diff options
author | Chris Robinson <[email protected]> | 2016-01-27 08:16:47 -0800 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2016-01-27 08:16:47 -0800 |
commit | 2fa3ae85c9a4050eab3a4f140fb6accd0a02ce85 (patch) | |
tree | 1aeb9e34ba5a5081de853f84a9d03159ef48fd00 /Alc/effects | |
parent | fd387beda183c2c443ec45f7193f81cf82ede164 (diff) |
Pass a pointer to the input samples array for effect processing
Diffstat (limited to 'Alc/effects')
-rw-r--r-- | Alc/effects/autowah.c | 4 | ||||
-rw-r--r-- | Alc/effects/chorus.c | 6 | ||||
-rw-r--r-- | Alc/effects/compressor.c | 6 | ||||
-rw-r--r-- | Alc/effects/dedicated.c | 4 | ||||
-rw-r--r-- | Alc/effects/distortion.c | 4 | ||||
-rw-r--r-- | Alc/effects/echo.c | 4 | ||||
-rw-r--r-- | Alc/effects/equalizer.c | 4 | ||||
-rw-r--r-- | Alc/effects/flanger.c | 6 | ||||
-rw-r--r-- | Alc/effects/modulator.c | 8 | ||||
-rw-r--r-- | Alc/effects/null.c | 2 | ||||
-rw-r--r-- | Alc/effects/reverb.c | 6 |
11 files changed, 27 insertions, 27 deletions
diff --git a/Alc/effects/autowah.c b/Alc/effects/autowah.c index 2da75c56..09aecf15 100644 --- a/Alc/effects/autowah.c +++ b/Alc/effects/autowah.c @@ -78,7 +78,7 @@ static ALvoid ALautowahState_update(ALautowahState *state, const ALCdevice *devi ComputeAmbientGains(device->AmbiCoeffs, device->NumChannels, slot->Gain, state->Gain); } -static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALuint it, kt; ALuint base; @@ -91,7 +91,7 @@ static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, for(it = 0;it < td;it++) { - ALfloat smp = SamplesIn[it+base]; + ALfloat smp = SamplesIn[0][it+base]; ALfloat a[3], b[3]; ALfloat alpha, w0; ALfloat amplitude; diff --git a/Alc/effects/chorus.c b/Alc/effects/chorus.c index 1477d58b..1c9efd47 100644 --- a/Alc/effects/chorus.c +++ b/Alc/effects/chorus.c @@ -204,7 +204,7 @@ DECL_TEMPLATE(Sinusoid) #undef DECL_TEMPLATE -static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALuint it, kt; ALuint base; @@ -217,10 +217,10 @@ static ALvoid ALchorusState_process(ALchorusState *state, ALuint SamplesToDo, co switch(state->waveform) { case CWF_Triangle: - ProcessTriangle(state, td, SamplesIn+base, temps); + ProcessTriangle(state, td, SamplesIn[0]+base, temps); break; case CWF_Sinusoid: - ProcessSinusoid(state, td, SamplesIn+base, temps); + ProcessSinusoid(state, td, SamplesIn[0]+base, temps); break; } diff --git a/Alc/effects/compressor.c b/Alc/effects/compressor.c index 52a6324d..6c3b8375 100644 --- a/Alc/effects/compressor.c +++ b/Alc/effects/compressor.c @@ -62,7 +62,7 @@ static ALvoid ALcompressorState_update(ALcompressorState *state, const ALCdevice ComputeAmbientGains(device->AmbiCoeffs, device->NumChannels, slot->Gain, state->Gain); } -static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALuint it, kt; ALuint base; @@ -79,7 +79,7 @@ static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint Samples for(it = 0;it < td;it++) { - smp = SamplesIn[it+base]; + smp = SamplesIn[0][it+base]; amplitude = fabsf(smp); if(amplitude > gain) @@ -100,7 +100,7 @@ static ALvoid ALcompressorState_process(ALcompressorState *state, ALuint Samples for(it = 0;it < td;it++) { - smp = SamplesIn[it+base]; + smp = SamplesIn[0][it+base]; amplitude = 1.0f; if(amplitude > gain) diff --git a/Alc/effects/dedicated.c b/Alc/effects/dedicated.c index fdc20601..06aad17c 100644 --- a/Alc/effects/dedicated.c +++ b/Alc/effects/dedicated.c @@ -76,7 +76,7 @@ static ALvoid ALdedicatedState_update(ALdedicatedState *state, const ALCdevice * } } -static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { const ALfloat *gains = state->gains; ALuint i, c; @@ -87,7 +87,7 @@ static ALvoid ALdedicatedState_process(ALdedicatedState *state, ALuint SamplesTo continue; for(i = 0;i < SamplesToDo;i++) - SamplesOut[c][i] += SamplesIn[i] * gains[c]; + SamplesOut[c][i] += SamplesIn[0][i] * gains[c]; } } diff --git a/Alc/effects/distortion.c b/Alc/effects/distortion.c index dc2e280a..697758f0 100644 --- a/Alc/effects/distortion.c +++ b/Alc/effects/distortion.c @@ -86,7 +86,7 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice ComputeAmbientGains(Device->AmbiCoeffs, Device->NumChannels, Slot->Gain, state->Gain); } -static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { const ALfloat fc = state->edge_coeff; ALuint base; @@ -108,7 +108,7 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples /* Fill oversample buffer using zero stuffing */ for(it = 0;it < td;it++) { - oversample_buffer[it][0] = SamplesIn[it+base]; + oversample_buffer[it][0] = SamplesIn[0][it+base]; oversample_buffer[it][1] = 0.0f; oversample_buffer[it][2] = 0.0f; oversample_buffer[it][3] = 0.0f; diff --git