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-rw-r--r--Alc/alcReverb.c151
1 files changed, 105 insertions, 46 deletions
diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c
index 42c823ce..f0d9e0be 100644
--- a/Alc/alcReverb.c
+++ b/Alc/alcReverb.c
@@ -31,6 +31,10 @@
#include "alError.h"
#include "alu.h"
+// Just a soft maximum. Being higher will cause VerbUpdate to reallocate the
+// sample buffer which may cause an abort if realloc fails
+#define MAX_REVERB_FREQ 192000
+
typedef struct DelayLine
{
// The delay lines use sample lengths that are powers of 2 to allow
@@ -46,6 +50,7 @@ typedef struct ALverbState {
// All delay lines are allocated as a single buffer to reduce memory
// fragmentation and management code.
ALfloat *SampleBuffer;
+ ALuint TotalLength;
// Master effect low-pass filter (2 chained 1-pole filters).
FILTER LpFilter;
ALfloat LpHistory[2];
@@ -143,6 +148,46 @@ static ALuint NextPowerOf2(ALuint value)
return powerOf2;
}
+static ALuint CalcLengths(ALuint length[13], ALuint frequency)
+{
+ ALuint samples, totalLength, index;
+
+ // All line lengths are powers of 2, calculated from their lengths, with
+ // an additional sample in case of rounding errors.
+
+ // See VerbUpdate() for an explanation of the additional calculation
+ // added to the master line length.
+ samples = (ALuint)
+ ((MASTER_LINE_LENGTH +
+ (LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER) *
+ (DECO_FRACTION * ((DECO_MULTIPLIER * DECO_MULTIPLIER *
+ DECO_MULTIPLIER) - 1.0f)))) *
+ frequency) + 1;
+ length[0] = NextPowerOf2(samples);
+ totalLength = length[0];
+ for(index = 0;index < 4;index++)
+ {
+ samples = (ALuint)(EARLY_LINE_LENGTH[index] * frequency) + 1;
+ length[1 + index] = NextPowerOf2(samples);
+ totalLength += length[1 + index];
+ }
+ for(index = 0;index < 4;index++)
+ {
+ samples = (ALuint)(ALLPASS_LINE_LENGTH[index] * frequency) + 1;
+ length[5 + index] = NextPowerOf2(samples);
+ totalLength += length[5 + index];
+ }
+ for(index = 0;index < 4;index++)
+ {
+ samples = (ALuint)(LATE_LINE_LENGTH[index] *
+ (1.0f + LATE_LINE_MULTIPLIER) * frequency) + 1;
+ length[9 + index] = NextPowerOf2(samples);
+ totalLength += length[9 + index];
+ }
+
+ return totalLength;
+}
+
// Basic delay line input/output routines.
static __inline ALfloat DelayLineOut(DelayLine *Delay, ALuint offset)
{
@@ -369,12 +414,62 @@ static __inline ALint aluCart2LUTpos(ALfloat re, ALfloat im)
// This updates the reverb state. This is called any time the reverb effect
// is loaded into a slot.
-ALvoid VerbUpdate(ALeffectState *effect, ALCcontext *Context, ALeffect *Effect)
+ALvoid VerbUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffect *Effect)
{
ALverbState *State = (ALverbState*)effect;
ALuint index;
ALfloat length, mixCoeff, cw, g, coeff;
ALfloat hfRatio = Effect->Reverb.DecayHFRatio;
+ ALuint lengths[13], totalLength;
+
+ totalLength = CalcLengths(lengths, Context->Frequency);
+ if(totalLength > State->TotalLength)
+ {
+ void *temp;
+
+ State->TotalLength = totalLength;
+ temp = realloc(State->SampleBuffer, State->TotalLength * sizeof(ALfloat));
+ if(!temp)
+ {
+ AL_PRINT("Failed reallocation!");
+ abort();
+ }
+ State->SampleBuffer = temp;
+
+ for(index = 0; index < totalLength;index++)
+ State->SampleBuffer[index] = 0.0f;
+
+ // All lines share a single sample buffer
+ State->Delay.Mask = lengths[0] - 1;
+ State->Delay.Line = &State->SampleBuffer[0];
+ totalLength = lengths[0];
+ for(index = 0;index < 4;index++)
+ {
+ State->Early.Delay[index].Mask = lengths[1 + index] - 1;
+ State->Early.Delay[index].Line = &State->SampleBuffer[totalLength];
+ totalLength += lengths[1 + index];
+ }
+ for(index = 0;index < 4;index++)
+ {
+ State->Late.ApDelay[index].Mask = lengths[5 + index] - 1;
+ State->Late.ApDelay[index].Line = &State->SampleBuffer[totalLength];
+ totalLength += lengths[5 + index];
+ }
+ for(index = 0;index < 4;index++)
+ {
+ State->Late.Delay[index].Mask = lengths[9 + index] - 1;
+ State->Late.Delay[index].Line = &State->SampleBuffer[totalLength];
+ totalLength += lengths[9 + index];
+ }
+ }
+
+ for(index = 0;index < 4;index++)
+ {
+ State->Early.Offset[index] = (ALuint)(EARLY_LINE_LENGTH[index] *
+ Context->Frequency);
+ State->Late.ApOffset[index] = (ALuint)(ALLPASS_LINE_LENGTH[index] *
+ Context->Frequency);
+ }
// Calculate the master low-pass filter (from the master effect HF gain).
