diff options
Diffstat (limited to 'audio/mods/paula.cpp')
| -rw-r--r-- | audio/mods/paula.cpp | 117 |
1 files changed, 112 insertions, 5 deletions
diff --git a/audio/mods/paula.cpp b/audio/mods/paula.cpp index 0ebf3bc32a..283c0cbb7b 100644 --- a/audio/mods/paula.cpp +++ b/audio/mods/paula.cpp @@ -20,14 +20,37 @@ * */ +/* + * The low-pass filter code is based on UAE's audio filter code + * found in audio.c. UAE is licensed under the terms of the GPLv2. + * + * audio.c in UAE states the following: + * Copyright 1995, 1996, 1997 Bernd Schmidt + * Copyright 1996 Marcus Sundberg + * Copyright 1996 Manfred Thole + * Copyright 2006 Toni Wilen + */ + +#include <math.h> + +#include "common/scummsys.h" + #include "audio/mods/paula.h" #include "audio/null.h" namespace Audio { -Paula::Paula(bool stereo, int rate, uint interruptFreq) : +Paula::Paula(bool stereo, int rate, uint interruptFreq, FilterMode filterMode) : _stereo(stereo), _rate(rate), _periodScale((double)kPalPaulaClock / rate), _intFreq(interruptFreq) { + _filterState.mode = filterMode; + _filterState.ledFilter = false; + filterResetState(); + + _filterState.a0[0] = filterCalculateA0(rate, 6200); + _filterState.a0[1] = filterCalculateA0(rate, 20000); + _filterState.a0[2] = filterCalculateA0(rate, 7000); + clearVoices(); _voice[0].panning = 191; _voice[1].panning = 63; @@ -73,12 +96,70 @@ int Paula::readBuffer(int16 *buffer, const int numSamples) { return readBufferIntern<false>(buffer, numSamples); } +/* Denormals are very small floating point numbers that force FPUs into slow + * mode. All lowpass filters using floats are suspectible to denormals unless + * a small offset is added to avoid very small floating point numbers. + */ +#define DENORMAL_OFFSET (1E-10) + +/* Based on UAE. + * Original comment in UAE: + * + * Amiga has two separate filtering circuits per channel, a static RC filter + * on A500 and the LED filter. This code emulates both. + * + * The Amiga filtering circuitry depends on Amiga model. Older Amigas seem + * to have a 6 dB/oct RC filter with cutoff frequency such that the -6 dB + * point for filter is reached at 6 kHz, while newer Amigas have no filtering. + * + * The LED filter is complicated, and we are modelling it with a pair of + * RC filters, the other providing a highboost. The LED starts to cut + * into signal somewhere around 5-6 kHz, and there's some kind of highboost + * in effect above 12 kHz. Better measurements are required. + * + * The current filtering should be accurate to 2 dB with the filter on, + * and to 1 dB with the filter off. + */ +inline int32 filter(int32 input, Paula::FilterState &state, int voice) { + float normalOutput, ledOutput; + + switch (state.mode) { + case Paula::kFilterModeA500: + state.rc[voice][0] = state.a0[0] * input + (1 - state.a0[0]) * state.rc[voice][0] + DENORMAL_OFFSET; + state.rc[voice][1] = state.a0[1] * state.rc[voice][0] + (1-state.a0[1]) * state.rc[voice][1]; + normalOutput = state.rc[voice][1]; + + state.rc[voice][2] = state.a0[2] * normalOutput + (1 - state.a0[2]) * state.rc[voice][2]; + state.rc[voice][3] = state.a0[2] * state.rc[voice][2] + (1 - state.a0[2]) * state.rc[voice][3]; + state.rc[voice][4] = state.a0[2] * state.rc[voice][3] + (1 - state.a0[2]) * state.rc[voice][4]; + + ledOutput = state.rc[voice][4]; + break; + + case Paula::kFilterModeA1200: + normalOutput = input; + + state.rc[voice][1] = state.a0[2] * normalOutput + (1 - state.a0[2]) * state.rc[voice][1] + DENORMAL_OFFSET; + state.rc[voice][2] = state.a0[2] * state.rc[voice][1] + (1 - state.a0[2]) * state.rc[voice][2]; + state.rc[voice][3] = state.a0[2] * state.rc[voice][2] + (1 - state.a0[2]) * state.rc[voice][3]; + + ledOutput = state.rc[voice][3]; + break; + + case Paula::kFilterModeNone: + default: + return input; + + } + + return CLIP<int32>(state.ledFilter ? ledOutput : normalOutput, -32768, 32767); +} template<bool stereo> -inline int mixBuffer(int16 *&buf, const int8 *data, Paula::Offset &offset, frac_t rate, int neededSamples, uint bufSize, byte volume, byte panning) { +inline int mixBuffer(int16 *&buf, const int8 *data, Paula::Offset &offset, frac_t rate, int neededSamples, uint bufSize, byte volume, byte panning, Paula::FilterState &filterState, int voice) { int samples; for (samples = 0; samples < neededSamples && offset.int_off < bufSize; ++samples) { - const int32 tmp = ((int32) data[offset.int_off]) * volume; + const int32 tmp = filter(((int32) data[offset.int_off]) * volume, filterState, voice); if (stereo) { *buf++ += (tmp * (255 - panning)) >> 7; *buf++ += (tmp * (panning)) >> 7; @@ -142,7 +223,7 @@ int Paula::readBufferIntern(int16 *buffer, const int numSamples) { // by the OS/2 version of Hopkins FBI. // Mix the generated samples into the output buffer - neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning); + neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning, _filterState, voice); // Wrap around if necessary if (ch.offset.int_off >= ch.length) { @@ -164,7 +245,7 @@ int Paula::readBufferIntern(int16 *buffer, const int numSamples) { // Repeat as long as necessary. while (neededSamples > 0) { // Mix the generated samples into the output buffer - neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning); + neededSamples -= mixBuffer<stereo>(p, ch.data, ch.offset, rate, neededSamples, ch.length, ch.volume, ch.panning, _filterState, voice); if (ch.offset.int_off >= ch.length) { // Wrap around. See also the note above. @@ -182,6 +263,32 @@ int Paula::readBufferIntern(int16 *buffer, const int numSamples) { return numSamples; } +void Paula::filterResetState() { + for (int i = 0; i < NUM_VOICES; i++) + for (int j = 0; j < 5; j++) + _filterState.rc[i][j] = 0.0f; +} + +/* Based on UAE. + * Original comment in UAE: + * + * This computes the 1st order low-pass filter term b0. + * The a1 term is 1.0 - b0. The center frequency marks the -3 dB point. + */ +float Paula::filterCalculateA0(int rate, int cutoff) { + float omega; + /* The BLT correction formula below blows up if the cutoff is above nyquist. */ + if (cutoff >= rate / 2) + return 1.0; + + omega = 2 * M_PI * cutoff / rate; + /* Compensate for the bilinear transformation. This allows us to specify the + * stop frequency more exactly, but the filter becomes less steep further + * from stopband. */ + omega = tan(omega / 2) * 2; + return 1 / (1 + 1 / omega); +} + } // End of namespace Audio |