diff options
| -rw-r--r-- | Alc/effects/reverb.c | 791 |
1 files changed, 386 insertions, 405 deletions
diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c index c9397b67..f506486e 100644 --- a/Alc/effects/reverb.c +++ b/Alc/effects/reverb.c @@ -60,22 +60,20 @@ typedef struct ALreverbState { ALboolean IsEax; - // For HRTF and UHJ - ALfloat (*ExtraOut)[BUFFERSIZE]; - ALuint ExtraChannels; - // All delay lines are allocated as a single buffer to reduce memory // fragmentation and management code. ALfloat *SampleBuffer; ALuint TotalSamples; // Master effect filters - ALfilterState LpFilter; - ALfilterState HpFilter; // EAX only + struct { + ALfilterState Lp; + ALfilterState Hp; // EAX only + } Filter[4]; struct { // Modulator delay line. - DelayLine Delay; + DelayLine Delay[4]; // The vibrato time is tracked with an index over a modulus-wrapped // range (in samples). @@ -97,7 +95,7 @@ typedef struct ALreverbState { /* There are actually 4 decorrelator taps, but the first occurs at the late * reverb tap. */ - ALuint DecoTap[3]; + ALuint LateDecoTap[3]; struct { // Early reflections are done with 4 delay lines. @@ -106,8 +104,8 @@ typedef struct ALreverbState { ALuint Offset[4]; // The gain for each output channel based on 3D panning. - ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS+2]; - ALfloat PanGain[4][MAX_OUTPUT_CHANNELS+2]; + ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS]; + ALfloat PanGain[4][MAX_OUTPUT_CHANNELS]; } Early; struct { @@ -139,8 +137,8 @@ typedef struct ALreverbState { ALfloat LpSample[4]; // The gain for each output channel based on 3D panning. - ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS+2]; - ALfloat PanGain[4][MAX_OUTPUT_CHANNELS+2]; + ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS]; + ALfloat PanGain[4][MAX_OUTPUT_CHANNELS]; } Late; struct { @@ -149,8 +147,10 @@ typedef struct ALreverbState { ALfloat DensityGain; // Echo delay and all-pass lines. - DelayLine Delay; - DelayLine ApDelay; + struct { + DelayLine Feedback; + DelayLine Ap; + } Delay[4]; ALfloat Coeff; ALfloat ApFeedCoeff; @@ -161,7 +161,7 @@ typedef struct ALreverbState { // The echo line is 1-pole low-pass filtered. ALfloat LpCoeff; - ALfloat LpSample; + ALfloat LpSample[4]; // Echo mixing coefficient. ALfloat MixCoeff; @@ -171,6 +171,7 @@ typedef struct ALreverbState { ALuint Offset; /* Temporary storage used when processing. */ + alignas(16) ALfloat AFormatSamples[4][MAX_UPDATE_SAMPLES]; alignas(16) ALfloat ReverbSamples[4][MAX_UPDATE_SAMPLES]; alignas(16) ALfloat EarlySamples[4][MAX_UPDATE_SAMPLES]; } ALreverbState; @@ -192,16 +193,19 @@ static void ALreverbState_Construct(ALreverbState *state) SET_VTABLE2(ALreverbState, ALeffectState, state); state->IsEax = AL_FALSE; - state->ExtraChannels = 0; state->TotalSamples = 0; state->SampleBuffer = NULL; - ALfilterState_clear(&state->LpFilter); - ALfilterState_clear(&state->HpFilter); + for(index = 0;index < 4;index++) + { + ALfilterState_clear(&state->Filter[index].Lp); + ALfilterState_clear(&state->Filter[index].Hp); + + state->Mod.Delay[index].Mask = 0; + state->Mod.Delay[index].Line = NULL; + } - state->Mod.Delay.Mask = 0; - state->Mod.Delay.Line = NULL; state->Mod.Index = 0; state->Mod.Range = 1; state->Mod.Depth = 0.0f; @@ -212,9 +216,9 @@ static void ALreverbState_Construct(ALreverbState *state) state->Delay.Line = NULL; state->DelayTap[0] = 0; state->DelayTap[1] = 0; - state->DecoTap[0] = 0; - state->DecoTap[1] = 0; - state->DecoTap[2] = 0; + state->LateDecoTap[0] = 0; + state->LateDecoTap[1] = 0; + state->LateDecoTap[2] = 0; for(index = 0;index < 4;index++) { @@ -248,23 +252,29 @@ static void ALreverbState_Construct(ALreverbState *state) { for(index = 0;index < MAX_OUTPUT_CHANNELS;index++) { + state->Early.CurrentGain[l][index] = 0.0f; state->Early.PanGain[l][index] = 0.0f; + state->Late.CurrentGain[l][index] = 0.0f; state->Late.PanGain[l][index] = 0.0f; } } state->Echo.DensityGain = 0.0f; - state->Echo.Delay.Mask = 0; - state->Echo.Delay.Line = NULL; - state->Echo.ApDelay.Mask = 0; - state->Echo.ApDelay.Line = NULL; + for(l = 0;l < 4;l++) + { + state->Echo.Delay[l].Feedback.Mask = 0; + state->Echo.Delay[l].Feedback.Line = NULL; + state->Echo.Delay[l].Ap.Mask = 0; + state->Echo.Delay[l].Ap.Line = NULL; + } state->Echo.Coeff = 0.0f; state->Echo.ApFeedCoeff = 0.0f; state->Echo.ApCoeff = 0.0f; state->Echo.Offset = 0; state->Echo.ApOffset = 0; state->Echo.LpCoeff = 0.0f; - state->Echo.LpSample = 0.0f; + for(l = 0;l < 4;l++) + state->Echo.LpSample[l] = 0.0f; state->Echo.MixCoeff = 0.0f; state->Offset = 0; @@ -322,18 +332,18 @@ static const ALfloat EARLY_LINE_LENGTH[4] = 0.0015f, 0.0045f, 0.0135f, 0.0405f }; -// The lengths of the late all-pass delay lines. -static const ALfloat ALLPASS_LINE_LENGTH[4] = -{ - 0.0151f, 0.0167f, 0.0183f, 0.0200f, -}; - // The lengths of the late cyclical delay lines. static const ALfloat LATE_LINE_LENGTH[4] = { 0.0211f, 0.0311f, 0.0461f, 0.0680f }; +// The lengths of the late all-pass delay lines. +static const ALfloat ALLPASS_LINE_LENGTH[4] = +{ + 0.0151f, 0.0167f, 0.0183f, 0.0200f, +}; + // The late cyclical delay lines have a variable length dependent on the // effect's density parameter (inverted for some reason) and this multiplier. static const ALfloat LATE_LINE_MULTIPLIER = 4.0f; @@ -400,8 +410,9 @@ static ALboolean AllocLines(ALuint frequency, ALreverbState *State) * modulation. */ length = (AL_EAXREVERB_MAX_MODULATION_TIME*MODULATION_DEPTH_COEFF/2.0f); - totalSamples += CalcLineLength(length, totalSamples, frequency, 