cw = cos(2.0 * M_PI * Effect->Reverb.HFReference / Context->Frequency);
@@ -659,10 +754,10 @@ ALvoid EAXVerbProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint Sa
// This creates the reverb state. It should be called only when the reverb
// effect is loaded into a slot that doesn't already have a reverb effect.
-ALeffectState *VerbCreate(ALCcontext *Context)
+ALeffectState *VerbCreate(void)
{
ALverbState *State = NULL;
- ALuint samples, length[13], totalLength, index;
+ ALuint length[13], totalLength, index;
State = malloc(sizeof(ALverbState));
if(!State)
@@ -675,42 +770,10 @@ ALeffectState *VerbCreate(ALCcontext *Context)
State->state.Update = VerbUpdate;
State->state.Process = VerbProcess;
- // All line lengths are powers of 2, calculated from their lengths, with
- // an additional sample in case of rounding errors.
-
- // See VerbUpdate() for an explanation of the additional calculation
- // added to the master line length.
- samples = (ALuint)
- ((MASTER_LINE_LENGTH +
- (LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER) *
- (DECO_FRACTION * ((DECO_MULTIPLIER * DECO_MULTIPLIER *
- DECO_MULTIPLIER) - 1.0f)))) *
- Context->Frequency) + 1;
- length[0] = NextPowerOf2(samples);
- totalLength = length[0];
- for(index = 0;index < 4;index++)
- {
- samples = (ALuint)(EARLY_LINE_LENGTH[index] * Context->Frequency) + 1;
- length[1 + index] = NextPowerOf2(samples);
- totalLength += length[1 + index];
- }
- for(index = 0;index < 4;index++)
- {
- samples = (ALuint)(ALLPASS_LINE_LENGTH[index] * Context->Frequency) + 1;
- length[5 + index] = NextPowerOf2(samples);
- totalLength += length[5 + index];
- }
- for(index = 0;index < 4;index++)
- {
- samples = (ALuint)(LATE_LINE_LENGTH[index] *
- (1.0f + LATE_LINE_MULTIPLIER) * Context->Frequency) + 1;
- length[9 + index] = NextPowerOf2(samples);
- totalLength += length[9 + index];
- }
+ totalLength = CalcLengths(length, MAX_REVERB_FREQ);
- // All lines share a single sample buffer and have their masks and start
- // addresses calculated once.
- State->SampleBuffer = malloc(totalLength * sizeof(ALfloat));
+ State->TotalLength = totalLength;
+ State->SampleBuffer = malloc(State->TotalLength * sizeof(ALfloat));
if(!State->SampleBuffer)
{
free(State);
@@ -741,9 +804,7 @@ ALeffectState *VerbCreate(ALCcontext *Context)
State->Early.Delay[index].Line = &State->SampleBuffer[totalLength];
totalLength += length[1 + index];
- // The early delay lines have their read offsets calculated once.
- State->Early.Offset[index] = (ALuint)(EARLY_LINE_LENGTH[index] *
- Context->Frequency);
+ State->Early.Offset[index] = 0;
}
State->Late.Gain = 0.0f;
@@ -758,9 +819,7 @@ ALeffectState *VerbCreate(ALCcontext *Context)
State->Late.ApDelay[index].Line = &State->SampleBuffer[totalLength];
totalLength += length[5 + index];
- // The late all-pass lines have their read offsets calculated once.
- State->Late.ApOffset[index] = (ALuint)(ALLPASS_LINE_LENGTH[index] *
- Context->Frequency);
+ State->Late.ApOffset[index] = 0;
}
for(index = 0;index < 4;index++)
@@ -787,9 +846,9 @@ ALeffectState *VerbCreate(ALCcontext *Context)
return &State->state;
}
-ALeffectState *EAXVerbCreate(ALCcontext *Context)
+ALeffectState *EAXVerbCreate(void)
{
- ALeffectState *State = VerbCreate(Context);
+ ALeffectState *State = VerbCreate();
if(State) State->Process = EAXVerbProcess;
return State;
}
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-02 23:00:24 +0000