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bsnes
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Update to v098r07 release. 

byuu says:

Changelog:
- GB: support modeSelect and RAM for MBC1M (Momotarou Collection)
- audio: implemented native resampling support into Emulator::Stream
- audio: removed nall::DSP completely

Unfortunately, the new resampler didn't turn out quite as fast as I had
hoped. The final hermite resampling added some overhead; and I had to
bump up the kernel count to 500 from 400 to get the buzzing to go away
on my main PC. I think that's due to it running at 48000hz output
instead of 44100hz output, maybe?

Compared to Ryphecha's:
(NES) Mega Man 2: 167fps -> 166fps
(GB) Mega Man II: 224fps -> 200fps
(WSC) Riviera: 143fps -> 151fps

Odd that the WS/WSC ends up faster while the DMG/CGB ends up slower.

But this knocks 922 lines down to 146 lines. The only files left in all
of higan not written (or rewritten) by me are ruby/xaudio2.h and
libco/ppc.c
This commit is contained in:
Tim Allen 2016-04-23 17:55:59 +10:00
parent e2ee6689a0
commit 7cdae5195a
26 changed files with 248 additions and 1293 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

60
higan/audio/audio.cpp
View File

@ -2,26 +2,9 @@
namespace Emulator { namespace Emulator {
#include "stream.cpp"
Audio audio; Audio audio;
Stream::Stream(double inputFrequency, double outputFrequency, double volume, double balance) {
dsp.setChannels(2);
dsp.setPrecision(16);
dsp.setFrequency(inputFrequency);
dsp.setResampler(DSP::ResampleEngine::Sinc);
dsp.setResamplerFrequency(outputFrequency);
dsp.setVolume(volume);
dsp.setBalance(balance);
}
auto Stream::sample(int16 left, int16 right) -> void {
int samples[] = {left, right};
dsp.sample(samples);
audio.poll();
}
//
auto Audio::reset() -> void { auto Audio::reset() -> void {
streams.reset(); streams.reset();
setReverbDelay(reverbDelay); setReverbDelay(reverbDelay);
@ -33,17 +16,15 @@ auto Audio::setInterface(Interface* interface) -> void {
auto Audio::setFrequency(double frequency) -> void { auto Audio::setFrequency(double frequency) -> void {
this->frequency = frequency; this->frequency = frequency;
for(auto& stream : streams) stream->dsp.setResamplerFrequency(frequency); for(auto& stream : streams) stream->setFrequency(frequency);
} }
auto Audio::setVolume(double volume) -> void { auto Audio::setVolume(double volume) -> void {
this->volume = volume; this->volume = volume;
for(auto& stream : streams) stream->dsp.setVolume(volume);
} }
auto Audio::setBalance(double balance) -> void { auto Audio::setBalance(double balance) -> void {
this->balance = balance; this->balance = balance;
for(auto& stream : streams) stream->dsp.setBalance(balance);
} }
auto Audio::setReverbDelay(uint reverbDelay) -> void { auto Audio::setReverbDelay(uint reverbDelay) -> void {
@ -58,8 +39,9 @@ auto Audio::setReverbLevel(double reverbLevel) -> void {
this->reverbLevel = reverbLevel; this->reverbLevel = reverbLevel;
} }
auto Audio::createStream(double frequency) -> shared_pointer<Stream> { auto Audio::createStream(uint channels, double frequency) -> shared_pointer<Stream> {
shared_pointer<Stream> stream = new Stream{frequency, this->frequency, volume, balance}; shared_pointer<Stream> stream = new Stream{channels, frequency};
stream->setFrequency(this->frequency);
streams.append(stream); streams.append(stream);
return stream; return stream;
} }
@ -68,25 +50,37 @@ auto Audio::createStream(double frequency) -> shared_pointer<Stream> {
auto Audio::poll() -> void { auto Audio::poll() -> void {
while(true) { while(true) {
for(auto& stream : streams) { for(auto& stream : streams) {
if(!stream->dsp.pending()) return; if(!stream->pending()) return;
} }
int left = 0, right = 0; double left = 0.0, right = 0.0;
for(auto& stream : streams) { for(auto& stream : streams) {
int samples[2]; double samples[2];
stream->dsp.read(samples); stream->read(samples);
left += samples[0]; left += samples[0];
right += samples[1]; right += samples[1];
} }
left /= streams.size();
right /= streams.size();
if(balance < 0.0) right *= 1.0 + balance;
if(balance > 0.0) left *= 1.0 - balance;
//todo: apply volume, reverb before denormalization?
int ileft = (left * 65535.0) - 32768.0;
int iright = (right * 65535.0) - 32768.0;
ileft *= volume;
iright *= volume;
if(reverbDelay) { if(reverbDelay) {
reverbLeft.append(left); reverbLeft.append(ileft);
reverbRight.append(right); reverbRight.append(iright);
left += reverbLeft.takeFirst() * reverbLevel; ileft += reverbLeft.takeFirst() * reverbLevel;
right += reverbRight.takeFirst() * reverbLevel; iright += reverbRight.takeFirst() * reverbLevel;
} }
interface->audioSample(sclamp<16>(left), sclamp<16>(right)); interface->audioSample(sclamp<16>(ileft), sclamp<16>(iright));
} }
} }

60
higan/audio/audio.hpp
View File

@ -1,20 +1,10 @@
#pragma once #pragma once
#include "core.hpp"
namespace Emulator { namespace Emulator {
struct Interface; struct Interface;
struct Audio;
struct Stream { struct Stream;
Stream(double inputFrequency, double outputFrequency, double volume, double balance);
auto sample(int16 left, int16 right) -> void;
private:
nall::DSP dsp;
friend class Audio;
};
struct Audio { struct Audio {
auto reset() -> void; auto reset() -> void;
@ -26,7 +16,7 @@ struct Audio {
auto setReverbDelay(uint milliseconds) -> void; auto setReverbDelay(uint milliseconds) -> void;
auto setReverbLevel(double level) -> void; auto setReverbLevel(double level) -> void;
auto createStream(double frequency) -> shared_pointer<Stream>; auto createStream(uint channels, double frequency) -> shared_pointer<Stream>;
auto poll() -> void; auto poll() -> void;
@ -44,6 +34,50 @@ private:
friend class Stream; friend class Stream;
}; };
struct Stream {
Stream(uint channels, double inputFrequency);
~Stream();
auto reset() -> void;
auto setFrequency(double outputFrequency) -> void;
auto pending() const -> bool;
auto read(double* samples) -> void;
auto write(int16* samples) -> void;
template<typename... P> auto sample(P&&... p) -> void {
int16 samples[sizeof...(P)] = {forward<P>(p)...};
write(samples);
}
private:
const uint channels;
const double inputFrequency;
double outputFrequency = 0.0;
double cutoffFrequency = 0.0;
double* tap = nullptr;
uint taps = 0;
uint decimationRate = 0;
uint decimationOffset = 0;
double** input = nullptr;
uint inputOffset = 0;
double resamplerFrequency = 0.0;
double resamplerFraction = 0.0;
double resamplerStep = 0.0;
double** queue = nullptr;
double** output = nullptr;
uint outputs = 0;
uint outputReadOffset = 0;
uint outputWriteOffset = 0;
friend class Audio;
};
extern Audio audio; extern Audio audio;
} }