1, - &State->Mod.Delay); + for(index = 0;index < 4;index++) + totalSamples += CalcLineLength(length, totalSamples, frequency, 1, + &State->Mod.Delay[index]); /* The initial delay is the sum of the reflections and late reverb delays. * The decorrelator length is calculated from the lowest reverb density (a @@ -412,19 +423,17 @@ static ALboolean AllocLines(ALuint frequency, ALreverbState *State) AL_EAXREVERB_MAX_LATE_REVERB_DELAY; length += (DECO_FRACTION * DECO_MULTIPLIER * DECO_MULTIPLIER) * LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER); - totalSamples += CalcLineLength(length, totalSamples, frequency, - MAX_UPDATE_SAMPLES, &State->Delay); + /* Multiply length by 4, since we're storing 4 interleaved channels in the + * main delay line. + */ + totalSamples += CalcLineLength(length*4, totalSamples, frequency, + MAX_UPDATE_SAMPLES*4, &State->Delay); // The early reflection lines. for(index = 0;index < 4;index++) totalSamples += CalcLineLength(EARLY_LINE_LENGTH[index], totalSamples, frequency, 0, &State->Early.Delay[index]); - // The late all-pass lines. - for(index = 0;index < 4;index++) - totalSamples += CalcLineLength(ALLPASS_LINE_LENGTH[index], totalSamples, - frequency, 0, &State->Late.ApDelay[index]); - // The late delay lines are calculated from the lowest reverb density. for(index = 0;index < 4;index++) { @@ -433,17 +442,25 @@ static ALboolean AllocLines(ALuint frequency, ALreverbState *State) &State->Late.Delay[index]); } + // The late all-pass lines. + for(index = 0;index < 4;index++) + totalSamples += CalcLineLength(ALLPASS_LINE_LENGTH[index], totalSamples, + frequency, 0, &State->Late.ApDelay[index]); + // The echo all-pass and delay lines. - totalSamples += CalcLineLength(ECHO_ALLPASS_LENGTH, totalSamples, - frequency, 0, &State->Echo.ApDelay); - totalSamples += CalcLineLength(AL_EAXREVERB_MAX_ECHO_TIME, totalSamples, - frequency, 0, &State->Echo.Delay); + for(index = 0;index < 4;index++) + { + totalSamples += CalcLineLength(ECHO_ALLPASS_LENGTH, totalSamples, + frequency, 0, &State->Echo.Delay[index].Ap); + totalSamples += CalcLineLength(AL_EAXREVERB_MAX_ECHO_TIME, totalSamples, + frequency, 0, &State->Echo.Delay[index].Feedback); + } if(totalSamples != State->TotalSamples) { ALfloat *newBuffer; - TRACE("New reverb buffer length: %u samples (%f sec)\n", totalSamples, totalSamples/(float)frequency); + TRACE("New reverb buffer length: %u samples\n", totalSamples); newBuffer = al_calloc(16, sizeof(ALfloat) * totalSamples); if(!newBuffer) return AL_FALSE; @@ -456,13 +473,16 @@ static ALboolean AllocLines(ALuint frequency, ALreverbState *State) RealizeLineOffset(State->SampleBuffer, &State->Delay); for(index = 0;index < 4;index++) { + RealizeLineOffset(State->SampleBuffer, &State->Mod.Delay[index]); + RealizeLineOffset(State->SampleBuffer, &State->Early.Delay[index]); + RealizeLineOffset(State->SampleBuffer, &State->Late.ApDelay[index]); RealizeLineOffset(State->SampleBuffer, &State->Late.Delay[index]); + + RealizeLineOffset(State->SampleBuffer, &State->Echo.Delay[index].Ap); + RealizeLineOffset(State->SampleBuffer, &State->Echo.Delay[index].Feedback); } - RealizeLineOffset(State->SampleBuffer, &State->Mod.Delay); - RealizeLineOffset(State->SampleBuffer, &State->Echo.ApDelay); - RealizeLineOffset(State->SampleBuffer, &State->Echo.Delay); // Clear the sample buffer. for(index = 0;index < State->TotalSamples;index++) @@ -479,18 +499,6 @@ static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Dev if(!AllocLines(frequency, State)) return AL_FALSE; - /* HRTF and UHJ will mix to the real output for ambient output. */ - if(Device->Hrtf.Handle || Device->Uhj_Encoder) - { - State->ExtraOut = Device->RealOut.Buffer; - State->ExtraChannels = Device->RealOut.NumChannels; - } - else - { - State->ExtraOut = NULL; - State->ExtraChannels = 0; - } - // Calculate the modulation filter coefficient. Notice that the exponent // is calculated given the current sample rate. This ensures that the // resulting filter response over time is consistent across all sample @@ -688,7 +696,7 @@ static ALvoid UpdateDecorrelator(ALfloat density, ALuint frequency, ALreverbStat { length = (DECO_FRACTION * powf(DECO_MULTIPLIER, (ALfloat)index)) * LATE_LINE_LENGTH[0] * (1.0f + (density * LATE_LINE_MULTIPLIER)); - State->DecoTap[index] = fastf2u(length * frequency) + State->DelayTap[1]; + State->LateDecoTap[index] = fastf2u(length * frequency) + State->DelayTap[1]; } } @@ -718,7 +726,10 @@ static ALvoid UpdateLateLines(ALfloat xMix, ALfloat density, ALfloat decayTime, length = (LATE_LINE_LENGTH[0] + LATE_LINE_LENGTH[1] + LATE_LINE_LENGTH[2] + LATE_LINE_LENGTH[3]) / 4.0f; length *= 1.0f + (density * LATE_LINE_MULTIPLIER); - State->Late.DensityGain = CalcDensityGain( + /* To account for each channel being a discrete input, also multiply by + * sqrt(num_channels). + */ + State->Late.DensityGain = 2.0f * CalcDensityGain( CalcDecayCoeff(length, decayTime) ); @@ -783,205 +794,129 @@ static ALvoid UpdateEchoLine(ALfloat echoTime, ALfloat decayTime, ALfloat diffus State->Echo.MixCoeff = echoDepth; } -// Update the early and late 3D panning gains. -static ALvoid UpdateMixedPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALfloat EarlyGain, ALfloat LateGain, ALreverbState *State) +/* Creates a transform matrix given a reverb vector. This works by creating a + * Z-focus transform, then a rotate transform around X, then Y, to place the + * focal point in the direction of the vector, using the vector length as a + * focus