/** * Ambisonic reverb engine for the OpenAL cross platform audio library * Copyright (C) 2008-2017 by Chris Robinson and Christopher Fitzgerald. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include "alMain.h" #include "alu.h" #include "alAuxEffectSlot.h" #include "alListener.h" #include "alError.h" #include "filters/defs.h" /* This is a user config option for modifying the overall output of the reverb * effect. */ ALfloat ReverbBoost = 1.0f; /* This is the maximum number of samples processed for each inner loop * iteration. */ #define MAX_UPDATE_SAMPLES 256 /* The number of samples used for cross-faded delay lines. This can be used * to balance the compensation for abrupt line changes and attenuation due to * minimally lengthed recursive lines. Try to keep this below the device * update size. */ #define FADE_SAMPLES 128 /* The number of spatialized lines or channels to process. Four channels allows * for a 3D A-Format response. NOTE: This can't be changed without taking care * of the conversion matrices, and a few places where the length arrays are * assumed to have 4 elements. */ #define NUM_LINES 4 /* The B-Format to A-Format conversion matrix. The arrangement of rows is * deliberately chosen to align the resulting lines to their spatial opposites * (0:above front left <-> 3:above back right, 1:below front right <-> 2:below * back left). It's not quite opposite, since the A-Format results in a * tetrahedron, but it's close enough. Should the model be extended to 8-lines * in the future, true opposites can be used. */ 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 } }}; /* Converts A-Format to B-Format. */ static const aluMatrixf A2B = {{ { 0.866025403785f, 0.866025403785f, 0.866025403785f, 0.866025403785f }, { 0.866025403785f, -0.866025403785f, 0.866025403785f, -0.866025403785f }, { 0.866025403785f, -0.866025403785f, -0.866025403785f, 0.866025403785f }, { 0.866025403785f, 0.866025403785f, -0.866025403785f, -0.866025403785f } }}; static const ALfloat FadeStep = 1.0f / FADE_SAMPLES; /* The all-pass and delay lines have a variable length dependent on the * effect's density parameter, which helps alter the perceived environment * size. The size-to-density conversion is a cubed scale: * * density = min(1.0, pow(size, 3.0) / DENSITY_SCALE); * * The line lengths scale linearly with room size, so the inverse density * conversion is needed, taking the cube root of the re-scaled density to * calculate the line length multiplier: * * length_mult = max(5.0, cbrtf(density*DENSITY_SCALE)); * * The density scale below will result in a max line multiplier of 50, for an * effective size range of 5m to 50m. */ static const ALfloat DENSITY_SCALE = 125000.0f; /* All delay line lengths are specified in seconds. * * To approximate early reflections, we break them up into primary (those * arriving from the same direction as the source) and secondary (those * arriving from the opposite direction). * * The early taps decorrelate the 4-channel signal to approximate an average * room response for the primary reflections after the initial early delay. * * Given an average room dimension (d_a) and the speed of sound (c) we can * calculate the average reflection delay (r_a) regardless of listener and * source positions as: * * r_a = d_a / c * c = 343.3 * * This can extended to finding the average difference (r_d) between the * maximum (r_1) and minimum (r_0) reflection delays: * * r_0 = 2 / 3 r_a * = r_a - r_d / 2 * = r_d * r_1 = 4 / 3 r_a * = r_a + r_d / 2 * = 2 r_d * r_d = 2 / 3 r_a * = r_1 - r_0 * * As can be determined by integrating the 1D model with a source (s) and * listener (l) positioned across the dimension of length (d_a): * * r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c * * The initial taps (T_(i=0)^N) are then specified by taking a power series * that ranges between r_0 and half of r_1 less r_0: * * R_i = 2^(i / (2 N - 1)) r_d * = r_0 + (2^(i / (2 N - 1)) - 1) r_d * = r_0 + T_i * T_i = R_i - r_0 * = (2^(i / (2 N - 1)) - 1) r_d * * Assuming an average of 1m, we get the following taps: */ static const ALfloat EARLY_TAP_LENGTHS[NUM_LINES] = { 0.0000000e+0f, 2.0213520e-4f, 4.2531060e-4f, 6.7171600e-4f }; /* The early all-pass filter lengths are based on the early tap lengths: * * A_i = R_i / a * * Where a is the approximate maximum all-pass cycle limit (20). */ static const ALfloat EARLY_ALLPASS_LENGTHS[NUM_LINES] = { 9.7096800e-5f, 1.0720356e-4f, 1.1836234e-4f, 1.3068260e-4f }; /* The early delay lines are used to transform the primary reflections into * the secondary reflections. The A-format is arranged in such a way that * the channels/lines are spatially opposite: * * C_i is opposite C_(N-i-1) * * The delays of the two opposing reflections (R_i and O_i) from a source * anywhere along a particular dimension always sum to twice its full delay: * * 2 r_a = R_i + O_i * * With that in mind we can determine the delay between the two reflections * and thus specify our early line lengths (L_(i=0)^N) using: * * O_i = 2 r_a - R_(N-i-1) * L_i = O_i - R_(N-i-1) * = 2 (r_a - R_(N-i-1)) * = 2 (r_a - T_(N-i-1) - r_0) * = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1))) * * Using an average dimension of 1m, we get: */ static const ALfloat EARLY_LINE_LENGTHS[NUM_LINES] = { 5.9850400e-4f, 1.0913150e-3f, 1.5376658e-3f, 1.9419362e-3f }; /* The late all-pass filter lengths are based on the late line lengths: * * A_i = (5 / 3) L_i / r_1 */ static const ALfloat LATE_ALLPASS_LENGTHS[NUM_LINES] = { 1.6182800e-4f, 2.0389060e-4f, 2.8159360e-4f, 3.2365600e-4f }; /* The late lines are used to approximate the decaying cycle of recursive * late reflections. * * Splitting the lines in half, we start with the shortest reflection paths * (L_(i=0)^(N/2)): * * L_i = 2^(i / (N - 1)) r_d * * Then for the opposite (longest) reflection paths (L_(i=N/2)^N): * * L_i = 2 r_a - L_(i-N/2) * = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d * * For our 1m average room, we get: */ static const ALfloat LATE_LINE_LENGTHS[NUM_LINES] = { 1.9419362e-3f, 2.4466860e-3f, 3.3791220e-3f, 3.8838720e-3f }; typedef struct DelayLineI { /* The delay lines use interleaved samples, with the lengths being powers * of 2 to allow the use of bit-masking instead of a modulus for wrapping. */ ALsizei Mask; ALfloat (*Line)[NUM_LINES]; } DelayLineI; typedef struct VecAllpass { DelayLineI Delay; ALfloat Coeff; ALsizei Offset[NUM_LINES][2]; } VecAllpass; typedef struct T60Filter { /* Two filters are used to adjust the signal. One to control the low * frequencies, and one to control the high frequencies. */ ALfloat MidGain[2]; BiquadFilter HFFilter, LFFilter; } T60Filter; typedef struct EarlyReflections { /* A Gerzon vector all-pass filter is used to simulate initial diffusion. * The spread from this filter also helps smooth out the reverb tail. */ VecAllpass VecAp; /* An echo line is used to complete the second half of the early * reflections. */ DelayLineI Delay; ALsizei Offset[NUM_LINES][2]; ALfloat Coeff[NUM_LINES][2]; /* The gain for each output channel based on 3D panning. */ ALfloat CurrentGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; } EarlyReflections; typedef struct LateReverb { /* A recursive delay line is used fill in the reverb tail. */ DelayLineI Delay; ALsizei Offset[NUM_LINES][2]; /* Attenuation to compensate for the modal density and decay rate of the * late lines. */ ALfloat DensityGain[2]; /* T60 decay filters are used to simulate absorption. */ T60Filter T60[NUM_LINES]; /* A Gerzon vector all-pass filter is used to simulate diffusion. */ VecAllpass VecAp; /* The gain for each output channel based on 3D panning. */ ALfloat CurrentGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; ALfloat PanGain[NUM_LINES][MAX_OUTPUT_CHANNELS]; } LateReverb; typedef struct ALreverbState { DERIVE_FROM_TYPE(ALeffectState); /* All delay lines are allocated as a single buffer to reduce memory * fragmentation and management code. */ ALfloat *SampleBuffer; ALuint TotalSamples; /* Master effect filters */ struct { BiquadFilter Lp; BiquadFilter Hp; } Filter[NUM_LINES]; /* Core delay line (early reflections and late reverb tap from this). */ DelayLineI Delay; /* Tap points for early reflection delay. */ ALsizei EarlyDelayTap[NUM_LINES][2]; ALfloat EarlyDelayCoeff[NUM_LINES][2]; /* Tap points for late reverb feed and delay. */ ALsizei LateFeedTap; ALsizei LateDelayTap[NUM_LINES][2]; /* Coefficients for the all-pass and line scattering matrices. */ ALfloat MixX; ALfloat MixY; EarlyReflections Early; LateReverb Late; /* Indicates the cross-fade point for delay line reads [0,FADE_SAMPLES]. */ ALsizei FadeCount; /* Maximum number of samples to process at once. */ ALsizei MaxUpdate[2]; /* The current write offset for all delay lines. */ ALsizei Offset; /* Temporary storage used when processing. */ alignas(16) ALfloat TempSamples[NUM_LINES][MAX_UPDATE_SAMPLES]; alignas(16) ALfloat MixSamples[NUM_LINES][MAX_UPDATE_SAMPLES]; } ALreverbState; static ALvoid ALreverbState_Destruct(ALreverbState *State); static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device); static ALvoid ALreverbState_update(ALreverbState *State, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props); static ALvoid ALreverbState_process(ALreverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALreverbState) DEFINE_ALEFFECTSTATE_VTABLE(ALreverbState); static void ALreverbState_Construct(ALreverbState *state) { ALsizei i, j; ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALreverbState, ALeffectState, state); state->TotalSamples = 0; state->SampleBuffer = NULL; for(i = 0;i < NUM_LINES;i++) { BiquadFilter_clear(&state->Filter[i].Lp); BiquadFilter_clear(&state->Filter[i].Hp); } state->Delay.Mask = 0; state->Delay.Line = NULL; for(i = 0;i < NUM_LINES;i++) { state->EarlyDelayTap[i][0] = 0; state->EarlyDelayTap[i][1] = 0; state->EarlyDelayCoeff[i][0] = 0.0f; state->EarlyDelayCoeff[i][1] = 0.0f; } state->LateFeedTap = 0; for(i = 0;i < NUM_LINES;i++) { state->LateDelayTap[i][0] = 0; state->LateDelayTap[i][1] = 0; } state->MixX = 0.0f; state->MixY = 0.0f; state->Early.VecAp.Delay.Mask = 0; state->Early.VecAp.Delay.Line = NULL; state->Early.VecAp.Coeff = 0.0f; state->Early.Delay.Mask = 0; state->Early.Delay.Line = NULL; for(i = 0;i < NUM_LINES;i++) { state->Early.VecAp.Offset[i][0] = 0; state->Early.VecAp.Offset[i][1] = 0; state->Early.Offset[i][0] = 0; state->Early.Offset[i][1] = 0; state->Early.Coeff[i][0] = 0.0f; state->Early.Coeff[i][1] = 0.0f; } state->Late.DensityGain[0] = 0.0f; state->Late.DensityGain[1] = 0.0f; state->Late.Delay.Mask = 0; state->Late.Delay.Line = NULL; state->Late.VecAp.Delay.Mask = 0; state->Late.VecAp.Delay.Line = NULL; state->Late.VecAp.Coeff = 0.0f; for(i = 0;i < NUM_LINES;i++) { state->Late.Offset[i][0] = 0; state->Late.Offset[i][1] = 0; state->Late.VecAp.Offset[i][0] = 0; state->Late.VecAp.Offset[i][1] = 0; state->Late.T60[i].MidGain[0] = 0.0f; state->Late.T60[i].MidGain[1] = 0.0f; BiquadFilter_clear(&state->Late.T60[i].HFFilter); BiquadFilter_clear(&state->Late.T60[i].LFFilter); } for(i = 0;i < NUM_LINES;i++) { for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) { state->Early.CurrentGain[i][j] = 0.0f; state->Early.PanGain[i][j] = 0.0f; state->Late.CurrentGain[i][j] = 0.0f; state->Late.PanGain[i][j] = 0.0f; } } state->FadeCount = 0; state->MaxUpdate[0] = MAX_UPDATE_SAMPLES; state->MaxUpdate[1] = MAX_UPDATE_SAMPLES; state->Offset = 0; } static ALvoid ALreverbState_Destruct(ALreverbState *State) { al_free(State->SampleBuffer); State->SampleBuffer = NULL; ALeffectState_Destruct(STATIC_CAST(ALeffectState,State)); } /************************************** * Device Update * **************************************/ static inline ALfloat CalcDelayLengthMult(ALfloat density) { return maxf(5.0f, cbrtf(density*DENSITY_SCALE)); } /* Given the allocated sample buffer, this function updates each delay line * offset. */ static inline ALvoid RealizeLineOffset(ALfloat *sampleBuffer, DelayLineI *Delay) { union { ALfloat *f; ALfloat (*f4)[NUM_LINES]; } u; u.f = &sampleBuffer[(ptrdiff_t)Delay->Line * NUM_LINES]; Delay->Line = u.f4; } /* Calculate the length of a delay line and store its mask and offset. */ static ALuint CalcLineLength(const ALfloat length, const ptrdiff_t offset, const ALuint frequency, const ALuint extra, DelayLineI *Delay) { ALuint samples; /* All line lengths are powers of 2, calculated from their lengths in * seconds, rounded up. */ samples = float2int(ceilf(length*frequency)); samples = NextPowerOf2(samples + extra); /* All lines share a single sample buffer. */ Delay->Mask = samples - 1; Delay->Line = (ALfloat(*)[NUM_LINES])offset; /* Return the sample count for accumulation. */ return samples; } /* Calculates the delay line metrics and allocates the shared sample buffer * for all lines given the sample rate (frequency). If an allocation failure * occurs, it returns AL_FALSE. */ static ALboolean AllocLines(const ALuint frequency, ALreverbState *State) { ALuint totalSamples, i; ALfloat multiplier, length; /* All delay line lengths are calculated to accomodate the full range of * lengths given their respective paramters. */ totalSamples = 0; /* Multiplier for the maximum density value, i.e. density=1, which is * actually the least density... */ multiplier = CalcDelayLengthMult(AL_EAXREVERB_MAX_DENSITY); /* The main delay length includes the maximum early reflection delay, the * largest early tap width, the maximum late reverb delay, and the * largest late tap width. Finally, it must also be extended by the * update size (MAX_UPDATE_SAMPLES) for block processing. */ length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS[NUM_LINES-1]*multiplier + AL_EAXREVERB_MAX_LATE_REVERB_DELAY + (LATE_LINE_LENGTHS[NUM_LINES-1] - LATE_LINE_LENGTHS[0])*0.25f*multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, MAX_UPDATE_SAMPLES, &State->Delay); /* The early vector all-pass line. */ length = EARLY_ALLPASS_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Early.VecAp.Delay); /* The early reflection line. */ length = EARLY_LINE_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Early.Delay); /* The late vector all-pass line. */ length = LATE_ALLPASS_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Late.VecAp.Delay); /* The late delay lines are calculated from the largest maximum density * line length. */ length = LATE_LINE_LENGTHS[NUM_LINES-1] * multiplier; totalSamples += CalcLineLength(length, totalSamples, frequency, 0, &State->Late.Delay); if(totalSamples != State->TotalSamples) { ALfloat *newBuffer; TRACE("New reverb buffer length: %ux4 samples\n", totalSamples); newBuffer = al_calloc(16, sizeof(ALfloat[NUM_LINES]) * totalSamples); if(!newBuffer) return AL_FALSE; al_free(State->SampleBuffer); State->SampleBuffer = newBuffer; State->TotalSamples = totalSamples; } /* Update all delays to reflect the new sample buffer. */ RealizeLineOffset(State->SampleBuffer, &State->Delay); RealizeLineOffset(State->SampleBuffer, &State->Early.VecAp.Delay); RealizeLineOffset(State->SampleBuffer, &State->Early.Delay); RealizeLineOffset(State->SampleBuffer, &State->Late.VecAp.Delay); RealizeLineOffset(State->SampleBuffer, &State->Late.Delay); /* Clear the sample buffer. */ for(i = 0;i < State->TotalSamples;i++) State->SampleBuffer[i] = 0.0f; return AL_TRUE; } static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device) { ALuint