50
higan/audio/buffer.hpp
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@ -1,50 +0,0 @@
#pragma once
struct Buffer {
Buffer() {
}
~Buffer() {
setChannels(0);
}
auto setChannels(uint channels) -> void {
if(sample) {
for(auto c : range(this->channels)) {
if(sample[c]) delete[] sample[c];
}
delete[] sample;
}
this->channels = channels;
if(channels == 0) return;
sample = new double*[channels];
for(auto c : range(channels)) {
sample[c] = new double[65536]();
}
}
inline auto read(uint channel, int offset = 0) -> double& {
return sample[channel][(uint16_t)(rdoffset + offset)];
}
inline auto write(uint channel, int offset = 0) -> double& {
return sample[channel][(uint16_t)(wroffset + offset)];
}
inline auto clear() -> void {
for(auto c : range(channels)) {
for(auto n : range(65536)) {
sample[c][n] = 0;
}
}
rdoffset = 0;
wroffset = 0;
}
double** sample = nullptr;
uint16_t rdoffset = 0;
uint16_t wroffset = 0;
uint channels = 0;
};

164
higan/audio/core.hpp
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@ -1,164 +0,0 @@
#pragma once
#include <math.h>
#include <vector>
#include <nall/stdint.hpp>
namespace nall {
struct DSP;
struct Resampler {
Resampler(DSP& dsp) : dsp(dsp) {}
virtual ~Resampler() {}
virtual auto setFrequency() -> void = 0;
virtual auto clear() -> void = 0;
virtual auto sample() -> void = 0;
DSP& dsp;
double frequency = 44100.0;
};
struct DSP {
enum class ResampleEngine : uint {
Nearest,
Linear,
Cosine,
Cubic,
Hermite,
Average,
Sinc,
};
inline DSP();
inline ~DSP();
inline auto setChannels(uint channels) -> void;
inline auto setPrecision(uint precision) -> void;
inline auto setFrequency(double frequency) -> void; //inputFrequency
inline auto setVolume(double volume) -> void;
inline auto setBalance(double balance) -> void;
inline auto setResampler(ResampleEngine resamplingEngine) -> void;
inline auto setResamplerFrequency(double frequency) -> void; //outputFrequency
inline auto sample(int channel[]) -> void;
inline auto pending() const -> bool;
inline auto read(int channel[]) -> void;
inline auto clear() -> void;
protected:
inline auto write(double channel[]) -> void;
inline auto adjustVolume() -> void;
inline auto adjustBalance() -> void;
inline auto clamp(const uint bits, const int input) -> int;
struct Settings {
uint channels;
uint precision;
double frequency;
double volume;
double balance;
//internal
double intensity;
double intensityInverse;
} settings;
Resampler* resampler = nullptr;
#include "buffer.hpp"
Buffer buffer;
Buffer output;
friend class ResampleNearest;
friend class ResampleLinear;
friend class ResampleCosine;
friend class ResampleCubic;
friend class ResampleAverage;
friend class ResampleHermite;
friend class ResampleSinc;
};
#include "resample/nearest.hpp"
#include "resample/linear.hpp"
#include "resample/cosine.hpp"
#include "resample/cubic.hpp"
#include "resample/hermite.hpp"
#include "resample/average.hpp"
#include "resample/sinc.hpp"
#include "settings.hpp"
DSP::DSP() {
setResampler(ResampleEngine::Hermite);
setResamplerFrequency(44100.0);
setChannels(2);
setPrecision(16);
setFrequency(44100.0);
setVolume(1.0);
setBalance(0.0);
clear();
}
DSP::~DSP() {
if(resampler) delete resampler;
}
auto DSP::sample(int channel[]) -> void {
for(auto c : range(settings.channels)) {
buffer.write(c) = (double)channel[c] * settings.intensityInverse;
}
buffer.wroffset++;
resampler->sample();
}
auto DSP::pending() const -> bool {
return output.rdoffset != output.wroffset;
}
auto DSP::read(int channel[]) -> void {
adjustVolume();
adjustBalance();
for(auto c : range(settings.channels)) {
channel[c] = clamp(settings.precision, output.read(c) * settings.intensity);
}
output.rdoffset++;
}
auto DSP::write(double channel[]) -> void {
for(auto c : range(settings.channels)) {
output.write(c) = channel[c];
}
output.wroffset++;
}
auto DSP::adjustVolume() -> void {
for(auto c : range(settings.channels)) {
output.read(c) *= settings.volume;
}
}
auto DSP::adjustBalance() -> void {
if(settings.channels != 2) return; //TODO: support > 2 channels
if(settings.balance < 0.0) output.read(1) *= 1.0 + settings.balance;
if(settings.balance > 0.0) output.read(0) *= 1.0 - settings.balance;
}
auto DSP::clamp(const uint bits, const int x) -> int {
const int b = 1U << (bits - 1);
const int m = (1U << (bits - 1)) - 1;
return (x > m) ? m : (x < -b) ? -b : x;
}
auto DSP::clear() -> void {
buffer.clear();
output.clear();
resampler->clear();
}
}

72
higan/audio/resample/average.hpp
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@ -1,72 +0,0 @@
#pragma once
struct ResampleAverage : Resampler {
ResampleAverage(DSP& dsp) : Resampler(dsp) {}
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
inline auto sampleLinear() -> void;
private:
double fraction;
double step;
};
auto ResampleAverage::setFrequency() -> void {
fraction = 0.0;
step = dsp.settings.frequency / frequency;
}
auto ResampleAverage::clear() -> void {
fraction = 0.0;
}
auto ResampleAverage::sample() -> void {
//can only average if input frequency >= output frequency
if(step < 1.0) return sampleLinear();
fraction += 1.0;
double scalar = 1.0;
if(fraction > step) scalar = 1.0 - (fraction - step);
for(auto c : range(dsp.settings.channels)) {
dsp.output.write(c) += dsp.buffer.read(c) * scalar;
}
if(fraction >= step) {
for(auto c : range(dsp.settings.channels)) {
dsp.output.write(c) /= step;
}
dsp.output.wroffset++;
fraction -= step;
for(auto c : range(dsp.settings.channels)) {
dsp.output.write(c) = dsp.buffer.read(c) * fraction;
}
}
dsp.buffer.rdoffset++;
}
auto ResampleAverage::sampleLinear() -> void {
while(fraction <= 1.0) {
double channel[dsp.settings.channels];
for(auto n : range(dsp.settings.channels)) {
double a = dsp.buffer.read(n, -1);
double b = dsp.buffer.read(n, -0);
double mu = fraction;
channel[n] = a * (1.0 - mu) + b * mu;
}
dsp.write(channel);
fraction += step;
}
dsp.buffer.rdoffset++;
fraction -= 1.0;
}

44
higan/audio/resample/cosine.hpp
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@ -1,44 +0,0 @@
#pragma once
struct ResampleCosine : Resampler {
ResampleCosine(DSP& dsp) : Resampler(dsp) {}
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
private:
double fraction;
double step;
};
auto ResampleCosine::setFrequency() -> void {
fraction = 0.0;
step = dsp.settings.frequency / frequency;
}
auto ResampleCosine::clear() -> void {
fraction = 0.0;
}
auto ResampleCosine::sample() -> void {
while(fraction <= 1.0) {
double channel[dsp.settings.channels];
for(auto n : range(dsp.settings.channels)) {
double a = dsp.buffer.read(n, -1);
double b = dsp.buffer.read(n, -0);
double mu = fraction;
mu = (1.0 - cos(mu * 3.14159265)) / 2.0;
channel[n] = a * (1.0 - mu) + b * mu;
}
dsp.write(channel);
fraction += step;
}
dsp.buffer.rdoffset++;
fraction -= 1.0;
}