strength. + * + * This isn't technically correct since the vector is supposed to define the + * aperture and not rotate the perceived soundfield, but in practice it's + * probably good enough. + */ +static aluMatrixf GetTransformFromVector(const ALfloat *vec) { - ALfloat DirGains[MAX_OUTPUT_CHANNELS]; - ALfloat coeffs[MAX_AMBI_COEFFS]; + aluMatrixf zfocus, xrot, yrot; + aluMatrixf tmp1, tmp2; ALfloat length; - ALuint i; + ALfloat sa, a; - /* With HRTF or UHJ, the normal output provides a panned reverb channel - * when a non-0-length vector is specified, while the real stereo output - * provides two other "direct" non-panned reverb channels. - */ - memset(State->Early.PanGain, 0, sizeof(State->Early.PanGain)); - length = sqrtf(ReflectionsPan[0]*ReflectionsPan[0] + ReflectionsPan[1]*ReflectionsPan[1] + ReflectionsPan[2]*ReflectionsPan[2]); - if(!(length > FLT_EPSILON)) - { - for(i = 0;i < Device->RealOut.NumChannels;i++) - State->Early.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * EarlyGain; - } - else - { - /* Note that EAX Reverb's panning vectors are using right-handed - * coordinates, rather than OpenAL's left-handed coordinates. Negate Z - * to fix this. - */ - ALfloat pan[3] = { - ReflectionsPan[0] / length, - ReflectionsPan[1] / length, - -ReflectionsPan[2] / length, - }; - length = minf(length, 1.0f); - - CalcDirectionCoeffs(pan, 0.0f, coeffs); - ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); - for(i = 0;i < Device->Dry.NumChannels;i++) - State->Early.PanGain[3][i] = DirGains[i] * EarlyGain * length; - for(i = 0;i < Device->RealOut.NumChannels;i++) - State->Early.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * EarlyGain * (1.0f-length); - } + length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]); - memset(State->Late.PanGain, 0, sizeof(State->Late.PanGain)); - length = sqrtf(LateReverbPan[0]*LateReverbPan[0] + LateReverbPan[1]*LateReverbPan[1] + LateReverbPan[2]*LateReverbPan[2]); - if(!(length > FLT_EPSILON)) - { - for(i = 0;i < Device->RealOut.NumChannels;i++) - State->Late.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * LateGain; - } - else - { - ALfloat pan[3] = { - LateReverbPan[0] / length, - LateReverbPan[1] / length, - -LateReverbPan[2] / length, - }; - length = minf(length, 1.0f); - - CalcDirectionCoeffs(pan, 0.0f, coeffs); - ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); - for(i = 0;i < Device->Dry.NumChannels;i++) - State->Late.PanGain[3][i] = DirGains[i] * LateGain * length; - for(i = 0;i < Device->RealOut.NumChannels;i++) - State->Late.PanGain[i&3][Device->Dry.NumChannels+i] = Gain * LateGain * (1.0f-length); - } -} + /* Define a Z-focus (X in Ambisonics) transform, given the panning vector + * length. + */ + sa = sinf(minf(length, 1.0f) * (F_PI/4.0f)); + aluMatrixfSet(&zfocus, + 1.0f/(1.0f+sa), 0.0f, 0.0f, (sa/(1.0f+sa))/1.732050808f, + 0.0f, sqrtf((1.0f-sa)/(1.0f+sa)), 0.0f, 0.0f, + 0.0f, 0.0f, sqrtf((1.0f-sa)/(1.0f+sa)), 0.0f, + (sa/(1.0f+sa))*1.732050808f, 0.0f, 0.0f, 1.0f/(1.0f+sa) + ); -static ALvoid UpdateDirectPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALfloat EarlyGain, ALfloat LateGain, ALreverbState *State) -{ - ALfloat AmbientGains[MAX_OUTPUT_CHANNELS]; - ALfloat DirGains[MAX_OUTPUT_CHANNELS]; - ALfloat coeffs[MAX_AMBI_COEFFS]; - ALfloat length; - ALuint i; + /* Define rotation around X (Y in Ambisonics) */ + a = atan2f(vec[1], sqrtf(vec[0]*vec[0] + vec[2]*vec[2])); + aluMatrixfSet(&xrot, + 1.0f, 0.0f, 0.0f, 0.0f, + 0.0f, 1.0f, 0.0f, 0.0f, + 0.0f, 0.0f, cosf(a), sinf(a), + 0.0f, 0.0f, -sinf(a), cosf(a) + ); - /* Apply a boost of about 3dB to better match the expected stereo output volume. */ - ComputeAmbientGains(Device->Dry, Gain*1.414213562f, AmbientGains); + /* Define rotation around Y (Z in Ambisonics). NOTE: EFX's reverb vectors + * use a right-handled coordinate system, compared to the rest of OpenAL + * which uses left-handed. This is fixed by negating Z, however it would + * need to also be negated to get a proper Ambisonics angle, thus + * cancelling it out. + */ + a = atan2f(-vec[0], vec[2]); + aluMatrixfSet(&yrot, + 1.0f, 0.0f, 0.0f, 0.0f, + 0.0f, cosf(a), 0.0f, sinf(a), + 0.0f, 0.0f, 1.0f, 0.0f, + 0.0f, -sinf(a), 0.0f, cosf(a) + ); - memset(State->Early.PanGain, 0, sizeof(State->Early.PanGain)); - length = sqrtf(ReflectionsPan[0]*ReflectionsPan[0] + ReflectionsPan[1]*ReflectionsPan[1] + ReflectionsPan[2]*ReflectionsPan[2]); - if(!(length > FLT_EPSILON)) - { - for(i = 0;i < Device->Dry.NumChannels;i++) - State->Early.PanGain[i&3][i] = AmbientGains[i] * EarlyGain; - } - else - { - ALfloat pan[3] = { - ReflectionsPan[0] / length, - ReflectionsPan[1] / length, - -ReflectionsPan[2] / length, - }; - length = minf(length, 1.0f); - - CalcDirectionCoeffs(pan, 0.0f, coeffs); - ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); - for(i = 0;i < Device->Dry.NumChannels;i++) - State->Early.PanGain[i&3][i] = lerp(AmbientGains[i], DirGains[i], length) * EarlyGain; - } +#define MATRIX_MULT(_res, _m1, _m2) do { \ + int row, col; \ + for(col = 0;col < 4;col++) \ + { \ + for(row = 0;row < 4;row++) \ + _res.m[row][col] = _m1.m[row][0]*_m2.m[0][col] + _m1.m[row][1]*_m2.m[1][col] + \ + _m1.m[row][2]*_m2.m[2][col] + _m1.m[row][3]*_m2.m[3][col]; \ + } \ +} while(0) + /* Define a matrix that first focuses on Z, then rotates around X then Y to + * focus the output in the direction of the vector. + */ + MATRIX_MULT(tmp1, xrot, zfocus); + MATRIX_MULT(tmp2, yrot, tmp1); +#undef MATRIX_MULT - memset(State->Late.PanGain, 0, sizeof(State->Late.PanGain)); - length = sqrtf(LateReverbPan[0]*LateReverbPan[0] + LateReverbPan[1]*LateReverbPan[1] + LateReverbPan[2]*LateReverbPan[2]); - if(!