/* Copyright (C) 2003, 2004, 2005, 2006, 2008, 2009 Dean Beeler, Jerome Fisher * Copyright (C) 2011-2016 Dean Beeler, Jerome Fisher, Sergey V. Mikayev * * This program is free software: you can redistribute it and/or modify * it under the terms of the GNU Lesser General Public License as published by * the Free Software Foundation, either version 2.1 of the License, or * (at your option) any later version. * * This program 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 Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public License * along with this program. If not, see . */ #include #include "internals.h" #include "Analog.h" #include "Synth.h" namespace MT32Emu { #if MT32EMU_USE_FLOAT_SAMPLES /* FIR approximation of the overall impulse response of the cascade composed of the sample & hold circuit and the low pass filter * of the MT-32 first generation. * The coefficients below are found by windowing the inverse DFT of the 1024 pin frequency response converted to the minimum phase. * The frequency response of the LPF is computed directly, the effect of the S&H is approximated by multiplying the LPF frequency * response by the corresponding sinc. Although, the LPF has DC gain of 3.2, we ignore this in the emulation and use normalised model. * The peak gain of the normalised cascade appears about 1.7 near 11.8 kHz. Relative error doesn't exceed 1% for the frequencies * below 12.5 kHz. In the higher frequency range, the relative error is below 8%. Peak error value is at 16 kHz. */ static const float COARSE_LPF_TAPS_MT32[] = { 1.272473681f, -0.220267785f, -0.158039905f, 0.179603785f, -0.111484097f, 0.054137498f, -0.023518029f, 0.010997169f, -0.006935698f }; // Similar approximation for new MT-32 and CM-32L/LAPC-I LPF. As the voltage controlled amplifier was introduced, LPF has unity DC gain. // The peak gain value shifted towards higher frequencies and a bit higher about 1.83 near 13 kHz. static const float COARSE_LPF_TAPS_CM32L[] = { 1.340615635f, -0.403331694f, 0.036005517f, 0.066156844f, -0.069672532f, 0.049563806f, -0.031113416f, 0.019169774f, -0.012421368f }; #else static const unsigned int COARSE_LPF_FRACTION_BITS = 14; // Integer versions of the FIRs above multiplied by (1 << 14) and rounded. static const SampleEx COARSE_LPF_TAPS_MT32[] = { 20848, -3609, -2589, 2943, -1827, 887, -385, 180, -114 }; static const SampleEx COARSE_LPF_TAPS_CM32L[] = { 21965, -6608, 590, 1084, -1142, 812, -510, 314, -204 }; #endif /* Combined FIR that both approximates the impulse response of the analogue circuits of sample & hold and the low pass filter * in the audible frequency range (below 20 kHz) and attenuates unwanted mirror spectra above 28 kHz as well. It is a polyphase * filter intended for resampling the signal to 48 kHz yet for applying high frequency boost. * As with the filter above, the analogue LPF frequency response is obtained for 1536 pin grid for range up to 96 kHz and multiplied * by the corresponding sinc. The result is further squared, windowed and passed to generalised Parks-McClellan routine as a desired response. * Finally, the minimum phase factor is found that's essentially the coefficients below. * Relative error in the audible frequency range doesn't exceed 0.0006%, attenuation in the stopband is better than 100 dB. * This level of performance makes it nearly bit-accurate for standard 16-bit sample resolution. */ // FIR version for MT-32 first generation. static const float ACCURATE_LPF_TAPS_MT32[] = { 