50
higan/audio/resample/cubic.hpp
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@ -1,50 +0,0 @@
#pragma once
struct ResampleCubic : Resampler {
ResampleCubic(DSP& dsp) : Resampler(dsp) {}
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
private:
double fraction;
double step;
};
auto ResampleCubic::setFrequency() -> void {
fraction = 0.0;
step = dsp.settings.frequency / frequency;
}
auto ResampleCubic::clear() -> void {
fraction = 0.0;
}
auto ResampleCubic::sample() -> void {
while(fraction <= 1.0) {
double channel[dsp.settings.channels];
for(auto n : range(dsp.settings.channels)) {
double a = dsp.buffer.read(n, -3);
double b = dsp.buffer.read(n, -2);
double c = dsp.buffer.read(n, -1);
double d = dsp.buffer.read(n, -0);
double mu = fraction;
double A = d - c - a + b;
double B = a - b - A;
double C = c - a;
double D = b;
channel[n] = A * (mu * 3) + B * (mu * 2) + C * mu + D;
}
dsp.write(channel);
fraction += step;
}
dsp.buffer.rdoffset++;
fraction -= 1.0;
}

62
higan/audio/resample/hermite.hpp
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@ -1,62 +0,0 @@
#pragma once
struct ResampleHermite : Resampler {
ResampleHermite(DSP& dsp) : Resampler(dsp) {}
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
private:
double fraction;
double step;
};
auto ResampleHermite::setFrequency() -> void {
fraction = 0.0;
step = dsp.settings.frequency / frequency;
}
auto ResampleHermite::clear() -> void {
fraction = 0.0;
}
auto ResampleHermite::sample() -> void {
while(fraction <= 1.0) {
double channel[dsp.settings.channels];
for(auto n : range(dsp.settings.channels)) {
double a = dsp.buffer.read(n, -3);
double b = dsp.buffer.read(n, -2);
double c = dsp.buffer.read(n, -1);
double d = dsp.buffer.read(n, -0);
const double tension = 0.0; //-1 = low, 0 = normal, +1 = high
const double bias = 0.0; //-1 = left, 0 = even, +1 = right
double mu1, mu2, mu3, m0, m1, a0, a1, a2, a3;
mu1 = fraction;
mu2 = mu1 * mu1;
mu3 = mu2 * mu1;
m0 = (b - a) * (1.0 + bias) * (1.0 - tension) / 2.0;
m0 += (c - b) * (1.0 - bias) * (1.0 - tension) / 2.0;
m1 = (c - b) * (1.0 + bias) * (1.0 - tension) / 2.0;
m1 += (d - c) * (1.0 - bias) * (1.0 - tension) / 2.0;
a0 = +2 * mu3 - 3 * mu2 + 1;
a1 = mu3 - 2 * mu2 + mu1;
a2 = mu3 - mu2;
a3 = -2 * mu3 + 3 * mu2;
channel[n] = (a0 * b) + (a1 * m0) + (a2 * m1) + (a3 * c);
}
dsp.write(channel);
fraction += step;
}
dsp.buffer.rdoffset++;
fraction -= 1.0;
}