frequency = Device->Frequency; ALfloat multiplier; /* Allocate the delay lines. */ if(!AllocLines(frequency, State)) return AL_FALSE; multiplier = CalcDelayLengthMult(AL_EAXREVERB_MAX_DENSITY); /* The late feed taps are set a fixed position past the latest delay tap. */ State->LateFeedTap = float2int((AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS[NUM_LINES-1]*multiplier) * frequency); return AL_TRUE; } /************************************** * Effect Update * **************************************/ /* Calculate a decay coefficient given the length of each cycle and the time * until the decay reaches -60 dB. */ static inline ALfloat CalcDecayCoeff(const ALfloat length, const ALfloat decayTime) { return powf(REVERB_DECAY_GAIN, length/decayTime); } /* Calculate a decay length from a coefficient and the time until the decay * reaches -60 dB. */ static inline ALfloat CalcDecayLength(const ALfloat coeff, const ALfloat decayTime) { return log10f(coeff) * decayTime / log10f(REVERB_DECAY_GAIN); } /* Calculate an attenuation to be applied to the input of any echo models to * compensate for modal density and decay time. */ static inline ALfloat CalcDensityGain(const ALfloat a) { /* The energy of a signal can be obtained by finding the area under the * squared signal. This takes the form of Sum(x_n^2), where x is the * amplitude for the sample n. * * Decaying feedback matches exponential decay of the form Sum(a^n), * where a is the attenuation coefficient, and n is the sample. The area * under this decay curve can be calculated as: 1 / (1 - a). * * Modifying the above equation to find the area under the squared curve * (for energy) yields: 1 / (1 - a^2). Input attenuation can then be * calculated by inverting the square root of this approximation, * yielding: 1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2). */ return sqrtf(1.0f - a*a); } /* Calculate the scattering matrix coefficients given a diffusion factor. */ static inline ALvoid CalcMatrixCoeffs(const ALfloat diffusion, ALfloat *x, ALfloat *y) { ALfloat n, t; /* The matrix is of order 4, so n is sqrt(4 - 1). */ n = sqrtf(3.0f); t = diffusion * atanf(n); /* Calculate the first mixing matrix coefficient. */ *x = cosf(t); /* Calculate the second mixing matrix coefficient. */ *y = sinf(t) / n; } /* Calculate the limited HF ratio for use with the late reverb low-pass * filters. */ static ALfloat CalcLimitedHfRatio(const ALfloat hfRatio, const ALfloat airAbsorptionGainHF, const ALfloat decayTime, const ALfloat SpeedOfSound) { ALfloat limitRatio; /* Find the attenuation due to air absorption in dB (converting delay * time to meters using the speed of sound). Then reversing the decay * equation, solve for HF ratio. The delay length is cancelled out of * the equation, so it can be calculated once for all lines. */ limitRatio = 1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) * SpeedOfSound); /* Using the limit calculated above, apply the upper bound to the HF ratio. */ return minf(limitRatio, hfRatio); } /* Calculates the 3-band T60 damping coefficients for a particular delay line * of specified length, using a combination of two shelf filter sections given * decay times for each band split at two reference frequencies. */ static void CalcT60DampingCoeffs(const ALfloat length, const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lf0norm, const ALfloat hf0norm, T60Filter *filter) { ALfloat lfGain = CalcDecayCoeff(length, lfDecayTime); ALfloat mfGain = CalcDecayCoeff(length, mfDecayTime); ALfloat hfGain = CalcDecayCoeff(length, hfDecayTime); filter->MidGain[1] = mfGain; BiquadFilter_setParams(&filter->LFFilter, BiquadType_LowShelf, lfGain/mfGain, lf0norm, calc_rcpQ_from_slope(lfGain/mfGain, 1.0f)); BiquadFilter_setParams(&filter->HFFilter, BiquadType_HighShelf, hfGain/mfGain, hf0norm, calc_rcpQ_from_slope(hfGain/mfGain, 1.0f)); } /* 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) { ALfloat multiplier, length; ALuint i; multiplier = CalcDelayLengthMult(density); /* Early reflection taps are decorrelated by means of an average room * reflection approximation described above the definition of the taps. * This approximation is linear and so the above density multiplier can * be applied to adjust the width of the taps. A single-band decay * coefficient is applied to simulate initial attenuation and absorption. * * Late reverb taps are based on the late line lengths to allow a zero- * delay path and offsets that would continue the propagation naturally * into the late lines. */ for(i = 0;i < NUM_LINES;i++) { length = earlyDelay + EARLY_TAP_LENGTHS[i]*multiplier; State->EarlyDelayTap[i][1] = float2int(length * frequency); length = EARLY_TAP_LENGTHS[i]*multiplier; State->EarlyDelayCoeff[i][1] = CalcDecayCoeff(length, decayTime); length = lateDelay + (LATE_LINE_LENGTHS[i] - LATE_LINE_LENGTHS[0])*0.25f*multiplier; State->LateDelayTap[i][1] = State->LateFeedTap + float2int(length * frequency); } } /* Update the early reflection line lengths and gain coefficients. */ static ALvoid UpdateEarlyLines(const ALfloat density, const ALfloat diffusion, const ALfloat decayTime, const ALuint frequency, EarlyReflections *Early) { ALfloat multiplier, length; ALsizei i; multiplier = CalcDelayLengthMult(density); /* Calculate the all-pass feed-back/forward coefficient. */ Early->VecAp.Coeff = sqrtf(0.5f) * powf(diffusion, 2.0f); for(i = 0;i < NUM_LINES;i++) { /* Calculate the length (in seconds) of each all-pass line. */ length = EARLY_ALLPASS_LENGTHS[i] * multiplier; /* Calculate the delay offset for each all-pass line. */ Early->VecAp.Offset[i][1] = float2int(length * frequency); /* Calculate the length (in seconds) of each delay line. */ length = EARLY_LINE_LENGTHS[i] * multiplier; /* Calculate the delay offset for each delay line. */ Early->Offset[i][1] = float2int(length * frequency); /* Calculate the gain (coefficient) for each line. */ Early->Coeff[i][1] = CalcDecayCoeff(length, decayTime); } } /* Update the late reverb line lengths and T60 coefficients. */ static ALvoid UpdateLateLines(const ALfloat density, const ALfloat diffusion, const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lf0norm, const ALfloat hf0norm, const ALuint frequency, LateReverb *Late) { /* Scaling factor to convert the normalized reference frequencies from * representing 0...freq to 0...max_reference. */ const ALfloat norm_weight_factor = (ALfloat)frequency / AL_EAXREVERB_MAX_HFREFERENCE; ALfloat multiplier, length, bandWeights[3]; ALsizei i; /* To compensate for changes in modal density and decay time of the late * reverb signal, the input is attenuated based on the maximal energy of * the outgoing signal. This approximation is used to keep the apparent * energy of the signal equal for all ranges of density and decay time. * * The average length of the delay lines is used to calculate the * attenuation coefficient. */ multiplier = CalcDelayLengthMult(density); length = (LATE_LINE_LENGTHS[0] + LATE_LINE_LENGTHS[1] + LATE_LINE_LENGTHS[2] + LATE_LINE_LENGTHS[3]) / 4.0f * multiplier; length += (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] + LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f * multiplier; /* The density gain calculation uses an average decay time weighted by * approximate bandwidth. This attempts to compensate for losses of energy * that reduce decay time due to scattering into highly attenuated bands. */ bandWeights[0] = lf0norm*norm_weight_factor; bandWeights[1] = hf0norm*norm_weight_factor - lf0norm*norm_weight_factor; bandWeights[2] = 1.0f - hf0norm*norm_weight_factor; Late->DensityGain[1] = CalcDensityGain( CalcDecayCoeff(length, bandWeights[0]*lfDecayTime + bandWeights[1]*mfDecayTime + bandWeights[2]*hfDecayTime ) ); /* Calculate the all-pass feed-back/forward coefficient. */ Late->VecAp.Coeff = sqrtf(0.5f) * powf(diffusion, 2.0f); for(i = 0;i < NUM_LINES;i++) { /* Calculate the length (in seconds) of each all-pass line. */ length = LATE_ALLPASS_LENGTHS[i] * multiplier; /* Calculate the delay offset for each all-pass line. */ Late->VecAp.Offset[i][1] = float2int(length * frequency); /* Calculate the length (in seconds) of each delay line. */ length = LATE_LINE_LENGTHS[i] * multiplier; /* Calculate the delay offset for each delay line. */ Late->Offset[i][1] = float2int(length*frequency + 0.5f); /* Approximate the absorption that the vector all-pass would exhibit * given the current diffusion so we don't have to process a full T60 * filter for each of its four lines. */ length += lerp(LATE_ALLPASS_LENGTHS[i], (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] + LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f, diffusion) * multiplier; /* Calculate the T60 damping coefficients for each line. */ CalcT60DampingCoeffs(length, lfDecayTime, mfDecayTime, hfDecayTime, lf0norm, hf0norm, &Late->T60[i]); } } /* Creates a transform matrix given a reverb vector. The vector pans the reverb * reflections toward the given direction, using its magnitude (up to 1) as a * focal strength. This function results in a B-Format transformation matrix * that spatially focuses the signal in the desired direction. */ static aluMatrixf GetTransformFromVector(const ALfloat *vec) { aluMatrixf focus; ALfloat norm[3]; ALfloat mag; /* Normalize the panning vector according to the N3D scale, which has an * extra sqrt(3) term on the directional components. Converting from OpenAL * to B-Format also requires negating X (ACN 1) and Z (ACN 3). Note however * that the reverb panning vectors use left-handed coordinates, unlike the * rest of OpenAL which use right-handed. This is fixed by negating Z, * which cancels out with the B-Format Z negation. */ mag = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]); if(mag > 1.0f) { norm[0] = vec[0] / mag * -SQRTF_3; norm[1] = vec[1] / mag * SQRTF_3; norm[2] = vec[2] / mag * SQRTF_3; mag = 1.0f; } else { /* If the magnitude is less than or equal to 1, just apply the sqrt(3) * term. There's no need to renormalize the magnitude since it would * just be reapplied in the matrix. */ norm[0] = vec[0] * -SQRTF_3; norm[1] = vec[1] * SQRTF_3; norm[2] = vec[2] * SQRTF_3; } aluMatrixfSet(&focus, 1.0f, 0.0f, 0.0f, 0.0f, norm[0], 1.0f-mag, 0.0f, 0.0f, norm[1], 0.0f, 1.0f-mag, 0.0f, norm[2], 0.0f, 0.0f, 1.0f-mag ); return focus; } /* Update the early and late 3D panning gains. */ static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, const ALfloat earlyGain, const ALfloat lateGain, ALreverbState *State) { aluMatrixf transform, rot; ALsizei i; STATIC_CAST(ALeffectState,State)->OutBuffer = Device->FOAOut.Buffer; STATIC_CAST(ALeffectState,State)->OutChannels = Device->FOAOut.NumChannels; /* Note: _res is 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[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) /* Create a matrix that first converts A-Format to B-Format, then * transforms the B-Format signal according to the panning vector. */ rot = GetTransformFromVector(ReflectionsPan); MATRIX_MULT(transform, rot, A2B); memset(&State->Early.PanGain, 0, sizeof(State->Early.PanGain)); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputeFirstOrderGains(&Device->FOAOut, transform.m[i], earlyGain, State->Early.PanGain[i]); rot = GetTransformFromVector(LateReverbPan); MATRIX_MULT(transform, rot, A2B); memset(&State->Late.PanGain, 0, sizeof(State->Late.PanGain)); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputeFirstOrderGains(&Device->FOAOut, transform.m[i], lateGain, State->Late.PanGain[i]); #undef MATRIX_MULT } static ALvoid ALreverbState_update(ALreverbState *State, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props) { const ALCdevice *Device = Context->Device; const ALlistener *Listener = Context->Listener; ALuint frequency = Device->Frequency; ALfloat lf0norm, hf0norm, hfRatio; ALfloat lfDecayTime, hfDecayTime; ALfloat gain, gainlf, gainhf; ALsizei i; /* Calculate the master filters */ hf0norm = minf(props->Reverb.HFReference / frequency, 0.49f); /* Restrict the filter gains from going below -60dB to keep the filter from * killing most of the signal. */ gainhf = maxf(props->Reverb.GainHF, 0.001f); BiquadFilter_setParams(&State->Filter[0].Lp, BiquadType_HighShelf, gainhf, hf0norm, calc_rcpQ_from_slope(gainhf, 1.0f)); lf0norm = minf(props->Reverb.LFReference / frequency, 0.49f); gainlf = maxf(props->Reverb.GainLF, 0.001f); BiquadFilter_setParams(&State->Filter[0].Hp, BiquadType_LowShelf, gainlf, lf0norm, calc_rcpQ_from_slope(gainlf, 1.0f)); for(i = 1;i < NUM_LINES;i++) { BiquadFilter_copyParams(&State->Filter[i].Lp, &State->Filter[0].Lp); BiquadFilter_copyParams(&State->Filter[i].Hp, &State->Filter[0].Hp); } /* Update the main effect delay and associated taps. */ UpdateDelayLine(props->Reverb.ReflectionsDelay, props->Reverb.LateReverbDelay, props->Reverb.Density, props->Reverb.DecayTime, frequency, State); /* Update the early lines. */ UpdateEarlyLines(props->Reverb.Density, props->Reverb.Diffusion, props->Reverb.DecayTime, frequency, &State->Early); /* Get the mixing matrix coefficients. */ CalcMatrixCoeffs(props->Reverb.Diffusion, &State->MixX, &State->MixY); /* If the HF limit parameter is flagged, calculate an appropriate limit * based on the air absorption parameter. */ hfRatio = props->Reverb.DecayHFRatio; if(props->Reverb.DecayHFLimit && props->Reverb.AirAbsorptionGainHF < 1.0f) hfRatio = CalcLimitedHfRatio(hfRatio, props->Reverb.AirAbsorptionGainHF, props->Reverb.DecayTime, Listener->Params.ReverbSpeedOfSound ); /* Calculate the LF/HF decay times. */ lfDecayTime = clampf(props->Reverb.DecayTime * props->Reverb.DecayLFRatio, AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME); hfDecayTime = clampf(props->Reverb.DecayTime * hfRatio, AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME); /* Update the late lines. */ UpdateLateLines(props->Reverb.Density, props->Reverb.Diffusion, lfDecayTime, props->Reverb.DecayTime, hfDecayTime, lf0norm, hf0norm, frequency, &State->Late ); /* Update early and late 3D panning. */ gain = props->Reverb.Gain * Slot->Params.Gain * ReverbBoost; Update3DPanning(Device, props->Reverb.ReflectionsPan, props->Reverb.LateReverbPan, props->Reverb.ReflectionsGain*gain, props->Reverb.LateReverbGain*gain, State); /* Calculate the max update size from the smallest relevant delay. */ State->MaxUpdate[1] = mini(MAX_UPDATE_SAMPLES, mini(State->Early.Offset[0][1], State->Late.Offset[0][1]) ); /* Determine if delay-line cross-fading is required. TODO: Add some fuzz * for the float comparisons? The math should be stable enough that the * result should be the same if nothing's changed, and changes in the float * values should (though may not always) be matched by changes in delay * offsets. */ if(State->Late.DensityGain[1] != State->Late.DensityGain[0]) State->FadeCount = 0; else for(i = 0;i < NUM_LINES;i++) { if(State->EarlyDelayTap[i][1] != State->EarlyDelayTap[i][0] || State->EarlyDelayCoeff[i][1] != State->EarlyDelayCoeff[i][0] || State->Early.VecAp.Offset[i][1] != State->Early.VecAp.Offset[i][0] || State->Early.Offset[i][1] != State->Early.Offset[i][0] || State->Early.Coeff[i][1] != State->Early.Coeff[i][0] || State->LateDelayTap[i][1] != State->LateDelayTap[i][0] || State->Late.VecAp.Offset[i][1] != State->Late.VecAp.Offset[i][0] || State->Late.Offset[i][1] != State->Late.Offset[i][0] || State->Late.T60[i].MidGain[1] != State->Late.T60[i].MidGain[0]) { State->FadeCount = 0; break; } } } /************************************** * Effect Processing * **************************************/ /* Basic delay line input/output routines. */ static inline ALfloat DelayLineOut(const DelayLineI *Delay, const ALsizei offset, const ALsizei c) { return Delay->Line[offset&Delay->Mask][c]; } /* Cross-faded delay line output routine. Instead of interpolating the * offsets, this interpolates (cross-fades) the outputs at each offset. */ static inline ALfloat FadedDelayLineOut(const DelayLineI *Delay, const ALsizei off0, const ALsizei off1, const ALsizei c, const ALfloat sc0, const ALfloat sc1) { return Delay->Line[off0&Delay->Mask][c]*sc0 + Delay->Line[off1&Delay->Mask][c]*sc1; } static inline void DelayLineIn(const DelayLineI *Delay, ALsizei offset, const ALsizei c, const ALfloat *restrict in, ALsizei count) { ALsizei i; for(i = 0;i < count;i++) Delay->Line[(offset++)&Delay->Mask][c] = *(in++); } /* 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. * * The matrix is derived from a skew-symmetric matrix to form a 4D rotation * matrix with a single unitary rotational parameter: * * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2 * [ -a, d, c, -b ] * [ -b, -c, d, a ] * [ -c, b, -a, d ] * * The rotation is constructed from the effect's diffusion parameter, * yielding: * * 1 = x^2 + 3 y^2 * * Where a, b, and c are the coefficient y with differing signs, and d is the * coefficient x. The final matrix is thus: * * [ x, y, -y, y ] n = sqrt(matrix_order - 1) * [ -y, x, y, y ] t = diffusion_parameter * atan(n) * [ y, -y, x, y ] x = cos(t) * [ -y, -y, -y, x ] y = sin(t) / n * * Any square orthogonal matrix with an order that is a power of two will * work (where ^T is transpose, ^-1 is inverse): * * M^T = M^-1 * * Using that knowledge, finding an appropriate matrix can be accomplished * naively by searching all combinations of: * * M = D + S - S^T * * Where D is a diagonal matrix (of x), and S is a triangular matrix (of y) * whose combination of signs are being iterated. */ static inline void VectorPartialScatter(ALfloat *restrict out, const ALfloat *restrict in, const ALfloat xCoeff, const ALfloat yCoeff) { out[0] = xCoeff*in[0] + yCoeff*( in[1] + -in[2] + in[3]); out[1] = xCoeff*in[1] + yCoeff*(-in[0] + in[2] + in[3]); out[2] = xCoeff*in[2] + yCoeff*( in[0] + -in[1] + in[3]); out[3] = xCoeff*in[3] + yCoeff*(-in[0] + -in[1] + -in[2] ); } #define VectorScatterDelayIn(delay, o, in, xcoeff, ycoeff) \ VectorPartialScatter((delay)->Line[(o)&(delay)->Mask], in, xcoeff, ycoeff) /* Utilizes the above, but reverses the input channels. */ static inline void VectorScatterRevDelayIn(const DelayLineI *Delay, ALint offset, const ALfloat xCoeff, const ALfloat yCoeff, const ALfloat (*restrict in)[MAX_UPDATE_SAMPLES], const ALsizei count) { const DelayLineI delay = *Delay; ALsizei i, j; for(i = 0;i < count;++i) { ALfloat f[NUM_LINES]; for(j = 0;j < NUM_LINES;j++) f[NUM_LINES-1-j] = in[j][i]; VectorScatterDelayIn(&delay, offset++, f, xCoeff, yCoeff); } } /* This applies a Gerzon multiple-in/multiple-out (MIMO) vector all-pass * filter to the 4-line input. * * It works by vectorizing a regular all-pass filter and replacing the delay * element with a scattering matrix (like the one above) and a diagonal * matrix of delay elements. * * Two static specializations are used for transitional (cross-faded) delay * line processing and non-transitional processing. */ static void VectorAllpass_Unfaded(ALfloat (*restrict samples)[MAX_UPDATE_SAMPLES], ALsizei offset, const ALfloat xCoeff, const ALfloat yCoeff, ALsizei todo, VecAllpass *Vap) { const DelayLineI delay = Vap->Delay; const ALfloat feedCoeff = Vap->Coeff; ALsizei vap_offset[NUM_LINES]; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) vap_offset[j] = offset-Vap->Offset[j][0]; for(i = 0;i < todo;i++) { ALfloat f[NUM_LINES]; for(j = 0;j < NUM_LINES;j++) { ALfloat input = samples[j][i]; ALfloat out = DelayLineOut(&delay, vap_offset[j]++, j) - feedCoeff*input; f[j] = input + feedCoeff*out; samples[j][i] = out; } VectorScatterDelayIn(&delay, offset, f, xCoeff, yCoeff); ++offset; } } static void VectorAllpass_Faded(ALfloat (*restrict samples)[MAX_UPDATE_SAMPLES], ALsizei offset, const ALfloat xCoeff, const ALfloat yCoeff, ALfloat fade, ALsizei todo, VecAllpass *Vap) { const DelayLineI delay = Vap->Delay; const ALfloat feedCoeff = Vap->Coeff; ALsizei vap_offset[NUM_LINES][2]; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) { vap_offset[j][0] = offset-Vap->Offset[j][0]; vap_offset[j][1] = offset-Vap->Offset[j][1]; } for(i = 0;i < todo;i++) { ALfloat f[NUM_LINES]; for(j = 0;j < NUM_LINES;j++) { ALfloat input = samples[j][i]; ALfloat out = FadedDelayLineOut(&delay, vap_offset[j][0]++, vap_offset[j][1]++, j, 1.0f-fade, fade ) - feedCoeff*input; f[j] = input + feedCoeff*out; samples[j][i] = out; } fade += FadeStep; VectorScatterDelayIn(&delay, offset, f, xCoeff, yCoeff); ++offset; } } /* This generates early reflections. * * This is done by obtaining the primary reflections (those arriving from the * same direction as the source) from the main delay line. These are * attenuated and all-pass filtered (based on the diffusion parameter). * * The early lines are then fed in reverse (according to the approximately * opposite spatial location of the A-Format lines) to create the secondary * reflections (those arriving from the opposite direction as the source). * * The early response is then completed by combining the primary reflections * with the delayed and attenuated output from the early lines. * * Finally, the early response is reversed, scattered (based on diffusion), * and fed into the late reverb section of the main delay line. * * Two static specializations are used for transitional (cross-faded) delay * line processing and non-transitional processing. */ static void EarlyReflection_Unfaded(ALreverbState *State, ALsizei offset, const ALsizei todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI early_delay = State->Early.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei late_feed_tap; ALsizei i, j; ASSUME(todo > 0); /* First, load decorrelated samples from the main delay line as the primary * reflections. */ for(j = 0;j < NUM_LINES;j++) { ALsizei early_delay_tap = offset - State->EarlyDelayTap[j][0]; ALfloat coeff = State->EarlyDelayCoeff[j][0]; for(i = 0;i < todo;i++) temps[j][i] = DelayLineOut(&main_delay, early_delay_tap++, j) * coeff; } /* Apply a vector all-pass, to help color the initial reflections based on * the diffusion strength. */ VectorAllpass_Unfaded(temps, offset, mixX, mixY, todo, &State->Early.VecAp); /* Apply a delay and bounce to generate secondary reflections, combine with * the primary reflections and write out the result for mixing. */ for(j = 0;j < NUM_LINES;j++) { ALint early_feedb_tap = offset - State->Early.Offset[j][0]; ALfloat early_feedb_coeff = State->Early.Coeff[j][0]; for(i = 0;i < todo;i++) out[j][i] = DelayLineOut(&early_delay, early_feedb_tap++, j)*early_feedb_coeff + temps[j][i]; } for(j = 0;j < NUM_LINES;j++) DelayLineIn(&early_delay, offset, NUM_LINES-1-j, temps[j], todo); /* Also write the result back to the main delay line for the late reverb * stage to pick up at the appropriate time, appplying a scatter and * bounce to improve the initial diffusion in the late reverb. */ late_feed_tap = offset - State->LateFeedTap; VectorScatterRevDelayIn(&main_delay, late_feed_tap, mixX, mixY, out, todo); } static void EarlyReflection_Faded(ALreverbState *State, ALsizei offset, const ALsizei todo, const ALfloat fade, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI early_delay = State->Early.