a/Alc/effects/echo.c b/Alc/effects/echo.c index e2bb6407..ebb6cf12 100644 --- a/Alc/effects/echo.c +++ b/Alc/effects/echo.c @@ -111,7 +111,7 @@ static ALvoid ALechoState_update(ALechoState *state, const ALCdevice *Device, co ComputePanningGains(Device->AmbiCoeffs, Device->NumChannels, coeffs, gain, state->Gain[1]); } -static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { const ALuint mask = state->BufferLength-1; const ALuint tap1 = state->Tap[0].delay; @@ -135,7 +135,7 @@ static ALvoid ALechoState_process(ALechoState *state, ALuint SamplesToDo, const // Apply damping and feedback gain to the second tap, and mix in the // new sample - smp = ALfilterState_processSingle(&state->Filter, temps[i][1]+SamplesIn[i+base]); + smp = ALfilterState_processSingle(&state->Filter, temps[i][1]+SamplesIn[0][i+base]); state->SampleBuffer[offset&mask] = smp * state->FeedGain; offset++; } diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c index 4f7846fd..8ef52986 100644 --- a/Alc/effects/equalizer.c +++ b/Alc/effects/equalizer.c @@ -126,7 +126,7 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice * ); } -static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALuint base; ALuint it; @@ -140,7 +140,7 @@ static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesTo for(it = 0;it < td;it++) { - ALfloat smp = SamplesIn[base+it]; + ALfloat smp = SamplesIn[0][base+it]; for(ft = 0;ft < 4;ft++) smp = ALfilterState_processSingle(&state->filter[ft], smp); diff --git a/Alc/effects/flanger.c b/Alc/effects/flanger.c index 5fcc8be8..21536f04 100644 --- a/Alc/effects/flanger.c +++ b/Alc/effects/flanger.c @@ -204,7 +204,7 @@ DECL_TEMPLATE(Sinusoid) #undef DECL_TEMPLATE -static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALuint it, kt; ALuint base; @@ -217,10 +217,10 @@ static ALvoid ALflangerState_process(ALflangerState *state, ALuint SamplesToDo, switch(state->waveform) { case FWF_Triangle: - ProcessTriangle(state, td, SamplesIn+base, temps); + ProcessTriangle(state, td, SamplesIn[0]+base, temps); break; case FWF_Sinusoid: - ProcessSinusoid(state, td, SamplesIn+base, temps); + ProcessSinusoid(state, td, SamplesIn[0]+base, temps); break; } diff --git a/Alc/effects/modulator.c b/Alc/effects/modulator.c index 9ef5d0dc..1719efeb 100644 --- a/Alc/effects/modulator.c +++ b/Alc/effects/modulator.c @@ -151,20 +151,20 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCdevice * ComputeAmbientGains(Device->AmbiCoeffs, Device->NumChannels, Slot->Gain, state->Gain); } -static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { switch(state->Waveform) { case SINUSOID: - ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels); + ProcessSin(state, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); break; case SAWTOOTH: - ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels); + ProcessSaw(state, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); break; case SQUARE: - ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut, NumChannels); + ProcessSquare(state, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); break; } } diff --git a/Alc/effects/null.c b/Alc/effects/null.c index d5f5bf11..0600703d 100644 --- a/Alc/effects/null.c +++ b/Alc/effects/null.c @@ -41,7 +41,7 @@ static ALvoid ALnullState_update(ALnullState* UNUSED(state), const ALCdevice* UN * input to the output buffer. The result should be added to the output buffer, * not replace it. */ -static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALuint UNUSED(samplesToDo), const ALfloat *restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALuint UNUSED(NumChannels)) +static ALvoid ALnullState_process(ALnullState* UNUSED(state), ALuint UNUSED(samplesToDo), const ALfloatBUFFERSIZE*restrict UNUSED(samplesIn), ALfloatBUFFERSIZE*restrict UNUSED(samplesOut), ALuint UNUSED(NumChannels)) { } diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c index f3de2116..b822ec19 100644 --- a/Alc/effects/reverb.c +++ b/Alc/effects/reverb.c @@ -627,12 +627,12 @@ static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, } } -static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALreverbState_process(ALreverbState *State, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { if(State->IsEax) - ALreverbState_processEax(State, SamplesToDo, SamplesIn, SamplesOut, NumChannels); + ALreverbState_processEax(State, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); else - ALreverbState_processStandard(State, SamplesToDo, SamplesIn, SamplesOut, NumChannels); + ALreverbState_processStandard(State, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); } // Given the allocated sample buffer, this function updates each delay line |