600
higan/audio/resample/lib/sinc.hpp
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@ -1,600 +0,0 @@
// If these types are changed to anything other than "float", you should comment out the SSE detection directives below
// so that the SSE code is not used.
typedef float resample_coeff_t; // note: sizeof(resample_coeff_t) must be == to a power of 2, and not larger than 16
typedef float resample_samp_t;
// ...but don't comment this single RESAMPLE_SSEREGPARM define out when disabling SSE.
#define RESAMPLE_SSEREGPARM
#if defined(__SSE__)
#define SINCRESAMPLE_USE_SSE 1
#ifndef __x86_64__
#undef RESAMPLE_SSEREGPARM
#define RESAMPLE_SSEREGPARM __attribute__((sseregparm))
#endif
#else
// TODO: altivec here
#endif
namespace ResampleUtility
{
inline void kaiser_window(double* io, int count, double beta);
inline void gen_sinc(double* out, int size, double cutoff, double kaiser);
inline void gen_sinc_os(double* out, int size, double cutoff, double kaiser);
inline void normalize(double* io, int size, double gain = 1.0);
inline void* make_aligned(void* ptr, unsigned boundary); // boundary must be a power of 2
}
class SincResampleHR
{
private:
inline void Init(unsigned ratio_arg, double desired_bandwidth, double beta, double d);
inline void write(resample_samp_t sample) RESAMPLE_SSEREGPARM;
inline resample_samp_t read(void) RESAMPLE_SSEREGPARM;
inline bool output_avail(void);
private:
inline resample_samp_t mac(const resample_samp_t *wave, const resample_coeff_t *coeff, unsigned count);
unsigned ratio;
unsigned num_convolutions;
resample_coeff_t *coeffs;
std::vector<unsigned char> coeffs_mem;
// second half of ringbuffer should be copy of first half.
resample_samp_t *rb;
std::vector<unsigned char> rb_mem;
signed rb_readpos;
signed rb_writepos;
signed rb_in;
signed rb_eff_size;
friend class SincResample;
};
class SincResample
{
public:
enum
{
QUALITY_LOW = 0,
QUALITY_MEDIUM = 2,
QUALITY_HIGH = 4
};
inline SincResample(double input_rate, double output_rate, double desired_bandwidth, unsigned quality = QUALITY_HIGH);
inline void write(resample_samp_t sample) RESAMPLE_SSEREGPARM;
inline resample_samp_t read(void) RESAMPLE_SSEREGPARM;
inline bool output_avail(void);
private:
inline void Init(double input_rate, double output_rate, double desired_bandwidth, double beta, double d, unsigned pn_nume, unsigned phases_min);
inline resample_samp_t mac(const resample_samp_t *wave, const resample_coeff_t *coeffs_a, const resample_coeff_t *coeffs_b, const double ffract, unsigned count) RESAMPLE_SSEREGPARM;
unsigned num_convolutions;
unsigned num_phases;
unsigned step_int;
double step_fract;
double input_pos_fract;
std::vector<resample_coeff_t *> coeffs; // Pointers into coeff_mem.
std::vector<unsigned char> coeff_mem;
std::vector<resample_samp_t> rb; // second half should be copy of first half.
signed rb_readpos;
signed rb_writepos;
signed rb_in;
bool hr_used;
SincResampleHR hr;
};
//
// Code:
//
//#include "resample.hpp"
#if 0
namespace bit
{
inline unsigned round(unsigned x) {
if((x & (x - 1)) == 0) return x;
while(x & (x - 1)) x &= x - 1;
return x << 1;
}
}
#endif
void SincResampleHR::Init(unsigned ratio_arg, double desired_bandwidth, double beta, double d)
{
const unsigned align_boundary = 16;
std::vector<double> coeffs_tmp;
double cutoff; // 1.0 = f/2
ratio = ratio_arg;
//num_convolutions = ((unsigned)ceil(d / ((1.0 - desired_bandwidth) / ratio)) + 1) &~ 1; // round up to be even
num_convolutions = ((unsigned)ceil(d / ((1.0 - desired_bandwidth) / ratio)) | 1);
cutoff = (1.0 / ratio) - (d / num_convolutions);
//printf("%d %d %.20f\n", ratio, num_convolutions, cutoff);
assert(num_convolutions > ratio);
// Generate windowed sinc of POWER
coeffs_tmp.resize(num_convolutions);
//ResampleUtility::gen_sinc(&coeffs_tmp[0], num_convolutions, cutoff, beta);
ResampleUtility::gen_sinc_os(&coeffs_tmp[0], num_convolutions, cutoff, beta);
ResampleUtility::normalize(&coeffs_tmp[0], num_convolutions);
// Copy from coeffs_tmp to coeffs~
// We multiply many coefficients at a time in the mac loop, so make sure the last few that don't really
// exist are allocated, zero'd mem.
coeffs_mem.resize(((num_convolutions + 7) &~ 7) * sizeof(resample_coeff_t) + (align_boundary - 1));
coeffs = (resample_coeff_t *)ResampleUtility::make_aligned(&coeffs_mem[0], align_boundary);
for(unsigned i = 0; i < num_convolutions; i++)
coeffs[i] = coeffs_tmp[i];
rb_eff_size = nall::bit::round(num_convolutions * 2) >> 1;
rb_readpos = 0;
rb_writepos = 0;
rb_in = 0;
rb_mem.resize(rb_eff_size * 2 * sizeof(resample_samp_t) + (align_boundary - 1));
rb = (resample_samp_t *)ResampleUtility::make_aligned(&rb_mem[0], align_boundary);
}
inline bool SincResampleHR::output_avail(void)
{
return(rb_in >= (signed)num_convolutions);
}
inline void SincResampleHR::write(resample_samp_t sample)
{
assert(!output_avail());
rb[rb_writepos] = sample;
rb[rb_writepos + rb_eff_size] = sample;
rb_writepos = (rb_writepos + 1) & (rb_eff_size - 1);
rb_in++;
}
resample_samp_t SincResampleHR::mac(const resample_samp_t *wave, const resample_coeff_t *coeff, unsigned count)
{
#if SINCRESAMPLE_USE_SSE
__m128 accum_veca[2] = { _mm_set1_ps(0), _mm_set1_ps(0) };
resample_samp_t accum;
for(unsigned c = 0; c < count; c += 8)
{
for(unsigned i = 0; i < 2; i++)
{
__m128 co[2];
__m128 w[2];
co[i] = _mm_load_ps(&coeff[c + i * 4]);
w[i] = _mm_load_ps(&wave[c + i * 4]);
w[i] = _mm_mul_ps(w[i], co[i]);
accum_veca[i] = _mm_add_ps(w[i], accum_veca[i]);
}
}
__m128 accum_vec = _mm_add_ps(accum_veca[0], accum_veca[1]); //_mm_add_ps(_mm_add_ps(accum_veca[0], accum_veca[1]), _mm_add_ps(accum_veca[2], accum_veca[3]));
accum_vec = _mm_add_ps(accum_vec, _mm_shuffle_ps(accum_vec, accum_vec, (3 << 0) | (2 << 2) | (1 << 4) | (0 << 6)));
accum_vec = _mm_add_ps(accum_vec, _mm_shuffle_ps(accum_vec, accum_vec, (1 << 0) | (0 << 2) | (1 << 4) | (0 << 6)));
_mm_store_ss(&accum, accum_vec);
return accum;
#else
resample_samp_t accum[4] = { 0, 0, 0, 0 };
for(unsigned c = 0; c < count; c+= 4)
{
accum[0] += wave[c + 0] * coeff[c + 0];
accum[1] += wave[c + 1] * coeff[c + 1];
accum[2] += wave[c + 2] * coeff[c + 2];
accum[3] += wave[c + 3] * coeff[c + 3];
}
return (accum[0] + accum[1]) + (accum[2] + accum[3]); // don't mess with parentheses(assuming compiler doesn't already, which it may...
#endif
}
resample_samp_t SincResampleHR::read(void)
{
assert(output_avail());
resample_samp_t ret;
ret = mac(&rb[rb_readpos], &coeffs[0], num_convolutions);
rb_readpos = (rb_readpos + ratio) & (rb_eff_size - 1);
rb_in -= ratio;
return ret;
}
SincResample::SincResample(double input_rate, double output_rate, double desired_bandwidth, unsigned quality)
{
const struct
{
double beta;
double d;
unsigned pn_nume;
unsigned phases_min;
} qtab[5] =
{
{ 5.658, 3.62, 4096, 4 },
{ 6.764, 4.32, 8192, 4 },
{ 7.865, 5.0, 16384, 8 },
{ 8.960, 5.7, 32768, 16 },
{ 10.056, 6.4, 65536, 32 }
};
// Sanity checks
assert(ceil(input_rate) > 0);
assert(ceil(output_rate) > 0);
assert(ceil(input_rate / output_rate) <= 1024);
assert(ceil(output_rate / input_rate) <= 1024);
// The simplistic number-of-phases calculation code doesn't work well enough for when desired_bandwidth is close to 1.0 and when
// upsampling.
assert(desired_bandwidth >= 0.25 && desired_bandwidth < 0.96);
assert(quality >= 0 && quality <= 4);
hr_used = false;
#if 1
// Round down to the nearest multiple of 4(so wave buffer remains aligned)
// It also adjusts the effective intermediate sampling rate up slightly, so that the upper frequencies below f/2
// aren't overly attenuated so much. In the future, we might want to do an FFT or something to choose the intermediate rate more accurately
// to virtually eliminate over-attenuation.
unsigned ioratio_rd = (unsigned)floor(input_rate / (output_rate * (1.0 + (1.0 - desired_bandwidth) / 2) )) & ~3;
if(ioratio_rd >= 8)
{
hr.Init(ioratio_rd, desired_bandwidth, qtab[quality].beta, qtab[quality].d); //10.056, 6.4);
hr_used = true;
input_rate /= ioratio_rd;
}
#endif
Init(input_rate, output_rate, desired_bandwidth, qtab[quality].beta, qtab[quality].d, qtab[quality].pn_nume, qtab[quality].phases_min);
}
void SincResample::Init(double input_rate, double output_rate, double desired_bandwidth, double beta, double d, unsigned pn_nume, unsigned phases_min)
{
const unsigned max_mult_atatime = 8; // multiply "granularity". must be power of 2.
const unsigned max_mult_minus1 = (max_mult_atatime - 1);
const unsigned conv_alignment_bytes = 16; // must be power of 2
const double input_to_output_ratio = input_rate / output_rate;
const double output_to_input_ratio = output_rate / input_rate;