(length > FLT_EPSILON)) - { - for(i = 0;i < Device->Dry.NumChannels;i++) - State->Late.PanGain[i&3][i] = AmbientGains[i] * LateGain; - } - else - { - ALfloat pan[3] = { - LateReverbPan[0] / length, - LateReverbPan[1] / length, - -LateReverbPan[2] / length, - }; - length = minf(length, 1.0f); - - CalcDirectionCoeffs(pan, 0.0f, coeffs); - ComputePanningGains(Device->Dry, coeffs, Gain, DirGains); - for(i = 0;i < Device->Dry.NumChannels;i++) - State->Late.PanGain[i&3][i] = lerp(AmbientGains[i], DirGains[i], length) * LateGain; - } + return tmp2; } +// Update the early and late 3D panning gains. static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALfloat Gain, ALfloat EarlyGain, ALfloat LateGain, ALreverbState *State) { - static const ALfloat PanDirs[4][3] = { - { -0.707106781f, 0.0f, -0.707106781f }, /* Front left */ - { 0.707106781f, 0.0f, -0.707106781f }, /* Front right */ - { 0.707106781f, 0.0f, 0.707106781f }, /* Back right */ - { -0.707106781f, 0.0f, 0.707106781f } /* Back left */ - }; - ALfloat coeffs[MAX_AMBI_COEFFS]; - ALfloat gain[4]; - ALfloat length; + /* Converts early reflections A-Format to B-Format (transposed). */ + static const aluMatrixf EarlyA2B = {{ + { 0.8660254038f, 0.8660254038f, 0.8660254038f, 0.8660254038f }, + { 0.8660254038f, 0.8660254038f, -0.8660254038f, -0.8660254038f }, + { 0.8660254038f, -0.8660254038f, 0.8660254038f, -0.8660254038f }, + { 0.8660254038f, -0.8660254038f, -0.8660254038f, 0.8660254038f } + }}; + /* Converts late reverb A-Format to B-Format (transposed). */ + static const aluMatrixf LateA2B = {{ + { 0.8660254038f, -0.8660254038f, 0.8660254038f, 0.8660254038f }, + { 0.8660254038f, -0.8660254038f, -0.8660254038f, -0.8660254038f }, + { 0.8660254038f, 0.8660254038f, 0.8660254038f, -0.8660254038f }, + { 0.8660254038f, 0.8660254038f, -0.8660254038f, 0.8660254038f } +/* { 0.8660254038f, 1.2247448714f, 0.0f, 0.8660254038f }, + { 0.8660254038f, 0.0f, -1.2247448714f, -0.8660254038f }, + { 0.8660254038f, 0.0f, 1.2247448714f, -0.8660254038f }, + { 0.8660254038f, -1.2247448714f, 0.0f, 0.8660254038f }*/ + }}; + aluMatrixf transform, rot; ALuint i; - /* sqrt(0.5) would be the gain scaling when the panning vector is 0. This - * also equals sqrt(2/4), a nice gain scaling for the four virtual points - * producing an "ambient" response. + STATIC_CAST(ALeffectState,State)->OutBuffer = Device->FOAOut.Buffer; + STATIC_CAST(ALeffectState,State)->OutChannels = Device->FOAOut.NumChannels; + + /* Note: Both _m2 and _res are transposed. */ +#define MATRIX_MULT(_res, _m1, _m2) do { \ + int row, col; \ + for(col = 0;col < 4;col++) \ + { \ + for(row = 0;row < 4;row++) \ + _res.m[col][row] = _m1.m[row][0]*_m2.m[col][0] + _m1.m[row][1]*_m2.m[col][1] + \ + _m1.m[row][2]*_m2.m[col][2] + _m1.m[row][3]*_m2.m[col][3]; \ + } \ +} while(0) + /* Create a matrix that first converts A-Format to B-Format, then rotates + * the B-Format soundfield according to the panning vector. */ - gain[0] = gain[1] = gain[2] = gain[3] = 0.707106781f; - length = sqrtf(ReflectionsPan[0]*ReflectionsPan[0] + ReflectionsPan[1]*ReflectionsPan[1] + ReflectionsPan[2]*ReflectionsPan[2]); - if(length > 1.0f) - { - ALfloat pan[3] = { - ReflectionsPan[0] / length, - ReflectionsPan[1] / length, - -ReflectionsPan[2] / length, - }; - for(i = 0;i < 4;i++) - { - ALfloat dotp = pan[0]*PanDirs[i][0] + pan[1]*PanDirs[i][1] + pan[2]*PanDirs[i][2]; - gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); - } - } - else if(length > FLT_EPSILON) - { - for(i = 0;i < 4;i++) - { - ALfloat dotp = ReflectionsPan[0]*PanDirs[i][0] + ReflectionsPan[1]*PanDirs[i][1] + - -ReflectionsPan[2]*PanDirs[i][2]; - gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); - } - } - for(i = 0;i < 4;i++) - { - CalcDirectionCoeffs(PanDirs[i], 0.0f, coeffs); - ComputePanningGains(Device->Dry, coeffs, Gain*EarlyGain*gain[i], - State->Early.PanGain[i]); - } - - gain[0] = gain[1] = gain[2] = gain[3] = 0.707106781f; - length = sqrtf(LateReverbPan[0]*LateReverbPan[0] + LateReverbPan[1]*LateReverbPan[1] + LateReverbPan[2]*LateReverbPan[2]); - if(length > 1.0f) - { - ALfloat pan[3] = { - LateReverbPan[0] / length, - LateReverbPan[1] / length, - -LateReverbPan[2] / length, - }; - for(i = 0;i < 4;i++) - { - ALfloat dotp = pan[0]*PanDirs[i][0] + pan[1]*PanDirs[i][1] + pan[2]*PanDirs[i][2]; - gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); - } - } - else if(length > FLT_EPSILON) - { - for(i = 0;i < 4;i++) - { - ALfloat dotp = LateReverbPan[0]*PanDirs[i][0] + LateReverbPan[1]*PanDirs[i][1] + - -LateReverbPan[2]*PanDirs[i][2]; - gain[i] = sqrtf(clampf(dotp*0.5f + 0.5f, 0.0f, 1.0f)); - } - } - for(i = 0;i < 4;i++) - { - CalcDirectionCoeffs(PanDirs[i], 0.0f, coeffs); - ComputePanningGains(Device->Dry, coeffs, Gain*LateGain*gain[i], - State->Late.PanGain[i]); - } + rot = GetTransformFromVector(ReflectionsPan); + MATRIX_MULT(transform, rot, EarlyA2B); + memset(&State->Early.PanGain, 0, sizeof(State->Early.PanGain)); + for(i = 0;i < MAX_EFFECT_CHANNELS;i++) + ComputeFirstOrderGains(Device->FOAOut, transform.m[i], Gain*EarlyGain, State->Early.PanGain[i]); + + rot = GetTransformFromVector(LateReverbPan); + MATRIX_MULT(transform, rot, LateA2B); + memset(&State->Late.PanGain, 0, sizeof(State->Late.PanGain)); + for(i = 0;i < MAX_EFFECT_CHANNELS;i++) + ComputeFirstOrderGains(Device->FOAOut, transform.m[i], Gain*LateGain, State->Late.PanGain[i]); +#undef