ALuint frequency, ALreverbState *State)
                                    &State->Late.VecAp.Delay);
 
     /* The late delay lines are calculated from the larger of the maximum
-     * density line length or the maximum echo time, and includes the maximum
-     * modulation-related delay. The modulator's delay is calculated from the
-     * depth coefficient.
+     * density line length or the maximum echo time.
      */
-    length = maxf(AL_EAXREVERB_MAX_ECHO_TIME, LATE_LINE_LENGTHS[NUM_LINES-1]*multiplier) +
-             MODULATION_DEPTH_COEFF;
+    length = maxf(AL_EAXREVERB_MAX_ECHO_TIME, LATE_LINE_LENGTHS[NUM_LINES-1]*multiplier);
     totalSamples += CalcLineLength(length, totalSamples, frequency, 0,
                                    &State->Late.Delay);
 
@@ -965,39 +934,6 @@ static void CalcT60DampingCoeffs(const ALfloat length, const ALfloat lfDecayTime
     }
 }
 
-/* Update the EAX modulation index, range, and depth.  Keep in mind that this
- * kind of vibrato is additive and not multiplicative as one may expect.  The
- * downswing will sound stronger than the upswing.
- */
-static ALvoid UpdateModulator(const ALfloat modTime, const ALfloat modDepth,
-                              const ALuint frequency, Modulator *Mod)
-{
-    ALsizei range;
-
-    /* Modulation is calculated in two parts.
-     *
-     * The modulation time effects the speed of the sinus. An index out of the
-     * current range (both in samples) is incremented each sample, so a longer
-     * time implies a larger range. When the timing changes, the index is
-     * rescaled to the new range to keep the sinus consistent.
-     */
-    range = fastf2i(modTime*frequency + 0.5f);
-    Mod->Index = (ALsizei)(Mod->Index * (ALint64)range / Mod->Range)%range;
-    Mod->Range = range;
-    Mod->IdxScale = F_TAU / range;
-
-    /* The modulation depth effects the scale of the sinus, which varies the
-     * delay for the tapped output. This delay changing over time changes the
-     * pitch, creating the modulation effect. The scale needs to be multiplied
-     * by the modulation time (itself scaled by the max modulation time) so
-     * that a given depth produces a consistent shift in frequency over all
-     * ranges of time.
-     */
-    Mod->Depth[1] = modDepth * MODULATION_DEPTH_COEFF *
-                    (modTime / AL_EAXREVERB_MAX_MODULATION_TIME) *
-                    frequency * FRACTIONONE;
-}
-
 /* Update the offsets for the main effect delay line. */
 static ALvoid UpdateDelayLine(const ALfloat earlyDelay, const ALfloat lateDelay, const ALfloat density, const ALfloat decayTime, const ALuint frequency, ALreverbState *State)
 {
@@ -1313,10 +1249,6 @@ static ALvoid ALreverbState_update(ALreverbState *State, const ALCcontext *Conte
     hfDecayTime = clampf(props->Reverb.DecayTime * hfRatio,
                          AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME);
 
-    /* Update the modulator parameters. */
-    UpdateModulator(props->Reverb.ModulationTime, props->Reverb.ModulationDepth,
-                    frequency, &State->Mod);
-
     /* Update the late lines. */
     UpdateLateLines(props->Reverb.Density, props->Reverb.Diffusion,
                     lfDecayTime, props->Reverb.DecayTime, hfDecayTime,
@@ -1345,8 +1277,6 @@ static ALvoid ALreverbState_update(ALreverbState *State, const ALCcontext *Conte
             break;
         }
     }
-    if(State->Mod.Depth[1] != State->Mod.Depth[0])
-        State->FadeCount = 0;
 }
 
 
@@ -1392,34 +1322,6 @@ static inline ALvoid DelayLineIn4Rev(DelayLineI *Delay, ALsizei offset, const AL
         Delay->Line[offset][i] = in[NUM_LINES-1-i];
 }
 
-static void CalcModulationDelays(Modulator *Mod, ALsizei (*restrict delays)[MAX_UPDATE_SAMPLES][2],
-                                 const ALsizei (*restrict offsets)[2], const ALsizei todo)
-{
-    const ALsizei phase_offset = Mod->Range >> 2;
-    ALfloat sinus;
-    ALsizei c, i;
-
-    for(c = 0;c < NUM_LINES;c++)
-    {
-        ALsizei offset0 = offsets[c][0] << FRACTIONBITS;
-        ALsizei offset1 = offsets[c][1] << FRACTIONBITS;
-        ALsizei index = Mod->Index + phase_offset*c;
-        for(i = 0;i < todo;i++)
-        {
-            /* Calculate the sinus rhythm (dependent on modulation time and the
-             * sampling rate).
-             */
-            sinus = sinf(index * Mod->IdxScale);
-            index = (index+1) % Mod->Range;
-
-            /* Calculate the read offset. */
-            delays[c][i][0] = fastf2i(sinus*Mod->Depth[0]) + offset0;
-            delays[c][i][1] = fastf2i(sinus*Mod->Depth[1]) + offset1;
-        }
-    }
-    Mod->Index = (Mod->Index+todo) % Mod->Range;
-}
-
 /* Applies a scattering matrix to the 4-line (vector) input.  This is used
  * for both the below vector all-pass model and to perform modal feed-back
  * delay network (FDN) mixing.
@@ -1619,15 +1521,12 @@ static inline void LateT60Filter(ALfloat *restrict out, const ALfloat *restrict
 static ALvoid LateReverb_Faded(ALreverbState *State, const ALsizei todo, ALfloat fade,
                                ALfloat (*restrict out)[MAX_UPDATE_SAMPLES])
 {
-    ALsizei (*restrict moddelay)[MAX_UPDATE_SAMPLES][2] = State->ModulationDelays;
     const ALfloat apFeedCoeff = State->ApFeedCoeff;
     const ALfloat mixX = State->MixX;
     const ALfloat mixY = State->MixY;
     ALsizei offset;
     ALsizei i, j;
 
-    CalcModulationDelays(&State->Mod, moddelay, State->Late.Offset, todo);
-
     offset = State->Offset;
     for(i = 0;i < todo;i++)
     {
@@ -1664,15 +1563,12 @@ static ALvoid LateReverb_Faded(ALreverbState *State, const ALsizei todo, ALfloat
 static ALvoid LateReverb_Unfaded(ALreverbState *State, const ALsizei todo, ALfloat fade,
                                  ALfloat (*restrict out)[MAX_UPDATE_SAMPLES])
 {
-    ALsizei (*restrict moddelay)[MAX_UPDATE_SAMPLES][2] = State->ModulationDelays;
     const ALfloat apFeedCoeff = State->ApFeedCoeff;
     const ALfloat mixX = State->MixX;
     const ALfloat mixY = State->MixY;
     ALsizei offset;
     ALsizei i, j;
 
-    CalcModulationDelays(&State->Mod, moddelay, State->Late.Offset, todo);
-
     offset = State->Offset;
     for(i = 0;i < todo;i++)
     {
@@ -1774,7 +1670,6 @@ static ALvoid ALreverbState_process(ALreverbState *State, ALsizei SamplesToDo, c
                 State->Late.VecAp.Offset[c][0] = State->Late.VecAp.Offset[c][1];
                 State->Late.Offset[c][0] = State->Late.Offset[c][1];
             }
-            State->Mod.Depth[0] = State->Mod.Depth[1];
         }
 
         /* Mix the A-Format results to output, implicitly converting back to