double cutoff; // 1.0 = input_rate / 2
std::vector<double> coeff_init_buffer;
// Round up num_convolutions to be even.
if(output_rate > input_rate)
num_convolutions = ((unsigned)ceil(d / (1.0 - desired_bandwidth)) + 1) & ~1;
else
num_convolutions = ((unsigned)ceil(d / (output_to_input_ratio * (1.0 - desired_bandwidth))) + 1) & ~1;
if(output_rate > input_rate) // Upsampling
cutoff = desired_bandwidth;
else // Downsampling
cutoff = output_to_input_ratio * desired_bandwidth;
// Round up to be even.
num_phases = (std::max<unsigned>(pn_nume / num_convolutions, phases_min) + 1) &~1;
// Adjust cutoff to account for the multiple phases.
cutoff = cutoff / num_phases;
assert((num_convolutions & 1) == 0);
assert((num_phases & 1) == 0);
// fprintf(stderr, "num_convolutions=%u, num_phases=%u, total expected coeff byte size=%lu\n", num_convolutions, num_phases,
// (long)((num_phases + 2) * ((num_convolutions + max_mult_minus1) & ~max_mult_minus1) * sizeof(float) + conv_alignment_bytes));
coeff_init_buffer.resize(num_phases * num_convolutions);
coeffs.resize(num_phases + 1 + 1);
coeff_mem.resize((num_phases + 1 + 1) * ((num_convolutions + max_mult_minus1) &~ max_mult_minus1) * sizeof(resample_coeff_t) + conv_alignment_bytes);
// Assign aligned pointers into coeff_mem
{
resample_coeff_t *base_ptr = (resample_coeff_t *)ResampleUtility::make_aligned(&coeff_mem[0], conv_alignment_bytes);
for(unsigned phase = 0; phase < (num_phases + 1 + 1); phase++)
{
coeffs[phase] = base_ptr + (((num_convolutions + max_mult_minus1) & ~max_mult_minus1) * phase);
}
}
ResampleUtility::gen_sinc(&coeff_init_buffer[0], num_phases * num_convolutions, cutoff, beta);
ResampleUtility::normalize(&coeff_init_buffer[0], num_phases * num_convolutions, num_phases);
// Reorder coefficients to allow for more efficient convolution.
for(int phase = -1; phase < ((int)num_phases + 1); phase++)
{
for(int conv = 0; conv < (int)num_convolutions; conv++)
{
double coeff;
if(phase == -1 && conv == 0)
coeff = 0;
else if(phase == (int)num_phases && conv == ((int)num_convolutions - 1))
coeff = 0;
else
coeff = coeff_init_buffer[conv * num_phases + phase];
coeffs[phase + 1][conv] = coeff;
}
}
// Free a bit of mem
coeff_init_buffer.resize(0);
step_int = floor(input_to_output_ratio);
step_fract = input_to_output_ratio - step_int;
input_pos_fract = 0;
// Do NOT use rb.size() later in the code, since it'll include the padding.
// We should only need one "max_mult_minus1" here, not two, since it won't matter if it over-reads(due to doing "max_mult_atatime" multiplications at a time
// rather than just 1, in which case this over-read wouldn't happen), from the first half into the duplicated half,
// since those corresponding coefficients will be zero anyway; this is just to handle the case of reading off the end of the duplicated half to
// prevent illegal memory accesses.
rb.resize(num_convolutions * 2 + max_mult_minus1);
rb_readpos = 0;
rb_writepos = 0;
rb_in = 0;
}
resample_samp_t SincResample::mac(const resample_samp_t *wave, const resample_coeff_t *coeffs_a, const resample_coeff_t *coeffs_b, const double ffract, unsigned count)
{
resample_samp_t accum = 0;
#if SINCRESAMPLE_USE_SSE
__m128 accum_vec_a[2] = { _mm_set1_ps(0), _mm_set1_ps(0) };
__m128 accum_vec_b[2] = { _mm_set1_ps(0), _mm_set1_ps(0) };
for(unsigned c = 0; c < count; c += 8) //8) //4)
{
__m128 coeff_a[2];
__m128 coeff_b[2];
__m128 w[2];
__m128 result_a[2], result_b[2];
for(unsigned i = 0; i < 2; i++)
{
coeff_a[i] = _mm_load_ps(&coeffs_a[c + (i * 4)]);
coeff_b[i] = _mm_load_ps(&coeffs_b[c + (i * 4)]);
w[i] = _mm_loadu_ps(&wave[c + (i * 4)]);
result_a[i] = _mm_mul_ps(coeff_a[i], w[i]);
result_b[i] = _mm_mul_ps(coeff_b[i], w[i]);
accum_vec_a[i] = _mm_add_ps(result_a[i], accum_vec_a[i]);
accum_vec_b[i] = _mm_add_ps(result_b[i], accum_vec_b[i]);
}
}
__m128 accum_vec, av_a, av_b;
__m128 mult_a_vec = _mm_set1_ps(1.0 - ffract);
__m128 mult_b_vec = _mm_set1_ps(ffract);
av_a = _mm_mul_ps(mult_a_vec, /*accum_vec_a[0]);*/ _mm_add_ps(accum_vec_a[0], accum_vec_a[1]));
av_b = _mm_mul_ps(mult_b_vec, /*accum_vec_b[0]);*/ _mm_add_ps(accum_vec_b[0], accum_vec_b[1]));
accum_vec = _mm_add_ps(av_a, av_b);
accum_vec = _mm_add_ps(accum_vec, _mm_shuffle_ps(accum_vec, accum_vec, (3 << 0) | (2 << 2) | (1 << 4) | (0 << 6)));
accum_vec = _mm_add_ps(accum_vec, _mm_shuffle_ps(accum_vec, accum_vec, (1 << 0) | (0 << 2) | (1 << 4) | (0 << 6)));
_mm_store_ss(&accum, accum_vec);
#else
resample_coeff_t mult_a = 1.0 - ffract;
resample_coeff_t mult_b = ffract;
for(unsigned c = 0; c < count; c += 4)
{
accum += wave[c + 0] * (coeffs_a[c + 0] * mult_a + coeffs_b[c + 0] * mult_b);
accum += wave[c + 1] * (coeffs_a[c + 1] * mult_a + coeffs_b[c + 1] * mult_b);
accum += wave[c + 2] * (coeffs_a[c + 2] * mult_a + coeffs_b[c + 2] * mult_b);
accum += wave[c + 3] * (coeffs_a[c + 3] * mult_a + coeffs_b[c + 3] * mult_b);
}
#endif
return accum;
}
inline bool SincResample::output_avail(void)
{
return(rb_in >= (int)num_convolutions);
}
resample_samp_t SincResample::read(void)
{
assert(output_avail());
double phase = input_pos_fract * num_phases - 0.5;
signed phase_int = (signed)floor(phase);
double phase_fract = phase - phase_int;
unsigned phase_a = num_phases - 1 - phase_int;
unsigned phase_b = phase_a - 1;
resample_samp_t ret;
ret = mac(&rb[rb_readpos], &coeffs[phase_a + 1][0], &coeffs[phase_b + 1][0], phase_fract, num_convolutions);
unsigned int_increment = step_int;
input_pos_fract += step_fract;
int_increment += floor(input_pos_fract);
input_pos_fract -= floor(input_pos_fract);
rb_readpos = (rb_readpos + int_increment) % num_convolutions;
rb_in -= int_increment;
return ret;
}
inline void SincResample::write(resample_samp_t sample)
{
assert(!output_avail());
if(hr_used)
{
hr.write(sample);
if(hr.output_avail())
{
sample = hr.read();
}
else
{
return;
}
}
rb[rb_writepos + 0 * num_convolutions] = sample;
rb[rb_writepos + 1 * num_convolutions] = sample;
rb_writepos = (rb_writepos + 1) % num_convolutions;
rb_in++;
}
void ResampleUtility::kaiser_window( double* io, int count, double beta)
{
int const accuracy = 24; //16; //12;
double* end = io + count;
double beta2 = beta * beta * (double) -0.25;
double to_fract = beta2 / ((double) count * count);
double i = 0;
double rescale = 0; // Doesn't need an initializer, to shut up gcc
for ( ; io < end; ++io, i += 1 )
{
double x = i * i * to_fract - beta2;
double u = x;
double k = x + 1;
double n = 2;
do
{
u *= x / (n * n);
n += 1;
k += u;
}
while ( k <= u * (1 << accuracy) );
if ( !i )
rescale = 1 / k; // otherwise values get large
*io *= k * rescale;
}
}
void ResampleUtility::gen_sinc(double* out, int size, double cutoff, double kaiser)
{
assert( size % 2 == 0 ); // size must be even
int const half_size = size / 2;
double* const mid = &out [half_size];
// Generate right half of sinc
for ( int i = 0; i < half_size; i++ )
{
double angle = (i * 2 + 1) * (Math::Pi / 2);
mid [i] = sin( angle * cutoff ) / angle;
}
kaiser_window( mid, half_size, kaiser );
// Mirror for left half
for ( int i = 0; i < half_size; i++ )
out [i] = mid [half_size - 1 - i];
}
void ResampleUtility::gen_sinc_os(double* out, int size, double cutoff, double kaiser)
{
assert( size % 2 == 1); // size must be odd
for(int i = 0; i < size; i++)
{
if(i == (size / 2))
out[i] = 2 * Math::Pi * (cutoff / 2); //0.078478; //1.0; //sin(2 * M_PI * (cutoff / 2) * (i - size / 2)) / (i - (size / 2));
else
out[i] = sin(2 * Math::Pi * (cutoff / 2) * (i - size / 2)) / (i - (size / 2));
// out[i] *= 0.3635819 - 0.4891775 * cos(2 * M_PI * i / (size - 1)) + 0.1365995 * cos(4 * M_PI * i / (size - 1)) - 0.0106411 * cos(6 * M_PI * i / (size - 1));
//0.42 - 0.5 * cos(2 * M_PI * i / (size - 1)) + 0.08 * cos(4 * M_PI * i / (size - 1));
// printf("%d %f\n", i, out[i]);
}
kaiser_window(&out[size / 2], size / 2 + 1, kaiser);
// Mirror for left half
for ( int i = 0; i < size / 2; i++ )
out [i] = out [size - 1 - i];
}
void ResampleUtility::normalize(double* io, int size, double gain)
{
double sum = 0;
for ( int i = 0; i < size; i++ )
sum += io [i];
double scale = gain / sum;
for ( int i = 0; i < size; i++ )
io [i] *= scale;
}
void* ResampleUtility::make_aligned(void* ptr, unsigned boundary)
{
unsigned char* null_ptr = (unsigned char *)nullptr;
unsigned char* uc_ptr = (unsigned char *)ptr;
uc_ptr += (boundary - ((uc_ptr - null_ptr) & (boundary - 1))) & (boundary - 1);
//while((uc_ptr - null_ptr) & (boundary - 1))
// uc_ptr++;
//printf("%16llx %16llx\n", (unsigned long long)ptr, (unsigned long long)uc_ptr);
assert((uc_ptr - (unsigned char *)ptr) < boundary && (uc_ptr >= (unsigned char *)ptr));
return uc_ptr;
}