MATRIX_MULT } static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props) @@ -990,6 +925,7 @@ static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device ALfloat lfscale, hfscale, hfRatio; ALfloat gain, gainlf, gainhf; ALfloat cw, x, y; + ALuint i; if(Slot->Params.EffectType == AL_EFFECT_EAXREVERB && !EmulateEAXReverb) State->IsEax = AL_TRUE; @@ -999,12 +935,26 @@ static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device // Calculate the master filters hfscale = props->Reverb.HFReference / frequency; gainhf = maxf(props->Reverb.GainHF, 0.0001f); - ALfilterState_setParams(&State->LpFilter, ALfilterType_HighShelf, + ALfilterState_setParams(&State->Filter[0].Lp, ALfilterType_HighShelf, gainhf, hfscale, calc_rcpQ_from_slope(gainhf, 0.75f)); lfscale = props->Reverb.LFReference / frequency; gainlf = maxf(props->Reverb.GainLF, 0.0001f); - ALfilterState_setParams(&State->HpFilter, ALfilterType_LowShelf, + ALfilterState_setParams(&State->Filter[0].Hp, ALfilterType_LowShelf, gainlf, lfscale, calc_rcpQ_from_slope(gainlf, 0.75f)); + for(i = 1;i < 4;i++) + { + State->Filter[i].Lp.a1 = State->Filter[0].Lp.a1; + State->Filter[i].Lp.a2 = State->Filter[0].Lp.a2; + State->Filter[i].Lp.b0 = State->Filter[0].Lp.b0; + State->Filter[i].Lp.b1 = State->Filter[0].Lp.b1; + State->Filter[i].Lp.b2 = State->Filter[0].Lp.b2; + + State->Filter[i].Hp.a1 = State->Filter[0].Hp.a1; + State->Filter[i].Hp.a2 = State->Filter[0].Hp.a2; + State->Filter[i].Hp.b0 = State->Filter[0].Hp.b0; + State->Filter[i].Hp.b1 = State->Filter[0].Hp.b1; + State->Filter[i].Hp.b2 = State->Filter[0].Hp.b2; + } // Update the modulator line. UpdateModulator(props->Reverb.ModulationTime, props->Reverb.ModulationDepth, @@ -1045,22 +995,10 @@ static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device gain = props->Reverb.Gain * Slot->Params.Gain * ReverbBoost; // Update early and late 3D panning. - if(Device->Hrtf.Handle || Device->Uhj_Encoder) - UpdateMixedPanning(Device, props->Reverb.ReflectionsPan, - props->Reverb.LateReverbPan, gain, - props->Reverb.ReflectionsGain, - props->Reverb.LateReverbGain, State); - else if(Device->AmbiDecoder || (Device->FmtChans >= DevFmtAmbi1 && - Device->FmtChans <= DevFmtAmbi3)) - Update3DPanning(Device, props->Reverb.ReflectionsPan, - props->Reverb.LateReverbPan, gain, - props->Reverb.ReflectionsGain, - props->Reverb.LateReverbGain, State); - else - UpdateDirectPanning(Device, props->Reverb.ReflectionsPan, - props->Reverb.LateReverbPan, gain, - props->Reverb.ReflectionsGain, - props->Reverb.LateReverbGain, State); + Update3DPanning(Device, props->Reverb.ReflectionsPan, + props->Reverb.LateReverbPan, gain, + props->Reverb.ReflectionsGain, + props->Reverb.LateReverbGain, State); } @@ -1079,45 +1017,64 @@ static inline ALvoid DelayLineIn(DelayLine *Delay, ALuint offset, ALfloat in) Delay->Line[offset&Delay->Mask] = in; } -// Given some input samples, this function produces modulation for the late -// reverb. -static void EAXModulation(ALreverbState *State, ALuint offset, ALfloat*restrict dst, const ALfloat*restrict src, ALuint todo) +static inline ALfloat DelayLineInOut(DelayLine *Delay, ALuint offset, ALuint outoffset, ALfloat in) { - ALfloat sinus, frac, fdelay; - ALfloat out0, out1; - ALuint delay, i; + Delay->Line[offset&Delay->Mask] = in; + return Delay->Line[(offset-outoffset)&Delay->Mask]; +} +static void CalcModulationDelays(ALreverbState *State, ALfloat *restrict delays, ALuint todo) +{ + ALfloat sinus, range; + ALuint index, i; + + index = State->Mod.Index; + range = State->Mod.Filter; for(i = 0;i < todo;i++) { /* Calculate the sinus rythm (dependent on modulation time and the * sampling rate). The center of the sinus is moved to reduce the * delay of the effect when the time or depth are low. */ - sinus = 1.0f - cosf(F_TAU * State->Mod.Index / State->Mod.Range); + sinus = 1.0f - cosf(F_TAU * index / State->Mod.Range); /* Step the modulation index forward, keeping it bound to its range. */ - State->Mod.Index = (State->Mod.Index + 1) % State->Mod.Range; + index = (index+1) % State->Mod.Range; /* The depth determines the range over which to read the input samples * from, so it must be filtered to reduce the distortion caused by even * small parameter changes. */ - State->Mod.Filter = lerp(State->Mod.Filter, State->Mod.Depth, - State->Mod.Coeff); + range = lerp(range, State->Mod.Depth, State->Mod.Coeff); + + /* Calculate the read offset with fraction. */ + delays[i] = range*sinus; + } + State->Mod.Index = index; + State->Mod.Filter = range; +} + +// Given some input samples, this function produces modulation for the late +// reverb. +static void EAXModulation(DelayLine *ModDelay, ALuint offset, const ALfloat *restrict delays, ALfloat*restrict dst, const ALfloat*restrict src, ALuint todo) +{ + ALfloat frac, fdelay; + ALfloat out0, out1; + ALuint delay, i; - /* Calculate the read offset and fraction between it and the next + for(i = 0;i < todo;i++) + { + /* Separate the integer offset and fraction between it and the next * sample. */ - frac = modff(State->Mod.Filter*sinus, &fdelay); + frac = modff(delays[i], &fdelay); delay = fastf2u(fdelay); - /* Add the incoming sample to the delay line first, so a 0 delay gets - * the incoming sample. + /* Add the incoming sample to the delay line, and get the two samples + * crossed by the offset delay. */ - DelayLineIn(&State->Mod.Delay, offset, src[i]); - /* Get the two samples crossed by the offset delay */ - out0 = DelayLineOut(&State->Mod.Delay, offset - delay); - out1 = DelayLineOut(&State->Mod.Delay, offset - delay - 1); + out0 = DelayLineInOut(ModDelay, offset, delay, src[i]); + out1 = DelayLineOut(ModDelay, offset - delay - 1); offset++; /* The output is obtained by linearly interpolating the two samples @@ -1127,9 +1084,10 @@ static void EAXModulation(ALreverbState *State, ALuint offset, ALfloat*restrict } } -// Given some input sample, this function produces four-channel outputs for the -// early reflections. -static inline ALvoid EarlyReflection(ALreverbState *State, ALuint todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) +/* Given some input samples from the main delay line, this function produces + * four-channel outputs for the early reflections. + */ +static ALvoid EarlyReflection(ALreverbState *State, ALuint todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat d[4], v, f[4]; ALuint i; @@ -1138,11 +1096,11 @@ static inline ALvoid EarlyReflection(ALreverbState *State, ALuint todo, ALfloat { ALuint offset = State->Offset+i; - // Obtain the decayed results of each early delay line. - d[0] = DelayLineOut(&State->Early.Delay[0], offset-State->Early.Offset[0]) * State->Early.Coeff[0]; - d[1] = DelayLineOut(&State->Early.Delay[1], offset-State->Early.Offset[1]) * State->Early.Coeff[1]; - d[2] = DelayLineOut(&State->Early.Delay[2], offset-State->Early.Offset[2]) * State->Early.Coeff[2]; - d[3] = DelayLineOut(&State->Early.Delay[3], offset-State->Early.Offset[3]) * State->Early.Coeff[3]; + /* Obtain the first reflection samples from the main delay line. */ + f[0] = DelayLineOut(&State->Delay, (offset-State->DelayTap[0])*4 + 0); + f[1] = DelayLineOut(&State->Delay, (offset-State->DelayTap[0])*4 + 1); + f[2] = DelayLineOut(&State->Delay, (offset-State->DelayTap[0])*4 + 2); + f[3] = DelayLineOut(&State->Delay, (offset-State->DelayTap[0])*4 + 3); /* The following uses a lossless scattering junction from waveguide * theory. It actually amounts to a householder mixing matrix, which @@ -1155,29 +1113,27 @@ static inline ALvoid EarlyReflection(ALreverbState *State, ALuint todo, ALfloat * --- * i=1 */ - v = (d[0] + d[1] + d[2] + d[3]) * 0.5f; - // The junction is loaded with the input here. - v += DelayLineOut(&State->Delay, offset-State->DelayTap[0]); - - // Calculate the feed values for the delay lines. - f[0] = v - d[0]; - f[1] = v - d[1]; - f[2] = v - d[2]; - f[3] = v - d[3]; - - // Re-feed the delay lines. - DelayLineIn(&State->Early.Delay[0], offset, f[0]); - DelayLineIn(&State->Early.Delay[1], offset, f[1]); - DelayLineIn(&State->Early.Delay[2], offset, f[2]); - DelayLineIn(&State->Early.Delay[3], offset, f[3]); - - /* Output the results of the junction for all four channels with a - * constant attenuation of 0.5. + v = (f[0] + f[1] + f[2] + f[3]) * 0.5f; + + /* Calculate the feed values for the early delay lines. */ + d[0] = v - f[0]; + d[1] = v - f[1]; + d[2] = v - f[2]; + d[3] = v - f[3]; + + /* Feed the early delay lines, and load the delayed results. */ + d[0] = DelayLineInOut(&State->Early.Delay[0], offset, State->Early.Offset[0], d[0]); + d[1] = DelayLineInOut(&State->Early.Delay[1], offset, State->Early.Offset[1], d[1]); + d[2] = DelayLineInOut(&State->Early.Delay[2], offset, State->Early.Offset[2], d[2]); + d[3] = DelayLineInOut(&State->Early.Delay[3], offset, State->Early.Offset[3], d[3]); + + /* Output the initial reflection taps and the results of the delayed + * and decayed junction for all four channels. */ - out[0][i] = f[0] * 0.5f; - out[1][i] = f[1] * 0.5f; - out[2][i] = f[2] * 0.5f; - out[3][i] = f[3] * 0.5f; + out[0][i] = f[0] + d[0]*State->Early.Coeff[0]; + out[1][i] = f[1] + d[1]*State->Early.Coeff[1]; + out[2][i] = f[2] + d[2]*State->Early.Coeff[2]; + out[3][i] = f[3] + d[3]*State->Early.Coeff[3]; } } @@ -1215,7 +1171,7 @@ static inline ALfloat LateLowPassInOut(ALreverbState *State, ALuint index, ALflo // Given four decorrelated input samples, this function produces four-channel // output for the late reverb. -static inline ALvoid LateReverb(ALreverbState *State, ALuint todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) +static ALvoid LateReverb(ALreverbState *State, ALuint todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat d[4], f[4]; ALuint offset; @@ -1230,10 +1186,10 @@ static inline ALvoid LateReverb(ALreverbState *State, ALuint todo, ALfloat (*res for(i = 0;i < tmp_todo;i++) { /* Obtain four decorrelated input samples. */ - f[0] = DelayLineOut(&State->Delay, offset-State->DelayTap[1]) * State->Late.DensityGain; - f[1] = DelayLineOut(&State->Delay, offset-State->DecoTap[0]) * State->Late.DensityGain; - f[2] = DelayLineOut(&State->Delay, offset-State->DecoTap[1]) * State->Late.DensityGain; - f[3] = DelayLineOut(&State->Delay, offset-State->DecoTap[2]) * State->Late.DensityGain; + f[0] = DelayLineOut(&State->Delay, (offset-State->DelayTap[1])*4 + 0) * State->Late.DensityGain; + f[1] = DelayLineOut(&State->Delay, (offset-State->LateDecoTap[0])*4 + 1) * State->Late.DensityGain; + f[2] = DelayLineOut(&State->Delay, (offset-State->LateDecoTap[1])*4 + 2) * State->Late.DensityGain; + f[3] = DelayLineOut(&State->Delay, (offset-State->LateDecoTap[2])*4 + 3) * State->Late.DensityGain; /* Add the decayed results of the cyclical delay lines, then pass * the results through the low-pass filters. @@ -1243,13 +1199,13 @@ static inline ALvoid LateReverb(ALreverbState *State, ALuint todo, ALfloat (*res f[2] += DelayLineOut(&State->Late.Delay[2], offset-State->Late.Offset[2]) * State->Late.Coeff[2]; f[3] += DelayLineOut(&State->Late.Delay[3], offset-State->Late.Offset[3]) * State->Late.Coeff[3]; - /* This is where the feed-back cycles from line 0 to 1 to 3 to 2 + /* This is where the feed-back cycles from line 0 to 3 to 1 to 2 * and back to 0. */ d[0] = LateLowPassInOut(State, 2, f[2]); - d[1] = LateLowPassInOut(State, 0, f[0]); - d[2] = LateLowPassInOut(State, 3, f[3]); - d[3] = LateLowPassInOut(State, 1, f[1]); + d[1] = LateLowPassInOut(State, 3, f[3]); + d[2] = LateLowPassInOut(State, 1, f[1]); + d[3] = LateLowPassInOut(State, 0, f[0]); /* To help increase diffusion, run each line through an all-pass * filter. When there is no diffusion, the shortest all-pass filter @@ -1322,57 +1278,58 @@ static inline ALvoid LateReverb(ALreverbState *State, ALuint todo, ALfloat (*res // Given an input sample, this function mixes echo into the four-channel late // reverb. -static inline ALvoid EAXEcho(ALreverbState *State, ALuint todo, ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]) +static ALvoid EAXEcho(ALreverbState *State, ALuint todo, ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]) { - ALfloat out[MAX_UPDATE_SAMPLES]; ALfloat feed; ALuint offset; - ALuint i; + ALuint c, i; - offset = State->Offset; - for(i = 0;i < todo;i++) + for(c = 0;c < 4;c++) { - // Get the latest attenuated echo sample for output. - feed = DelayLineOut(&State->Echo.Delay, offset-State->Echo.Offset) * - State->Echo.Coeff; - - // Write the output. - out[i] = State->Echo.MixCoeff * feed; - - // Mix the energy-attenuated input with the output and pass it through - // the echo low-pass filter. - feed += DelayLineOut(&State->Delay, offset-State->DelayTap[1]) * - State->Echo.DensityGain; - feed = lerp(feed, State->Echo.LpSample, State->Echo.LpCoeff); - State->Echo.LpSample = feed; - - // Then the echo all-pass filter. - feed = AllpassInOut(&State->Echo.ApDelay, offset-State->Echo.ApOffset, - offset, feed, State->Echo.ApFeedCoeff, - State->Echo.ApCoeff); - - // Feed the delay with the mixed and filtered sample. - DelayLineIn(&State->Echo.Delay, offset, feed); - offset++; + offset = State->Offset; + for(i = 0;i < todo;i++) + { + // Get the latest attenuated echo sample for output. + feed = DelayLineOut(&State->Echo.Delay[c].Feedback, offset-State->Echo.Offset) * + State->Echo.Coeff; + + // Write the output. + late[c][i] += State->Echo.MixCoeff * feed; + + // Mix the energy-attenuated input with the output and pass it through + // the echo low-pass filter. + feed += DelayLineOut(&State->Delay, (offset-State->DelayTap[1])*4 + c) * + State->Echo.DensityGain; + feed = lerp(feed, State->Echo.LpSample[c], State->Echo.LpCoeff); + State->Echo.LpSample[c] = feed; + + // Then the echo all-pass filter. + feed = AllpassInOut(&State->Echo.Delay[c].Ap, offset-State->Echo.ApOffset, + offset, feed, State->Echo.ApFeedCoeff, + State->Echo.ApCoeff); + + // Feed the delay with the mixed and filtered sample. + DelayLineIn(&State->Echo.Delay[c].Feedback, offset, feed); + offset++; + } } - - // Mix the output into the late reverb channels. - for(i = 0;i < todo;i++) late[0][i] += out[i]; - for(i = 0;i < todo;i++) late[1][i] += out[i]; - for(i = 0;i < todo;i++) late[2][i] += out[i]; - for(i = 0;i < todo;i++) late[3][i] += out[i]; } // Perform the non-EAX reverb pass on a given input sample, resulting in // four-channel output. -static inline ALvoid VerbPass(ALreverbState *State, ALuint todo, const ALfloat *input, ALfloat (*restrict early)[MAX_UPDATE_SAMPLES], ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]) +static ALvoid VerbPass(ALreverbState *State, ALuint todo, ALfloat (*restrict input)[MAX_UPDATE_SAMPLES], ALfloat (*restrict early)[MAX_UPDATE_SAMPLES], ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]) { - ALuint i; + ALuint i, c; - // Low-pass filter the incoming samples (use the early buffer as temp storage). - ALfilterState_process(&State->LpFilter, &early[0][0], input, todo); - for(i = 0;i < todo;i++) - DelayLineIn(&State->Delay, State->Offset+i, early[0][i]); + for(c = 0;c < 4;c++) + { + /* Low-pass filter the incoming samples (use the early buffer as temp + * storage). + */ + ALfilterState_process(&State->Filter[c].Lp, &early[0][0], input[c], todo); + for(i = 0;i < todo;i++) + DelayLineIn(&State->Delay, (State->Offset+i)*4 + c, early[0][i]); + } // Calculate the early reflection from the first delay tap. EarlyReflection(State, todo, early); @@ -1386,19 +1343,27 @@ static inline ALvoid VerbPass(ALreverbState *State, ALuint todo, const ALfloat * // Perform the EAX reverb pass on a given input sample, resulting in four- // channel output. -static inline ALvoid EAXVerbPass(ALreverbState *State, ALuint todo, const ALfloat *input, ALfloat (*restrict early)[MAX_UPDATE_SAMPLES], ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]) +static ALvoid EAXVerbPass(ALreverbState *State, ALuint todo, ALfloat (*restrict input)[MAX_UPDATE_SAMPLES], ALfloat (*restrict early)[MAX_UPDATE_SAMPLES], ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]) { - ALuint i; + ALuint i, c; - /* Perform any modulation on the input (use the early buffer as temp storage). */ - EAXModulation(State, State->Offset, &early[0][0], input, todo); - /* Band-pass the incoming samples */ - ALfilterState_process(&State->LpFilter, &early[1][0], &early[0][0], todo); - ALfilterState_process(&State->HpFilter, &early[2][0], &early[1][0], todo); + /* Perform any modulation on the input (use the early and late buffers as + * temp storage). + */ + CalcModulationDelays(State, &late[0][0], todo); + for(c = 0;c < 4;c++) + { + EAXModulation(&State->Mod.Delay[c], State->Offset, &late[0][0], + &early[0][0], input[c], todo); - // Feed the initial delay line. - for(i = 0;i < todo;i++) - DelayLineIn(&State->Delay, State->Offset+i, early[2][i]); + /* Band-pass the incoming samples */ + ALfilterState_process(&State->Filter[c].Lp, &early[1][0], &early[0][0], todo); + ALfilterState_process(&State->Filter[c].Hp, &early[2][0], &early[1][0], todo); + + /* Feed the initial delay line. */ + for(i = 0;i < todo;i++) + DelayLineIn(&State->Delay, (State->Offset+i)*4 + c, early[2][i]); + } // Calculate the early reflection from the first delay tap. EarlyReflection(State, todo, early); @@ -1441,11 +1406,18 @@ static void DoMix(const ALfloat *restrict src, ALfloat (*dst)[BUFFERSIZE], ALuin current_gains[c] = gains[c].Current; } -static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { + static const aluMatrixf B2A = {{ + { 0.288675134595f, 0.288675134595f, 0.288675134595f, 0.288675134595f }, + { 0.288675134595f, 0.288675134595f, -0.288675134595f, -0.288675134595f }, + { 0.288675134595f, -0.288675134595f, 0.288675134595f, -0.288675134595f }, + { 0.288675134595f, -0.288675134595f, -0.288675134595f, 0.288675134595f } + }}; + ALfloat (*restrict afmt)[MAX_UPDATE_SAMPLES] = State->AFormatSamples; ALfloat (*restrict early)[MAX_UPDATE_SAMPLES] = State->EarlySamples; ALfloat (*restrict late)[MAX_UPDATE_SAMPLES] = State->ReverbSamples; - ALuint base, c; + ALuint base, c, c2, i; /* Process reverb for these samples. */ for(base = 0;base < SamplesToDo;) @@ -1453,42 +1425,53 @@ static ALvoid ALreverbState_processStandard(ALreverbState *State, ALuint Samples const ALfloat delta = 1.0f / (ALfloat)(SamplesToDo-base); ALuint todo = minu(SamplesToDo-base, MAX_UPDATE_SAMPLES); - VerbPass(State, todo, &SamplesIn[base], early, late); + /* Convert B-Foramt to A-Format for processing (could use the row + * mixers). + */ + memset(afmt, 0, sizeof(*afmt)*4); + for(c = 0;c < 4;c++) + { + for(c2 = 0;c2 < MAX_EFFECT_CHANNELS;c2++) + { + for(i = 0;i < todo;i++) + afmt[c][i] += SamplesIn[c2][base+i] * B2A.m[c][c2]; + } + } + + VerbPass(State, todo, afmt, early, late); + /* Mix the A-Format results to output, implicitly converting back to + * B-Format. + */ for(c = 0;c < 4;c++) { DoMix(early[c], SamplesOut, NumChannels, State->Early.PanGain[c], State->Early.CurrentGain[c], delta, base, SamplesToDo-base, todo ); - if(State->ExtraChannels > 0) - DoMix(early[c], State->ExtraOut, State->ExtraChannels, - State->Early.PanGain[c]+NumChannels, - State->Early.CurrentGain[c]+NumChannels, delta, base, - SamplesToDo-base, todo - ); } for(c = 0;c < 4;c++) { DoMix(late[c], SamplesOut, NumChannels, State->Late.PanGain[c], State->Late.CurrentGain[c], delta, base, SamplesToDo, todo ); - if(State->ExtraChannels > 0) - DoMix(late[c], State->ExtraOut, State->ExtraChannels, - State->Late.PanGain[c]+NumChannels, - State->Late.CurrentGain[c]+NumChannels, delta, base, - SamplesToDo-base, todo - ); } base += todo; } } -static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) +static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { + static const aluMatrixf B2A = {{ + { 0.288675134595f, 0.288675134595f, 0.288675134595f, 0.288675134595f }, + { 0.288675134595f, 0.288675134595f, -0.288675134595f, -0.288675134595f }, + { 0.288675134595f, -0.288675134595f, 0.288675134595f, -0.288675134595f }, + { 0.288675134595f, -0.288675134595f, -0.288675134595f, 0.288675134595f } + }}; + ALfloat (*restrict afmt)[MAX_UPDATE_SAMPLES] = State->AFormatSamples; ALfloat (*restrict early)[MAX_UPDATE_SAMPLES] = State->EarlySamples; ALfloat (*restrict late)[MAX_UPDATE_SAMPLES] = State->ReverbSamples; - ALuint base, c; + ALuint base, c, c2, i; /* Process reverb for these samples. */ for(base = 0;base < SamplesToDo;) @@ -1496,31 +1479,29 @@ static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, const ALfloat delta = 1.0f / (ALfloat)(SamplesToDo-base); ALuint todo = minu(SamplesToDo-base, MAX_UPDATE_SAMPLES); - EAXVerbPass(State, todo, &SamplesIn[base], early, late); + memset(afmt, 0, 4*MAX_UPDATE_SAMPLES*sizeof(float)); + for(c = 0;c < 4;c++) + { + for(c2 = 0;c2 < MAX_EFFECT_CHANNELS;c2++) + { + for(i = 0;i < todo;i++) + afmt[c][i] += SamplesIn[c2][base+i] * B2A.m[c][c2]; + } + } + + EAXVerbPass(State, todo, afmt, early, late); for(c = 0;c < 4;c++) { DoMix(early[c], SamplesOut, NumChannels, State->Early.PanGain[c], State->Early.CurrentGain[c], delta, base, SamplesToDo-base, todo ); - if(State->ExtraChannels > 0) - DoMix(early[c], State->ExtraOut, State->ExtraChannels, - State->Early.PanGain[c]+NumChannels, - State->Early.CurrentGain[c]+NumChannels, delta, base, - SamplesToDo-base, todo - ); } for(c = 0;c < 4;c++) { DoMix(late[c], SamplesOut, NumChannels, State->Late.PanGain[c], State->Late.CurrentGain[c], delta, base, SamplesToDo, todo ); - if(State->ExtraChannels > 0) - DoMix(late[c], State->ExtraOut, State->ExtraChannels, - State->Late.PanGain[c]+NumChannels, - State->Late.CurrentGain[c]+NumChannels, delta, base, - SamplesToDo-base, todo - ); } base += todo; @@ -1530,9 +1511,9 @@ static ALvoid ALreverbState_processEax(ALreverbState *State, ALuint SamplesToDo, 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[0], SamplesOut, NumChannels); + ALreverbState_processEax(State, SamplesToDo, SamplesIn, SamplesOut, NumChannels); else - ALreverbState_processStandard(State, SamplesToDo, SamplesIn[0], SamplesOut, NumChannels); + ALreverbState_processStandard(State, SamplesToDo, SamplesIn, SamplesOut, NumChannels); } |