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Mix reverb to output in the inner loop

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This commit is contained in:
Chris Robinson 2015-10-29 03:23:50 -07:00
parent 79ad54f0c3
commit c3d58947c3
1 changed files with 42 additions and 44 deletions
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86
Alc/effects/reverb.c
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@ -163,9 +163,9 @@ typedef struct ALreverbState {
// Late.PanGain)
ALfloat *Gain;
/* Temporary storage used when processing, before deinterlacing. */
ALfloat ReverbSamples[BUFFERSIZE][4];
ALfloat EarlySamples[BUFFERSIZE][4];
/* Temporary storage used when processing. */
ALfloat ReverbSamples[MAX_UPDATE_SAMPLES][4];
ALfloat EarlySamples[MAX_UPDATE_SAMPLES][4];
} ALreverbState;
/* This is a user config option for modifying the overall output of the reverb
@ -321,10 +321,10 @@ static inline ALvoid EarlyReflection(ALreverbState *State, ALuint todo, ALfloat
DelayLineIn(&State->Early.Delay[3], offset, f[3]);
// Output the results of the junction for all four channels.
out[i][0] += State->Early.Gain * f[0];
out[i][1] += State->Early.Gain * f[1];
out[i][2] += State->Early.Gain * f[2];
out[i][3] += State->Early.Gain * f[3];
out[i][0] = State->Early.Gain * f[0];
out[i][1] = State->Early.Gain * f[1];
out[i][2] = State->Early.Gain * f[2];
out[i][3] = State->Early.Gain * f[3];
}
}
@ -553,6 +553,7 @@ static inline ALvoid EAXVerbPass(ALreverbState *State, ALuint todo, const ALfloa
}
// Calculate the late reverb from the decorrelator taps.
memset(late, 0, sizeof(*late)*todo);
LateReverb(State, todo, late);
// Calculate and mix in any echo.
@ -565,65 +566,62 @@ static inline ALvoid EAXVerbPass(ALreverbState *State, ALuint todo, const ALfloa
static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALfloat (*restrict out)[4] = State->ReverbSamples;
ALuint index, c;
ALuint index, c, i;
memset(out, 0, SamplesToDo*4*sizeof(ALfloat));
/* Process reverb for these samples. */
for(index = 0;index < SamplesToDo;)
{
ALfloat todo = minu(SamplesToDo, MAX_UPDATE_SAMPLES);
ALuint todo = minu(SamplesToDo-index, MAX_UPDATE_SAMPLES);
VerbPass(State, todo, &SamplesIn[index], out);
for(c = 0;c < NumChannels;c++)
{
ALfloat gain = State->Gain[c];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < todo;i++)
SamplesOut[c][index+i] += gain*out[i][c&3];
}
VerbPass(State, todo, SamplesIn+index, out+index);
index += todo;
}
for(c = 0;c < NumChannels;c++)
{
ALfloat gain = State->Gain[c];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(index = 0;index < SamplesToDo;index++)
SamplesOut[c][index] += gain * out[index][c&3];
}
}
static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
ALfloat (*restrict early)[4] = State->EarlySamples;
ALfloat (*restrict late)[4] = State->ReverbSamples;
ALuint index, c;
memset(early, 0, SamplesToDo*4*sizeof(ALfloat));
memset(late, 0, SamplesToDo*4*sizeof(ALfloat));
ALuint index, c, i;
ALfloat gain;
/* Process reverb for these samples. */
for(index = 0;index < SamplesToDo;)
{
ALfloat todo = minu(SamplesToDo, MAX_UPDATE_SAMPLES);
ALuint todo = minu(SamplesToDo-index, MAX_UPDATE_SAMPLES);
EAXVerbPass(State, todo, &SamplesIn[index], early, late);
for(c = 0;c < NumChannels;c++)
{
gain = State->Early.PanGain[c];
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(i = 0;i < todo;i++)
SamplesOut[c][index+i] += gain*early[i][c&3];
}
gain = State->Late.PanGain[c];
if(fabsf(gain) > GAIN_SILENCE_THRESHOLD)
{
for(i = 0;i < todo;i++)
SamplesOut[c][index+i] += gain*late[i][c&3];
}
}
EAXVerbPass(State, todo, SamplesIn+index, early+index, late+index);
index += todo;
}
for(c = 0;c < NumChannels;c++)
{
ALfloat earlyGain, lateGain;
earlyGain = State->Early.PanGain[c];
if(fabsf(earlyGain) > GAIN_SILENCE_THRESHOLD)
{
for(index = 0;index < SamplesToDo;index++)
SamplesOut[c][index] += earlyGain*early[index][c&3];
}
lateGain = State->Late.PanGain[c];
if(fabsf(lateGain) > GAIN_SILENCE_THRESHOLD)
{
for(index = 0;index < SamplesToDo;index++)
SamplesOut[c][index] += lateGain*late[index][c&3];
}
}
}
static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)