43
higan/audio/resample/linear.hpp
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#pragma once
struct ResampleLinear : Resampler {
ResampleLinear(DSP& dsp) : Resampler(dsp) {}
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
private:
double fraction;
double step;
};
auto ResampleLinear::setFrequency() -> void {
fraction = 0.0;
step = dsp.settings.frequency / frequency;
}
auto ResampleLinear::clear() -> void {
fraction = 0.0;
}
auto ResampleLinear::sample() -> void {
while(fraction <= 1.0) {
double channel[dsp.settings.channels];
for(auto n : range(dsp.settings.channels)) {
double a = dsp.buffer.read(n, -1);
double b = dsp.buffer.read(n, -0);
double mu = fraction;
channel[n] = a * (1.0 - mu) + b * mu;
}
dsp.write(channel);
fraction += step;
}
dsp.buffer.rdoffset++;
fraction -= 1.0;
}

43
higan/audio/resample/nearest.hpp
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#pragma once
struct ResampleNearest : Resampler {
ResampleNearest(DSP& dsp) : Resampler(dsp) {}
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
private:
double fraction;
double step;
};
auto ResampleNearest::setFrequency() -> void {
fraction = 0.0;
step = dsp.settings.frequency / frequency;
}
auto ResampleNearest::clear() -> void {
fraction = 0.0;
}
auto ResampleNearest::sample() -> void {
while(fraction <= 1.0) {
double channel[dsp.settings.channels];
for(auto n : range(dsp.settings.channels)) {
double a = dsp.buffer.read(n, -1);
double b = dsp.buffer.read(n, -0);
double mu = fraction;
channel[n] = mu < 0.5 ? a : b;
}
dsp.write(channel);
fraction += step;
}
dsp.buffer.rdoffset++;
fraction -= 1.0;
}

62
higan/audio/resample/sinc.hpp
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#pragma once
#include "lib/sinc.hpp"
struct ResampleSinc : Resampler {
inline ResampleSinc(DSP& dsp);
inline ~ResampleSinc();
inline auto setFrequency() -> void;
inline auto clear() -> void;
inline auto sample() -> void;
private:
inline void remakeSinc();
SincResample* sincResampler[8] = {0};
};
ResampleSinc::ResampleSinc(DSP& dsp) : Resampler(dsp) {
for(auto n : range(8)) {
sincResampler[n] = nullptr;
}
}
ResampleSinc::~ResampleSinc() {
for(auto n : range(8)) {
if(sincResampler[n]) delete sincResampler[n];
}
}
auto ResampleSinc::setFrequency() -> void {
remakeSinc();
}
auto ResampleSinc::clear() -> void {
remakeSinc();
}
auto ResampleSinc::sample() -> void {
for(auto c : range(dsp.settings.channels)) {
sincResampler[c]->write(dsp.buffer.read(c));
}
if(sincResampler[0]->output_avail()) {
do {
for(auto c : range(dsp.settings.channels)) {
dsp.output.write(c) = sincResampler[c]->read();
}
dsp.output.wroffset++;
} while(sincResampler[0]->output_avail());
}
dsp.buffer.rdoffset++;
}
auto ResampleSinc::remakeSinc() -> void {
assert(dsp.settings.channels < 8);
for(auto c : range(dsp.settings.channels)) {
if(sincResampler[c]) delete sincResampler[c];
sincResampler[c] = new SincResample(dsp.settings.frequency, frequency, 0.85, SincResample::QUALITY_HIGH);
}
}