0.003429281f, 0.025929869f, 0.096587777f, 0.228884848f, 0.372413431f, 0.412386503f, 0.263980018f, -0.014504962f, -0.237394528f, -0.257043496f, -0.103436603f, 0.063996095f, 0.124562333f, 0.083703206f, 0.013921662f, -0.033475018f, -0.046239712f, -0.029310921f, 0.00126585f, 0.021060961f, 0.017925605f, 0.003559874f, -0.005105248f, -0.005647917f, -0.004157918f, -0.002065664f, 0.00158747f, 0.003762585f, 0.001867137f, -0.001090028f, -0.001433979f, -0.00022367f, 4.34308E-05f, -0.000247827f, 0.000157087f, 0.000605823f, 0.000197317f, -0.000370511f, -0.000261202f, 9.96069E-05f, 9.85073E-05f, -5.28754E-05f, -1.00912E-05f, 7.69943E-05f, 2.03162E-05f, -5.67967E-05f, -3.30637E-05f, 1.61958E-05f, 1.73041E-05f }; // FIR version for new MT-32 and CM-32L/LAPC-I. static const float ACCURATE_LPF_TAPS_CM32L[] = { 0.003917452f, 0.030693861f, 0.116424199f, 0.275101674f, 0.43217361f, 0.431247894f, 0.183255659f, -0.174955671f, -0.354240244f, -0.212401714f, 0.072259178f, 0.204655344f, 0.108336211f, -0.039099027f, -0.075138174f, -0.026261906f, 0.00582663f, 0.003052193f, 0.00613657f, 0.017017951f, 0.008732535f, -0.011027427f, -0.012933664f, 0.001158097f, 0.006765958f, 0.00046778f, -0.002191106f, 0.001561017f, 0.001842871f, -0.001996876f, -0.002315836f, 0.000980965f, 0.001817454f, -0.000243272f, -0.000972848f, 0.000149941f, 0.000498886f, -0.000204436f, -0.000347415f, 0.000142386f, 0.000249137f, -4.32946E-05f, -0.000131231f, 3.88575E-07f, 4.48813E-05f, -1.31906E-06f, -1.03499E-05f, 7.71971E-06f, 2.86721E-06f }; // According to the CM-64 PCB schematic, there is a difference in the values of the LPF entrance resistors for the reverb and non-reverb channels. // This effectively results in non-unity LPF DC gain for the reverb channel of 0.68 while the LPF has unity DC gain for the LA32 output channels. // In emulation, the reverb output gain is multiplied by this factor to compensate for the LPF gain difference. static const float CM32L_REVERB_TO_LA32_ANALOG_OUTPUT_GAIN_FACTOR = 0.68f; static const unsigned int OUTPUT_GAIN_FRACTION_BITS = 8; static const float OUTPUT_GAIN_MULTIPLIER = float(1 << OUTPUT_GAIN_FRACTION_BITS); static const unsigned int COARSE_LPF_DELAY_LINE_LENGTH = 8; // Must be a power of 2 static const unsigned int ACCURATE_LPF_DELAY_LINE_LENGTH = 16; // Must be a power of 2 static const unsigned int ACCURATE_LPF_NUMBER_OF_PHASES = 3; // Upsampling factor static const unsigned int ACCURATE_LPF_PHASE_INCREMENT_REGULAR = 2; // Downsampling factor static const unsigned int ACCURATE_LPF_PHASE_INCREMENT_OVERSAMPLED = 1; // No downsampling static const Bit32u ACCURATE_LPF_DELTAS_REGULAR[][ACCURATE_LPF_NUMBER_OF_PHASES] = { { 0, 0, 0 }, { 1, 1, 0 }, { 1, 2, 1 } }; static const Bit32u ACCURATE_LPF_DELTAS_OVERSAMPLED[][ACCURATE_LPF_NUMBER_OF_PHASES] = { { 0, 0, 0 }, { 1, 0, 0 }, { 1, 0, 1 } }; class AbstractLowPassFilter { public: static AbstractLowPassFilter &createLowPassFilter(AnalogOutputMode mode, bool oldMT32AnalogLPF); virtual ~AbstractLowPassFilter() {} virtual SampleEx process(SampleEx sample) = 0; virtual bool hasNextSample() const; virtual unsigned int getOutputSampleRate() const; virtual unsigned int estimateInSampleCount(unsigned int outSamples) const; virtual void addPositionIncrement(unsigned int) {} }; class NullLowPassFilter : public AbstractLowPassFilter { public: SampleEx process(SampleEx sample); }; class CoarseLowPassFilter : public AbstractLowPassFilter { private: const SampleEx * const LPF_TAPS; SampleEx