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Diffstat (limited to 'audio/softsynth/mt32/Analog.cpp')
| -rw-r--r-- | audio/softsynth/mt32/Analog.cpp | 348 |
1 files changed, 348 insertions, 0 deletions
diff --git a/audio/softsynth/mt32/Analog.cpp b/audio/softsynth/mt32/Analog.cpp new file mode 100644 index 0000000000..8ac28e401a --- /dev/null +++ b/audio/softsynth/mt32/Analog.cpp @@ -0,0 +1,348 @@ +/* Copyright (C) 2003, 2004, 2005, 2006, 2008, 2009 Dean Beeler, Jerome Fisher + * Copyright (C) 2011, 2012, 2013, 2014 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 <http://www.gnu.org/licenses/>. + */ + +//#include <cstring> +#include "Analog.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); + static void muteRingBuffer(SampleEx *ringBuffer, unsigned int length); + + 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 ControlROMFeatureSet *controlROMFeatures) : + leftChannelLPF(AbstractLowPassFilter::createLowPassFilter(mode, controlROMFeatures->isOldMT32AnalogLPF())), + rightChannelLPF(AbstractLowPassFilter::createLowPassFilter(mode, controlROMFeatures->isOldMT32AnalogLPF())), + 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; + } +} + +void AbstractLowPassFilter::muteRingBuffer(SampleEx *ringBuffer, unsigned int length) { + +#if MT32EMU_USE_FLOAT_SAMPLES + + SampleEx *p = ringBuffer; + while (length--) { + *(p++) = 0.0f; + } + +#else + + memset(ringBuffer, 0, length * sizeof(SampleEx)); + +#endif + +} + +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) +{ + muteRingBuffer(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) +{ + muteRingBuffer(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; +} + +} |