46
higan/audio/settings.hpp
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#pragma once
auto DSP::setChannels(uint channels) -> void {
channels = max(1u, channels);
buffer.setChannels(channels);
output.setChannels(channels);
settings.channels = channels;
}
auto DSP::setPrecision(uint precision) -> void {
settings.precision = precision;
settings.intensity = 1 << (settings.precision - 1);
settings.intensityInverse = 1.0 / settings.intensity;
}
auto DSP::setFrequency(double frequency) -> void {
settings.frequency = frequency;
resampler->setFrequency();
}
auto DSP::setVolume(double volume) -> void {
settings.volume = volume;
}
auto DSP::setBalance(double balance) -> void {
settings.balance = balance;
}
auto DSP::setResampler(ResampleEngine engine) -> void {
if(resampler) delete resampler;
switch(engine) { default:
case ResampleEngine::Nearest: resampler = new ResampleNearest(*this); return;
case ResampleEngine::Linear: resampler = new ResampleLinear (*this); return;
case ResampleEngine::Cosine: resampler = new ResampleCosine (*this); return;
case ResampleEngine::Cubic: resampler = new ResampleCubic (*this); return;
case ResampleEngine::Hermite: resampler = new ResampleHermite(*this); return;
case ResampleEngine::Average: resampler = new ResampleAverage(*this); return;
case ResampleEngine::Sinc: resampler = new ResampleSinc (*this); return;
}
}
auto DSP::setResamplerFrequency(double frequency) -> void {
resampler->frequency = frequency;
resampler->setFrequency();
}

146
higan/audio/stream.cpp Normal file
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//Emulator::Stream implements advanced audio resampling
//First, a lowpass sinc filter is used (with a Blackman window to reduce rippling) in order to remove aliasing
//Second, a decimator is used to reduce the CPU overhead of the sinc function
//Finally, a hermite resampler is used to resample to the exact requested output frequency
//Note: when the cutoff frequency is >= 0.5; only the hermite resampler is used
Stream::Stream(uint channels, double inputFrequency) : channels(channels), inputFrequency(inputFrequency) {
}
Stream::~Stream() {
reset();
}
auto Stream::reset() -> void {
if(tap) delete[] tap, tap = nullptr;
if(input) for(auto c : range(channels)) delete[] input[c];
delete[] input, input = nullptr;
if(queue) for(auto c : range(channels)) delete[] queue[c];
delete[] queue, queue = nullptr;
if(output) for(auto c : range(channels)) delete[] output[c];
delete[] output, output = nullptr;
}
auto Stream::setFrequency(double outputFrequency_) -> void {
reset();
const double pi = 3.141592;
auto sinc = [&](double x) -> double {
if(x == 0) return 1;
return sin(pi * x) / (pi * x);
};
outputFrequency = outputFrequency_;
cutoffFrequency = outputFrequency / inputFrequency;
if(cutoffFrequency < 0.5) {
double transitionBandwidth = 0.008; //lower = higher quality; more taps (slower)
taps = (uint)ceil(4.0 / transitionBandwidth) | 1;
tap = new double[taps];
double sum = 0.0;
for(uint t : range(taps)) {
//sinc filter
double s = sinc(2.0 * cutoffFrequency * (t - (taps - 1) / 2.0));
//blackman window
double b = 0.42 - 0.5 * cos(2.0 * pi * t / (taps - 1)) + 0.08 * cos(4.0 * pi * t / (taps - 1));
tap[t] = s * b;
sum += tap[t];
}
//normalize so that the sum of all coefficients is 1.0
for(auto t : range(taps)) tap[t] /= sum;
} else {
taps = 1;
tap = new double[taps];
tap[0] = 1.0;
}
decimationRate = max(1, (uint)floor(inputFrequency / outputFrequency));
decimationOffset = 0;
input = new double*[channels];
for(auto c : range(channels)) input[c] = new double[taps * 2]();
inputOffset = 0;
resamplerFrequency = inputFrequency / decimationRate;
resamplerFraction = 0.0;
resamplerStep = resamplerFrequency / outputFrequency;
queue = new double*[channels];
for(auto c : range(channels)) queue[c] = new double[4]();
output = new double*[channels];
outputs = inputFrequency * 0.02;
for(auto c : range(channels)) output[c] = new double[outputs]();
outputReadOffset = 0;
outputWriteOffset = 0;
}
auto Stream::pending() const -> bool {
return outputReadOffset != outputWriteOffset;
}
auto Stream::read(double* samples) -> void {
for(auto c : range(channels)) {
samples[c] = output[c][outputReadOffset];
}
if(channels == 1) samples[1] = samples[0]; //monaural->stereo hack
if(++outputReadOffset >= outputs) outputReadOffset = 0;
}
auto Stream::write(int16* samples) -> void {
inputOffset = !inputOffset ? taps - 1 : inputOffset - 1;
for(auto c : range(channels)) {
auto sample = (samples[c] + 32768.0) / 65535.0; //normalize
input[c][inputOffset] = input[c][inputOffset + taps] = sample;
}
if(++decimationOffset >= decimationRate) {
decimationOffset = 0;
for(auto c : range(channels)) {
double sample = 0.0;
for(auto t : range(taps)) sample += input[c][inputOffset + t] * tap[t];
auto& q = queue[c];
q[0] = q[1];
q[1] = q[2];
q[2] = q[3];
q[3] = sample;
}
//4-tap hermite
auto& mu = resamplerFraction;
while(mu <= 1.0) {
for(auto c : range(channels)) {
auto& q = queue[c];
const double tension = 0.0; //-1 = low, 0 = normal, +1 = high
const double bias = 0.0; //-1 = left, 0 = even, +1 = right
double mu1 = mu;
double mu2 = mu * mu;
double mu3 = mu * mu * mu;
double m0 = (q[1] - q[0]) * (1.0 + bias) * (1.0 - tension) / 2.0
+ (q[2] - q[1]) * (1.0 - bias) * (1.0 - tension) / 2.0;
double m1 = (q[2] - q[1]) * (1.0 + bias) * (1.0 - tension) / 2.0
+ (q[3] - q[2]) * (1.0 - bias) * (1.0 - tension) / 2.0;
double a0 = +2 * mu3 - 3 * mu2 + 1;
double a1 = mu3 - 2 * mu2 + mu1;
double a2 = mu3 - mu2;
double a3 = -2 * mu3 + 3 * mu2;
output[c][outputWriteOffset] = (a0 * q[1]) + (a1 * m0) + (a2 * m1) + (a3 * q[2]);
}
if(++outputWriteOffset >= outputs) outputWriteOffset = 0;
mu += resamplerStep;
audio.poll();
}
mu -= 1.0;
}
}

2
higan/emulator/emulator.hpp
View File

@ -8,7 +8,7 @@ using namespace nall;
namespace Emulator { namespace Emulator {
static const string Name = "higan"; static const string Name = "higan";
static const string Version = "098.06"; static const string Version = "098.07";
static const string Author = "byuu"; static const string Author = "byuu";
static const string License = "GPLv3"; static const string License = "GPLv3";
static const string Website = "http://byuu.org/"; static const string Website = "http://byuu.org/";