ringBuffer[COARSE_LPF_DELAY_LINE_LENGTH]; unsigned int ringBufferPosition; public: CoarseLowPassFilter(bool oldMT32AnalogLPF); SampleEx process(SampleEx sample); }; class AccurateLowPassFilter : public AbstractLowPassFilter { private: const float * const LPF_TAPS; const Bit32u (* const deltas)[ACCURATE_LPF_NUMBER_OF_PHASES]; const unsigned int phaseIncrement; const unsigned int outputSampleRate; SampleEx ringBuffer[ACCURATE_LPF_DELAY_LINE_LENGTH]; unsigned int ringBufferPosition; unsigned int phase; public: AccurateLowPassFilter(bool oldMT32AnalogLPF, bool oversample); SampleEx process(SampleEx sample); bool hasNextSample() const; unsigned int getOutputSampleRate() const; unsigned int estimateInSampleCount(unsigned int outSamples) const; void addPositionIncrement(unsigned int positionIncrement); }; Analog::Analog(const AnalogOutputMode mode, const bool oldMT32AnalogLPF) : leftChannelLPF(AbstractLowPassFilter::createLowPassFilter(mode, oldMT32AnalogLPF)), rightChannelLPF(AbstractLowPassFilter::createLowPassFilter(mode, oldMT32AnalogLPF)), synthGain(0), reverbGain(0) {} Analog::~Analog() { delete &leftChannelLPF; delete &rightChannelLPF; } void Analog::process(Sample *outStream, const Sample *nonReverbLeft, const Sample *nonReverbRight, const Sample *reverbDryLeft, const Sample *reverbDryRight, const Sample *reverbWetLeft, const Sample *reverbWetRight, Bit32u outLength) { if (outStream == NULL) { leftChannelLPF.addPositionIncrement(outLength); rightChannelLPF.addPositionIncrement(outLength); return; } while (0 < (outLength--)) { SampleEx outSampleL; SampleEx outSampleR; if (leftChannelLPF.hasNextSample()) { outSampleL = leftChannelLPF.process(0); outSampleR = rightChannelLPF.process(0); } else { SampleEx inSampleL = ((SampleEx)*(nonReverbLeft++) + (SampleEx)*(reverbDryLeft++)) * synthGain + (SampleEx)*(reverbWetLeft++) * reverbGain; SampleEx inSampleR = ((SampleEx)*(nonReverbRight++) + (SampleEx)*(reverbDryRight++)) * synthGain + (SampleEx)*(reverbWetRight++) * reverbGain; #if !MT32EMU_USE_FLOAT_SAMPLES inSampleL >>= OUTPUT_GAIN_FRACTION_BITS; inSampleR >>= OUTPUT_GAIN_FRACTION_BITS; #endif outSampleL = leftChannelLPF.process(inSampleL); outSampleR = rightChannelLPF.process(inSampleR); } *(outStream++) = Synth::clipSampleEx(outSampleL); *(outStream++) = Synth::clipSampleEx(outSampleR); } } unsigned int Analog::getOutputSampleRate() const { return leftChannelLPF.getOutputSampleRate(); } Bit32u Analog::getDACStreamsLength(Bit32u outputLength) const { return leftChannelLPF.estimateInSampleCount(outputLength); } void Analog::setSynthOutputGain(float useSynthGain) { #if MT32EMU_USE_FLOAT_SAMPLES synthGain = useSynthGain; #else if (OUTPUT_GAIN_MULTIPLIER < useSynthGain) useSynthGain = OUTPUT_GAIN_MULTIPLIER; synthGain = SampleEx(useSynthGain * OUTPUT_GAIN_MULTIPLIER); #endif } void Analog::setReverbOutputGain(float useReverbGain, bool mt32ReverbCompatibilityMode) { if (!mt32ReverbCompatibilityMode) useReverbGain *= CM32L_REVERB_TO_LA32_ANALOG_OUTPUT_GAIN_FACTOR; #if MT32EMU_USE_FLOAT_SAMPLES reverbGain = useReverbGain; #else if (OUTPUT_GAIN_MULTIPLIER < useReverbGain) useReverbGain = OUTPUT_GAIN_MULTIPLIER; reverbGain = SampleEx(useReverbGain * OUTPUT_GAIN_MULTIPLIER); #endif } AbstractLowPassFilter &AbstractLowPassFilter::createLowPassFilter(AnalogOutputMode mode, bool oldMT32AnalogLPF) { switch (mode) { case AnalogOutputMode_COARSE: return *new CoarseLowPassFilter(oldMT32AnalogLPF); case AnalogOutputMode_ACCURATE: return *new AccurateLowPassFilter(oldMT32AnalogLPF, false); case AnalogOutputMode_OVERSAMPLED: return *new AccurateLowPassFilter(oldMT32AnalogLPF, true); default: return *new NullLowPassFilter; } } bool AbstractLowPassFilter::hasNextSample() const { return false; } unsigned int AbstractLowPassFilter::getOutputSampleRate() const { return SAMPLE_RATE; } unsigned int AbstractLowPassFilter::estimateInSampleCount(unsigned int outSamples) const { return outSamples; } SampleEx NullLowPassFilter::process(const SampleEx inSample) { return inSample; } CoarseLowPassFilter::CoarseLowPassFilter(bool oldMT32AnalogLPF) : LPF_TAPS(oldMT32AnalogLPF ? COARSE_LPF_TAPS_MT32 : COARSE_LPF_TAPS_CM32L), ringBufferPosition(0) { Synth::muteSampleBuffer(ringBuffer, COARSE_LPF_DELAY_LINE_LENGTH); } SampleEx CoarseLowPassFilter::process(const SampleEx inSample) { static const unsigned int DELAY_LINE_MASK = COARSE_LPF_DELAY_LINE_LENGTH - 1; SampleEx sample = LPF_TAPS[COARSE_LPF_DELAY_LINE_LENGTH] * ringBuffer[ringBufferPosition]; ringBuffer[ringBufferPosition] = Synth::clipSampleEx(inSample); for (unsigned int i = 0; i < COARSE_LPF_DELAY_LINE_LENGTH; i++) { sample += LPF_TAPS[i] * ringBuffer[(i + ringBufferPosition) & DELAY_LINE_MASK]; } ringBufferPosition = (ringBufferPosition - 1) & DELAY_LINE_MASK; #if !MT32EMU_USE_FLOAT_SAMPLES sample >>= COARSE_LPF_FRACTION_BITS; #endif return sample; } AccurateLowPassFilter::AccurateLowPassFilter(const bool oldMT32AnalogLPF, const bool oversample) : LPF_TAPS(oldMT32AnalogLPF ? ACCURATE_LPF_TAPS_MT32 : ACCURATE_LPF_TAPS_CM32L), deltas(oversample ? ACCURATE_LPF_DELTAS_OVERSAMPLED : ACCURATE_LPF_DELTAS_REGULAR), phaseIncrement(oversample ? ACCURATE_LPF_PHASE_INCREMENT_OVERSAMPLED : ACCURATE_LPF_PHASE_INCREMENT_REGULAR), outputSampleRate(SAMPLE_RATE * ACCURATE_LPF_NUMBER_OF_PHASES / phaseIncrement), ringBufferPosition(0), phase(0) { Synth::muteSampleBuffer(ringBuffer, ACCURATE_LPF_DELAY_LINE_LENGTH); } SampleEx AccurateLowPassFilter::process(const SampleEx inSample) { static const unsigned int DELAY_LINE_MASK = ACCURATE_LPF_DELAY_LINE_LENGTH - 1; float sample = (phase == 0) ? LPF_TAPS[ACCURATE_LPF_DELAY_LINE_LENGTH * ACCURATE_LPF_NUMBER_OF_PHASES] * ringBuffer[ringBufferPosition] : 0.0f; if (!hasNextSample()) { ringBuffer[ringBufferPosition] = inSample; } for (unsigned int tapIx = phase, delaySampleIx = 0; delaySampleIx < ACCURATE_LPF_DELAY_LINE_LENGTH; delaySampleIx++, tapIx += ACCURATE_LPF_NUMBER_OF_PHASES) { sample += LPF_TAPS[tapIx] * ringBuffer[(delaySampleIx + ringBufferPosition) & DELAY_LINE_MASK]; } phase += phaseIncrement; if (ACCURATE_LPF_NUMBER_OF_PHASES <= phase) { phase -= ACCURATE_LPF_NUMBER_OF_PHASES; ringBufferPosition = (ringBufferPosition - 1) & DELAY_LINE_MASK; } return SampleEx(ACCURATE_LPF_NUMBER_OF_PHASES * sample); } bool AccurateLowPassFilter::hasNextSample() const { return phaseIncrement <= phase; } unsigned int AccurateLowPassFilter::getOutputSampleRate() const { return outputSampleRate; } unsigned int AccurateLowPassFilter::estimateInSampleCount(unsigned int outSamples) const { Bit32u cycleCount = outSamples / ACCURATE_LPF_NUMBER_OF_PHASES; Bit32u remainder = outSamples - cycleCount * ACCURATE_LPF_NUMBER_OF_PHASES; return cycleCount * phaseIncrement + deltas[remainder][phase]; } void AccurateLowPassFilter::addPositionIncrement(const unsigned int positionIncrement) { phase = (phase + positionIncrement * phaseIncrement) % ACCURATE_LPF_NUMBER_OF_PHASES; } } // namespace MT32Emu