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei late_feed_tap; ALfloat fadeCount; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) { ALsizei early_delay_tap0 = offset - State->EarlyDelayTap[j][0]; ALsizei early_delay_tap1 = offset - State->EarlyDelayTap[j][1]; ALfloat oldCoeff = State->EarlyDelayCoeff[j][0]; ALfloat oldCoeffStep = -oldCoeff / FADE_SAMPLES; ALfloat newCoeffStep = State->EarlyDelayCoeff[j][1] / FADE_SAMPLES; fadeCount = fade * FADE_SAMPLES; for(i = 0;i < todo;i++) { const ALfloat fade0 = oldCoeff + oldCoeffStep*fadeCount; const ALfloat fade1 = newCoeffStep*fadeCount; temps[j][i] = FadedDelayLineOut(&main_delay, early_delay_tap0++, early_delay_tap1++, j, fade0, fade1 ); fadeCount += 1.0f; } } VectorAllpass_Faded(temps, offset, mixX, mixY, fade, todo, &State->Early.VecAp); for(j = 0;j < NUM_LINES;j++) { ALint feedb_tap0 = offset - State->Early.Offset[j][0]; ALint feedb_tap1 = offset - State->Early.Offset[j][1]; ALfloat feedb_oldCoeff = State->Early.Coeff[j][0]; ALfloat feedb_oldCoeffStep = -feedb_oldCoeff / FADE_SAMPLES; ALfloat feedb_newCoeffStep = State->Early.Coeff[j][1] / FADE_SAMPLES; fadeCount = fade * FADE_SAMPLES; for(i = 0;i < todo;i++) { const ALfloat fade0 = feedb_oldCoeff + feedb_oldCoeffStep*fadeCount; const ALfloat fade1 = feedb_newCoeffStep*fadeCount; out[j][i] = FadedDelayLineOut(&early_delay, feedb_tap0++, feedb_tap1++, j, fade0, fade1 ) + temps[j][i]; fadeCount += 1.0f; } } for(j = 0;j < NUM_LINES;j++) DelayLineIn(&early_delay, offset, NUM_LINES-1-j, temps[j], todo); late_feed_tap = offset - State->LateFeedTap; VectorScatterRevDelayIn(&main_delay, late_feed_tap, mixX, mixY, out, todo); } /* Applies the two T60 damping filter sections. */ static inline void LateT60Filter(ALfloat *restrict samples, const ALsizei todo, T60Filter *filter) { ALfloat temp[MAX_UPDATE_SAMPLES]; BiquadFilter_process(&filter->HFFilter, temp, samples, todo); BiquadFilter_process(&filter->LFFilter, samples, temp, todo); } /* This generates the reverb tail using a modified feed-back delay network * (FDN). * * Results from the early reflections are mixed with the output from the late * delay lines. * * The late response is then completed by T60 and all-pass filtering the mix. * * Finally, the lines are reversed (so they feed their opposite directions) * and scattered with the FDN matrix before re-feeding the delay lines. * * Two variations are made, one for for transitional (cross-faded) delay line * processing and one for non-transitional processing. */ static void LateReverb_Unfaded(ALreverbState *State, ALsizei offset, const ALsizei todo, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI late_delay = State->Late.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei i, j; ASSUME(todo > 0); /* First, load decorrelated samples from the main and feedback delay lines. * Filter the signal to apply its frequency-dependent decay. */ for(j = 0;j < NUM_LINES;j++) { ALsizei late_delay_tap = offset - State->LateDelayTap[j][0]; ALsizei late_feedb_tap = offset - State->Late.Offset[j][0]; ALfloat midGain = State->Late.T60[j].MidGain[0]; const ALfloat densityGain = State->Late.DensityGain[0] * midGain; for(i = 0;i < todo;i++) temps[j][i] = DelayLineOut(&main_delay, late_delay_tap++, j)*densityGain + DelayLineOut(&late_delay, late_feedb_tap++, j)*midGain; LateT60Filter(temps[j], todo, &State->Late.T60[j]); } /* Apply a vector all-pass to improve micro-surface diffusion, and write * out the results for mixing. */ VectorAllpass_Unfaded(temps, offset, mixX, mixY, todo, &State->Late.VecAp); for(j = 0;j < NUM_LINES;j++) memcpy(out[j], temps[j], todo*sizeof(ALfloat)); /* Finally, scatter and bounce the results to refeed the feedback buffer. */ VectorScatterRevDelayIn(&late_delay, offset, mixX, mixY, out, todo); } static void LateReverb_Faded(ALreverbState *State, ALsizei offset, const ALsizei todo, const ALfloat fade, ALfloat (*restrict out)[MAX_UPDATE_SAMPLES]) { ALfloat (*restrict temps)[MAX_UPDATE_SAMPLES] = State->TempSamples; const DelayLineI late_delay = State->Late.Delay; const DelayLineI main_delay = State->Delay; const ALfloat mixX = State->MixX; const ALfloat mixY = State->MixY; ALsizei i, j; ASSUME(todo > 0); for(j = 0;j < NUM_LINES;j++) { const ALfloat oldMidGain = State->Late.T60[j].MidGain[0]; const ALfloat midGain = State->Late.T60[j].MidGain[1]; const ALfloat oldMidStep = -oldMidGain / FADE_SAMPLES; const ALfloat midStep = midGain / FADE_SAMPLES; const ALfloat oldDensityGain = State->Late.DensityGain[0] * oldMidGain; const ALfloat densityGain = State->Late.DensityGain[1] * midGain; const ALfloat oldDensityStep = -oldDensityGain / FADE_SAMPLES; const ALfloat densityStep = densityGain / FADE_SAMPLES; ALsizei late_delay_tap0 = offset - State->LateDelayTap[j][0]; ALsizei late_delay_tap1 = offset - State->LateDelayTap[j][1]; ALsizei late_feedb_tap0 = offset - State->Late.Offset[j][0]; ALsizei late_feedb_tap1 = offset - State->Late.Offset[j][1]; ALfloat fadeCount = fade * FADE_SAMPLES; for(i = 0;i < todo;i++) { const ALfloat fade0 = oldDensityGain + oldDensityStep*fadeCount; const ALfloat fade1 = densityStep*fadeCount; const ALfloat gfade0 = oldMidGain + oldMidStep*fadeCount; const ALfloat gfade1 = midStep*fadeCount; temps[j][i] = FadedDelayLineOut(&main_delay, late_delay_tap0++, late_delay_tap1++, j, fade0, fade1) + FadedDelayLineOut(&late_delay, late_feedb_tap0++, late_feedb_tap1++, j, gfade0, gfade1); fadeCount += 1.0f; } LateT60Filter(temps[j], todo, &State->Late.T60[j]); } VectorAllpass_Faded(temps, offset, mixX, mixY, fade, todo, &State->Late.VecAp); for(j = 0;j < NUM_LINES;j++) memcpy(out[j], temps[j], todo*sizeof(ALfloat)); VectorScatterRevDelayIn(&late_delay, offset, mixX, mixY, temps, todo); } static ALvoid ALreverbState_process(ALreverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { ALfloat (*restrict afmt)[MAX_UPDATE_SAMPLES] = State->TempSamples; ALfloat (*restrict samples)[MAX_UPDATE_SAMPLES] = State->MixSamples; ALsizei fadeCount = State->FadeCount; ALsizei offset = State->Offset; ALsizei base, c; /* Process reverb for these samples. */ for(base = 0;base < SamplesToDo;) { ALsizei todo = SamplesToDo - base; /* If cross-fading, don't do more samples than there are to fade. */ if(FADE_SAMPLES-fadeCount > 0) { todo = mini(todo, FADE_SAMPLES-fadeCount); todo = mini(todo, State->MaxUpdate[0]); } todo = mini(todo, State->MaxUpdate[1]); /* If this is not the final update, ensure the update size is a * multiple of 4 for the SIMD mixers. */ if(todo < SamplesToDo-base) todo &= ~3; /* Convert B-Format to A-Format for processing. */ memset(afmt, 0, sizeof(*afmt)*NUM_LINES); for(c = 0;c < NUM_LINES;c++) MixRowSamples(afmt[c], B2A.m[c], SamplesIn, MAX_EFFECT_CHANNELS, base, todo ); /* Process the samples for reverb. */ for(c = 0;c < NUM_LINES;c++) { /* Band-pass the incoming samples. */ BiquadFilter_process(&State->Filter[c].Lp, samples[0], afmt[c], todo); BiquadFilter_process(&State->Filter[c].Hp, samples[1], samples[0], todo); /* Feed the initial delay line. */ DelayLineIn(&State->Delay, offset, c, samples[1], todo); } if(UNLIKELY(fadeCount < FADE_SAMPLES)) { ALfloat fade = (ALfloat)fadeCount / FADE_SAMPLES; /* Generate early reflections. */ EarlyReflection_Faded(State, offset, todo, fade, samples); /* Mix the A-Format results to output, implicitly converting back * to B-Format. */ for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Early.CurrentGain[c], State->Early.PanGain[c], SamplesToDo-base, base, todo ); /* Generate and mix late reverb. */ LateReverb_Faded(State, offset, todo, fade, samples); for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Late.CurrentGain[c], State->Late.PanGain[c], SamplesToDo-base, base, todo ); /* Step fading forward. */ fadeCount += todo; if(LIKELY(fadeCount >= FADE_SAMPLES)) { /* Update the cross-fading delay line taps. */ fadeCount = FADE_SAMPLES; for(c = 0;c < NUM_LINES;c++) { State->EarlyDelayTap[c][0] = State->EarlyDelayTap[c][1]; State->EarlyDelayCoeff[c][0] = State->EarlyDelayCoeff[c][1]; State->Early.VecAp.Offset[c][0] = State->Early.VecAp.Offset[c][1]; State->Early.Offset[c][0] = State->Early.Offset[c][1]; State->Early.Coeff[c][0] = State->Early.Coeff[c][1]; State->LateDelayTap[c][0] = State->LateDelayTap[c][1]; State->Late.VecAp.Offset[c][0] = State->Late.VecAp.Offset[c][1]; State->Late.Offset[c][0] = State->Late.Offset[c][1]; State->Late.T60[c].MidGain[0] = State->Late.T60[c].MidGain[1]; } State->Late.DensityGain[0] = State->Late.DensityGain[1]; State->MaxUpdate[0] = State->MaxUpdate[1]; } } else { /* Generate and mix early reflections. */ EarlyReflection_Unfaded(State, offset, todo, samples); for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Early.CurrentGain[c], State->Early.PanGain[c], SamplesToDo-base, base, todo ); /* Generate and mix late reverb. */ LateReverb_Unfaded(State, offset, todo, samples); for(c = 0;c < NUM_LINES;c++) MixSamples(samples[c], NumChannels, SamplesOut, State->Late.CurrentGain[c], State->Late.PanGain[c], SamplesToDo-base, base, todo ); } /* Step all delays forward. */ offset += todo; base += todo; } State->Offset = offset; State->FadeCount = fadeCount; } typedef struct ReverbStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } ReverbStateFactory; static ALeffectState *ReverbStateFactory_create(ReverbStateFactory* UNUSED(factory)) { ALreverbState *state; NEW_OBJ0(state, ALreverbState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(ReverbStateFactory); EffectStateFactory *ReverbStateFactory_getFactory(void) { static ReverbStateFactory ReverbFactory = { { GET_VTABLE2(ReverbStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &ReverbFactory); } void ALeaxreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DECAY_HFLIMIT: if(!(val >= AL_EAXREVERB_MIN_DECAY_HFLIMIT && val <= AL_EAXREVERB_MAX_DECAY_HFLIMIT)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay hflimit out of range"); props->Reverb.DecayHFLimit = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb integer property 0x%04x", param); } } void ALeaxreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALeaxreverb_setParami(effect, context, param, vals[0]); } void ALeaxreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DENSITY: if(!(val >= AL_EAXREVERB_MIN_DENSITY && val <= AL_EAXREVERB_MAX_DENSITY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb density out of range"); props->Reverb.Density = val; break; case AL_EAXREVERB_DIFFUSION: if(!(val >= AL_EAXREVERB_MIN_DIFFUSION && val <= AL_EAXREVERB_MAX_DIFFUSION)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb diffusion out of range"); props->Reverb.Diffusion = val; break; case AL_EAXREVERB_GAIN: if(!(val >= AL_EAXREVERB_MIN_GAIN && val <= AL_EAXREVERB_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gain out of range"); props->Reverb.Gain = val; break; case AL_EAXREVERB_GAINHF: if(!(val >= AL_EAXREVERB_MIN_GAINHF && val <= AL_EAXREVERB_MAX_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gainhf out of range"); props->Reverb.GainHF = val; break; case AL_EAXREVERB_GAINLF: if(!(val >= AL_EAXREVERB_MIN_GAINLF && val <= AL_EAXREVERB_MAX_GAINLF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb gainlf out of range"); props->Reverb.GainLF = val; break; case AL_EAXREVERB_DECAY_TIME: if(!(val >= AL_EAXREVERB_MIN_DECAY_TIME && val <= AL_EAXREVERB_MAX_DECAY_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay time out of range"); props->Reverb.DecayTime = val; break; case AL_EAXREVERB_DECAY_HFRATIO: if(!(val >= AL_EAXREVERB_MIN_DECAY_HFRATIO && val <= AL_EAXREVERB_MAX_DECAY_HFRATIO)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay hfratio out of range"); props->Reverb.DecayHFRatio = val; break; case AL_EAXREVERB_DECAY_LFRATIO: if(!(val >= AL_EAXREVERB_MIN_DECAY_LFRATIO && val <= AL_EAXREVERB_MAX_DECAY_LFRATIO)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb decay lfratio out of range"); props->Reverb.DecayLFRatio = val; break; case AL_EAXREVERB_REFLECTIONS_GAIN: if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_GAIN && val <= AL_EAXREVERB_MAX_REFLECTIONS_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections gain out of range"); props->Reverb.ReflectionsGain = val; break; case AL_EAXREVERB_REFLECTIONS_DELAY: if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_DELAY && val <= AL_EAXREVERB_MAX_REFLECTIONS_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections delay out of range"); props->Reverb.ReflectionsDelay = val; break; case AL_EAXREVERB_LATE_REVERB_GAIN: if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_GAIN && val <= AL_EAXREVERB_MAX_LATE_REVERB_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb gain out of range"); props->Reverb.LateReverbGain = val; break; case AL_EAXREVERB_LATE_REVERB_DELAY: if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_DELAY && val <= AL_EAXREVERB_MAX_LATE_REVERB_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb delay out of range"); props->Reverb.LateReverbDelay = val; break; case AL_EAXREVERB_AIR_ABSORPTION_GAINHF: if(!(val >= AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb air absorption gainhf out of range"); props->Reverb.AirAbsorptionGainHF = val; break; case AL_EAXREVERB_ECHO_TIME: if(!(val >= AL_EAXREVERB_MIN_ECHO_TIME && val <= AL_EAXREVERB_MAX_ECHO_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb echo time out of range"); props->Reverb.EchoTime = val; break; case AL_EAXREVERB_ECHO_DEPTH: if(!(val >= AL_EAXREVERB_MIN_ECHO_DEPTH && val <= AL_EAXREVERB_MAX_ECHO_DEPTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb echo depth out of range"); props->Reverb.EchoDepth = val; break; case AL_EAXREVERB_MODULATION_TIME: if(!(val >= AL_EAXREVERB_MIN_MODULATION_TIME && val <= AL_EAXREVERB_MAX_MODULATION_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb modulation time out of range"); props->Reverb.ModulationTime = val; break; case AL_EAXREVERB_MODULATION_DEPTH: if(!(val >= AL_EAXREVERB_MIN_MODULATION_DEPTH && val <= AL_EAXREVERB_MAX_MODULATION_DEPTH)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb modulation depth out of range"); props->Reverb.ModulationDepth = val; break; case AL_EAXREVERB_HFREFERENCE: if(!(val >= AL_EAXREVERB_MIN_HFREFERENCE && val <= AL_EAXREVERB_MAX_HFREFERENCE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb hfreference out of range"); props->Reverb.HFReference = val; break; case AL_EAXREVERB_LFREFERENCE: if(!(val >= AL_EAXREVERB_MIN_LFREFERENCE && val <= AL_EAXREVERB_MAX_LFREFERENCE)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb lfreference out of range"); props->Reverb.LFReference = val; break; case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR: if(!