4
higan/fc/apu/apu.cpp
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@ -57,7 +57,7 @@ auto APU::main() -> void {
//output = filter.run_lopass(output); //output = filter.run_lopass(output);
output = sclamp<16>(output); output = sclamp<16>(output);
stream->sample(output, output); stream->sample(output);
tick(); tick();
} }
@ -89,7 +89,7 @@ auto APU::power() -> void {
auto APU::reset() -> void { auto APU::reset() -> void {
create(APU::Enter, 21'477'272); create(APU::Enter, 21'477'272);
stream = Emulator::audio.createStream(21'477'272.0 / 12.0); stream = Emulator::audio.createStream(1, 21'477'272.0 / 12.0);
pulse[0].reset(); pulse[0].reset();
pulse[1].reset(); pulse[1].reset();

2
higan/gb/apu/apu.cpp
View File

@ -62,7 +62,7 @@ auto APU::hipass(int16& sample, int64& bias) -> void {
auto APU::power() -> void { auto APU::power() -> void {
create(Enter, 2 * 1024 * 1024); create(Enter, 2 * 1024 * 1024);
if(!system.sgb()) stream = Emulator::audio.createStream(2 * 1024 * 1024); if(!system.sgb()) stream = Emulator::audio.createStream(2, 2 * 1024 * 1024);
for(uint n = 0xff10; n <= 0xff3f; n++) bus.mmio[n] = this; for(uint n = 0xff10; n <= 0xff3f; n++) bus.mmio[n] = this;
square1.power(); square1.power();

18
higan/gb/cartridge/mbc1m/mbc1m.cpp
View File

@ -1,10 +1,15 @@
auto Cartridge::MBC1M::mmio_read(uint16 addr) -> uint8 { auto Cartridge::MBC1M::mmio_read(uint16 addr) -> uint8 {
if((addr & 0xc000) == 0x0000) { //$0000-3fff if((addr & 0xc000) == 0x0000) { //$0000-3fff
return cartridge.rom_read((romHi << 4) * 0x4000 + addr.bits(0,13)); if(!modeSelect) return cartridge.rom_read(addr & 0x3fff);
return cartridge.rom_read((romHi << 4) * 0x4000 + (addr & 0x3fff));
} }
if((addr & 0xc000) == 0x4000) { //$4000-7fff if((addr & 0xc000) == 0x4000) { //$4000-7fff
return cartridge.rom_read((romHi << 4 | romLo) * 0x4000 + addr.bits(0,13)); return cartridge.rom_read((romHi << 4 | romLo) * 0x4000 + (addr & 0x3fff));
}
if((addr & 0xe000) == 0xa000) { //$a000-bfff
return cartridge.ram_read(addr & 0x1fff);
} }
return 0xff; return 0xff;
@ -18,9 +23,18 @@ auto Cartridge::MBC1M::mmio_write(uint16 addr, uint8 data) -> void {
if((addr & 0xe000) == 0x4000) { //$4000-5fff if((addr & 0xe000) == 0x4000) { //$4000-5fff
romHi = data.bits(0,1); romHi = data.bits(0,1);
} }
if((addr & 0xe000) == 0x6000) { //$6000-7fff
modeSelect = data.bit(0);
}
if((addr & 0xe000) == 0xa000) { //$a000-bfff
cartridge.ram_write(addr & 0x1fff, data);
}
} }
auto Cartridge::MBC1M::power() -> void { auto Cartridge::MBC1M::power() -> void {
romHi = 0; romHi = 0;
romLo = 1; romLo = 1;
modeSelect = 0;
} }

1
higan/gb/cartridge/mbc1m/mbc1m.hpp
View File

@ -5,4 +5,5 @@ struct MBC1M : MMIO {
uint4 romLo; uint4 romLo;
uint2 romHi; uint2 romHi;
uint1 modeSelect;
} mbc1m; } mbc1m;

1
higan/gb/cartridge/serialization.cpp
View File

@ -9,6 +9,7 @@ auto Cartridge::serialize(serializer& s) -> void {
s.integer(mbc1m.romLo); s.integer(mbc1m.romLo);
s.integer(mbc1m.romHi); s.integer(mbc1m.romHi);
s.integer(mbc1m.modeSelect);
s.integer(mbc2.ram_enable); s.integer(mbc2.ram_enable);
s.integer(mbc2.rom_select); s.integer(mbc2.rom_select);

2
higan/gba/apu/apu.cpp
View File

@ -74,7 +74,7 @@ auto APU::step(uint clocks) -> void {
auto APU::power() -> void { auto APU::power() -> void {
create(APU::Enter, 16'777'216); create(APU::Enter, 16'777'216);
stream = Emulator::audio.createStream(16'777'216.0 / 512.0); stream = Emulator::audio.createStream(2, 16'777'216.0 / 512.0);
square1.power(); square1.power();
square2.power(); square2.power();

2
higan/sfc/coprocessor/icd2/icd2.cpp
View File

@ -54,7 +54,7 @@ auto ICD2::power() -> void {
auto ICD2::reset(bool soft) -> void { auto ICD2::reset(bool soft) -> void {
create(ICD2::Enter, cpu.frequency / 5); create(ICD2::Enter, cpu.frequency / 5);
if(!soft) stream = Emulator::audio.createStream(4194304.0 / 2.0); if(!soft) stream = Emulator::audio.createStream(2, 4194304.0 / 2.0);
r6003 = 0x00; r6003 = 0x00;
r6004 = 0xff; r6004 = 0xff;

2
higan/sfc/coprocessor/msu1/msu1.cpp
View File

@ -59,7 +59,7 @@ auto MSU1::power() -> void {
auto MSU1::reset() -> void { auto MSU1::reset() -> void {
create(MSU1::Enter, 44100); create(MSU1::Enter, 44100);
stream = Emulator::audio.createStream(44100.0); stream = Emulator::audio.createStream(2, 44100.0);
mmio.dataSeekOffset = 0; mmio.dataSeekOffset = 0;
mmio.dataReadOffset = 0; mmio.dataReadOffset = 0;

2
higan/sfc/dsp/dsp.cpp
View File

@ -241,7 +241,7 @@ auto DSP::power() -> void {
auto DSP::reset() -> void { auto DSP::reset() -> void {
create(Enter, system.apuFrequency()); create(Enter, system.apuFrequency());
stream = Emulator::audio.createStream(system.apuFrequency() / 768.0); stream = Emulator::audio.createStream(2, system.apuFrequency() / 768.0);
REG(FLG) = 0xe0; REG(FLG) = 0xe0;
state.noise = 0x4000; state.noise = 0x4000;

2
higan/ws/apu/apu.cpp
View File

@ -66,7 +66,7 @@ auto APU::step(uint clocks) -> void {
auto APU::power() -> void { auto APU::power() -> void {
create(APU::Enter, 3'072'000); create(APU::Enter, 3'072'000);
stream = Emulator::audio.createStream(3'072'000.0); stream = Emulator::audio.createStream(2, 3'072'000.0);
bus.map(this, 0x004a, 0x004c); bus.map(this, 0x004a, 0x004c);
bus.map(this, 0x004e, 0x0050); bus.map(this, 0x004e, 0x0050);

1
ruby/audio/xaudio2.cpp
View File

@ -1,5 +1,6 @@
#include "xaudio2.hpp" #include "xaudio2.hpp"
#include <windows.h> #include <windows.h>
#include <audioclient.h>
struct AudioXAudio2 : Audio, public IXAudio2VoiceCallback { struct AudioXAudio2 : Audio, public IXAudio2VoiceCallback {
~AudioXAudio2() { term(); } ~AudioXAudio2() { term(); }