(val >= AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR)) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb room rolloff factor out of range"); props->Reverb.RoomRolloffFactor = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb float property 0x%04x", param); } } void ALeaxreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_REFLECTIONS_PAN: if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2]))) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb reflections pan out of range"); props->Reverb.ReflectionsPan[0] = vals[0]; props->Reverb.ReflectionsPan[1] = vals[1]; props->Reverb.ReflectionsPan[2] = vals[2]; break; case AL_EAXREVERB_LATE_REVERB_PAN: if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2]))) SETERR_RETURN(context, AL_INVALID_VALUE,, "EAX Reverb late reverb pan out of range"); props->Reverb.LateReverbPan[0] = vals[0]; props->Reverb.LateReverbPan[1] = vals[1]; props->Reverb.LateReverbPan[2] = vals[2]; break; default: ALeaxreverb_setParamf(effect, context, param, vals[0]); break; } } void ALeaxreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DECAY_HFLIMIT: *val = props->Reverb.DecayHFLimit; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb integer property 0x%04x", param); } } void ALeaxreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALeaxreverb_getParami(effect, context, param, vals); } void ALeaxreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_DENSITY: *val = props->Reverb.Density; break; case AL_EAXREVERB_DIFFUSION: *val = props->Reverb.Diffusion; break; case AL_EAXREVERB_GAIN: *val = props->Reverb.Gain; break; case AL_EAXREVERB_GAINHF: *val = props->Reverb.GainHF; break; case AL_EAXREVERB_GAINLF: *val = props->Reverb.GainLF; break; case AL_EAXREVERB_DECAY_TIME: *val = props->Reverb.DecayTime; break; case AL_EAXREVERB_DECAY_HFRATIO: *val = props->Reverb.DecayHFRatio; break; case AL_EAXREVERB_DECAY_LFRATIO: *val = props->Reverb.DecayLFRatio; break; case AL_EAXREVERB_REFLECTIONS_GAIN: *val = props->Reverb.ReflectionsGain; break; case AL_EAXREVERB_REFLECTIONS_DELAY: *val = props->Reverb.ReflectionsDelay; break; case AL_EAXREVERB_LATE_REVERB_GAIN: *val = props->Reverb.LateReverbGain; break; case AL_EAXREVERB_LATE_REVERB_DELAY: *val = props->Reverb.LateReverbDelay; break; case AL_EAXREVERB_AIR_ABSORPTION_GAINHF: *val = props->Reverb.AirAbsorptionGainHF; break; case AL_EAXREVERB_ECHO_TIME: *val = props->Reverb.EchoTime; break; case AL_EAXREVERB_ECHO_DEPTH: *val = props->Reverb.EchoDepth; break; case AL_EAXREVERB_MODULATION_TIME: *val = props->Reverb.ModulationTime; break; case AL_EAXREVERB_MODULATION_DEPTH: *val = props->Reverb.ModulationDepth; break; case AL_EAXREVERB_HFREFERENCE: *val = props->Reverb.HFReference; break; case AL_EAXREVERB_LFREFERENCE: *val = props->Reverb.LFReference; break; case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR: *val = props->Reverb.RoomRolloffFactor; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid EAX reverb float property 0x%04x", param); } } void ALeaxreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EAXREVERB_REFLECTIONS_PAN: vals[0] = props->Reverb.ReflectionsPan[0]; vals[1] = props->Reverb.ReflectionsPan[1]; vals[2] = props->Reverb.ReflectionsPan[2]; break; case AL_EAXREVERB_LATE_REVERB_PAN: vals[0] = props->Reverb.LateReverbPan[0]; vals[1] = props->Reverb.LateReverbPan[1]; vals[2] = props->Reverb.LateReverbPan[2]; break; default: ALeaxreverb_getParamf(effect, context, param, vals); break; } } DEFINE_ALEFFECT_VTABLE(ALeaxreverb); void ALreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DECAY_HFLIMIT: if(!(val >= AL_REVERB_MIN_DECAY_HFLIMIT && val <= AL_REVERB_MAX_DECAY_HFLIMIT)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay hflimit out of range"); props->Reverb.DecayHFLimit = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb integer property 0x%04x", param); } } void ALreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALreverb_setParami(effect, context, param, vals[0]); } void ALreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DENSITY: if(!(val >= AL_REVERB_MIN_DENSITY && val <= AL_REVERB_MAX_DENSITY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb density out of range"); props->Reverb.Density = val; break; case AL_REVERB_DIFFUSION: if(!(val >= AL_REVERB_MIN_DIFFUSION && val <= AL_REVERB_MAX_DIFFUSION)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb diffusion out of range"); props->Reverb.Diffusion = val; break; case AL_REVERB_GAIN: if(!(val >= AL_REVERB_MIN_GAIN && val <= AL_REVERB_MAX_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb gain out of range"); props->Reverb.Gain = val; break; case AL_REVERB_GAINHF: if(!(val >= AL_REVERB_MIN_GAINHF && val <= AL_REVERB_MAX_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb gainhf out of range"); props->Reverb.GainHF = val; break; case AL_REVERB_DECAY_TIME: if(!(val >= AL_REVERB_MIN_DECAY_TIME && val <= AL_REVERB_MAX_DECAY_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay time out of range"); props->Reverb.DecayTime = val; break; case AL_REVERB_DECAY_HFRATIO: if(!(val >= AL_REVERB_MIN_DECAY_HFRATIO && val <= AL_REVERB_MAX_DECAY_HFRATIO)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb decay hfratio out of range"); props->Reverb.DecayHFRatio = val; break; case AL_REVERB_REFLECTIONS_GAIN: if(!(val >= AL_REVERB_MIN_REFLECTIONS_GAIN && val <= AL_REVERB_MAX_REFLECTIONS_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb reflections gain out of range"); props->Reverb.ReflectionsGain = val; break; case AL_REVERB_REFLECTIONS_DELAY: if(!(val >= AL_REVERB_MIN_REFLECTIONS_DELAY && val <= AL_REVERB_MAX_REFLECTIONS_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb reflections delay out of range"); props->Reverb.ReflectionsDelay = val; break; case AL_REVERB_LATE_REVERB_GAIN: if(!(val >= AL_REVERB_MIN_LATE_REVERB_GAIN && val <= AL_REVERB_MAX_LATE_REVERB_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb late reverb gain out of range"); props->Reverb.LateReverbGain = val; break; case AL_REVERB_LATE_REVERB_DELAY: if(!(val >= AL_REVERB_MIN_LATE_REVERB_DELAY && val <= AL_REVERB_MAX_LATE_REVERB_DELAY)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb late reverb delay out of range"); props->Reverb.LateReverbDelay = val; break; case AL_REVERB_AIR_ABSORPTION_GAINHF: if(!(val >= AL_REVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_REVERB_MAX_AIR_ABSORPTION_GAINHF)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb air absorption gainhf out of range"); props->Reverb.AirAbsorptionGainHF = val; break; case AL_REVERB_ROOM_ROLLOFF_FACTOR: if(!(val >= AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR)) SETERR_RETURN(context, AL_INVALID_VALUE,, "Reverb room rolloff factor out of range"); props->Reverb.RoomRolloffFactor = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb float property 0x%04x", param); } } void ALreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALreverb_setParamf(effect, context, param, vals[0]); } void ALreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DECAY_HFLIMIT: *val = props->Reverb.DecayHFLimit; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb integer property 0x%04x", param); } } void ALreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALreverb_getParami(effect, context, param, vals); } void ALreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_REVERB_DENSITY: *val = props->Reverb.Density; break; case AL_REVERB_DIFFUSION: *val = props->Reverb.Diffusion; break; case AL_REVERB_GAIN: *val = props->Reverb.Gain; break; case AL_REVERB_GAINHF: *val = props->Reverb.GainHF; break; case AL_REVERB_DECAY_TIME: *val = props->Reverb.DecayTime; break; case AL_REVERB_DECAY_HFRATIO: *val = props->Reverb.DecayHFRatio; break; case AL_REVERB_REFLECTIONS_GAIN: *val = props->Reverb.ReflectionsGain; break; case AL_REVERB_REFLECTIONS_DELAY: *val = props->Reverb.ReflectionsDelay; break; case AL_REVERB_LATE_REVERB_GAIN: *val = props->Reverb.LateReverbGain; break; case AL_REVERB_LATE_REVERB_DELAY: *val = props->Reverb.LateReverbDelay; break; case AL_REVERB_AIR_ABSORPTION_GAINHF: *val = props->Reverb.AirAbsorptionGainHF; break; case AL_REVERB_ROOM_ROLLOFF_FACTOR: *val = props->Reverb.RoomRolloffFactor; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid reverb float property 0x%04x", param); } } void ALreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALreverb_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALreverb);