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(audio/test) Remove

This commit is contained in:
twinaphex 2016-11-03 22:54:11 +01:00
parent f4ab4c2975
commit a3aeb9c927
7 changed files with 0 additions and 652 deletions

@ -1,108 +0,0 @@
TESTS := test-sinc-lowest \
test-snr-sinc-lowest \
test-sinc-lower \
test-snr-sinc-lower \
test-sinc \
test-snr-sinc \
test-sinc-higher \
test-snr-sinc-higher \
test-sinc-highest \
test-snr-sinc-highest \
test-cc \
test-snr-cc
LIBRETRO_COMM_DIR = ../../libretro-common
CFLAGS += -O3 -ffast-math -g -Wall -pedantic -march=native -std=gnu99
CFLAGS += -DRESAMPLER_TEST -DRARCH_DUMMY_LOG -DRARCH_INTERNAL
CFLAGS += -I$(LIBRETRO_COMM_DIR)/include -I../../
LDFLAGS += -lm
SHAREDOBJ += $(LIBRETRO_COMM_DIR)/memmap/memalign.o \
$(LIBRETRO_COMM_DIR)/lists/string_list.o \
../..//config_file_userdata.o \
../audio_resampler_driver.o \
$(LIBRETRO_COMM_DIR)/file/config_file.o \
$(LIBRETRO_COMM_DIR)/streams/file_stream.o \
$(LIBRETRO_COMM_DIR)/string/stdstring.o \
$(LIBRETRO_COMM_DIR)/file/file_path.o \
$(LIBRETRO_COMM_DIR)/file/retro_stat.o \
$(LIBRETRO_COMM_DIR)/compat/compat_strl.o \
$(LIBRETRO_COMM_DIR)/hash/rhash.o \
all: $(TESTS)
main-cc.o: main.c
$(CC) -c -o $@ $< $(CFLAGS) -DRESAMPLER_IDENT='"CC"'
snr-cc.o: snr.c
$(CC) -c -o $@ $< $(CFLAGS) -DRESAMPLER_IDENT='"CC"'
cc-resampler.o: ../drivers_resampler/cc_resampler.c
$(CC) -c -o $@ $< $(CFLAGS)
sinc-lowest.o: ../drivers_resampler/sinc_resampler.c
$(CC) -c -o $@ $< $(CFLAGS) -DSINC_LOWEST_QUALITY
sinc-lower.o: ../drivers_resampler/sinc_resampler.c
$(CC) -c -o $@ $< $(CFLAGS) -DSINC_LOWER_QUALITY
sinc.o: ../drivers_resampler/sinc_resampler.c
$(CC) -c -o $@ $< $(CFLAGS)
nearest_resampler.o: ../drivers_resampler/nearest_resampler.c
$(CC) -c -o $@ $< $(CFLAGS)
sinc-higher.o: ../drivers_resampler/sinc.c
$(CC) -c -o $@ $< $(CFLAGS) -DSINC_HIGHER_QUALITY
sinc-highest.o: ../drivers_resampler/sinc.c
$(CC) -c -o $@ $< $(CFLAGS) -DSINC_HIGHEST_QUALITY
test-sinc-lowest: sinc-lowest.o ../audio_utils.o main.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-snr-sinc-lowest: sinc-lowest.o ../audio_utils.o snr.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-sinc-lower: sinc-lower.o ../audio_utils.o main.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-snr-sinc-lower: sinc-lower.o ../audio_utils.o snr.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-sinc: sinc.o ../audio_utils.o main.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-snr-sinc: sinc.o ../audio_utils.o snr.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-sinc-higher: sinc-higher.o ../audio_utils.o main.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-snr-sinc-higher: sinc-higher.o ../audio_utils.o snr.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-sinc-highest: sinc-highest.o ../audio_utils.o main.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-snr-sinc-highest: sinc-highest.o ../audio_utils.o snr.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-cc: cc-resampler.o ../audio_utils.o main-cc.o resampler-cc.o sinc.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
test-snr-cc: cc-resampler.o ../audio_utils.o snr-cc.o resampler-cc.o sinc.o nearest_resampler.o $(SHAREDOBJ)
$(CC) -o $@ $^ $(LDFLAGS)
%.o: %.c
$(CC) -c -o $@ $< $(CFLAGS)
clean:
rm -f $(TESTS)
rm -f *.o
rm -f ../*.o
.PHONY: clean

@ -1,6 +0,0 @@
function win = kaiser_window(N, beta)
% Create an N-point kaiser window with given beta.
indices = 2 * (0 : N - 1) / (N - 1) - 1;
mod = modified_bessel(beta);
win = modified_bessel(beta * sqrt(1 - indices.^2)) / mod;

@ -1,105 +0,0 @@
/* RetroArch - A frontend for libretro.
* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
*
* RetroArch is free software: you can redistribute it and/or modify it under the terms
* of the GNU General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* RetroArch 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along with RetroArch.
* If not, see <http://www.gnu.org/licenses/>.
*/
// Resampler that reads raw S16NE/stereo from stdin and outputs to stdout in S16NE/stereo.
// Used for testing and performance benchmarking.
#include "../audio_resampler_driver.h"
#include "../audio_utils.h"
#include <stdio.h>
#include <stdlib.h>
#include <time.h>
#ifndef RESAMPLER_IDENT
#define RESAMPLER_IDENT "sinc"
#endif
int main(int argc, char *argv[])
{
int16_t input_i[1024];
int16_t output_i[1024 * 8];
float input_f[1024];
float output_f[1024 * 8];
double in_rate, out_rate, ratio;
double ratio_max_deviation = 0.0;
const rarch_resampler_t *resampler = NULL;
void *re = NULL;
srand(time(NULL));
if (argc < 3 || argc > 4)
{
fprintf(stderr, "Usage: %s <in-rate> <out-rate> [ratio deviation] (max ratio: 8.0)\n", argv[0]);
return 1;
}
else if (argc == 4)
{
ratio_max_deviation = fabs(strtod(argv[3], NULL));
fprintf(stderr, "Ratio deviation: %.4f.\n", ratio_max_deviation);
}
in_rate = strtod(argv[1], NULL);
out_rate = strtod(argv[2], NULL);
ratio = out_rate / in_rate;
if (ratio >= 7.99)
{
fprintf(stderr, "Ratio is too high.\n");
return 1;
}
if (!rarch_resampler_realloc(&re, &resampler, RESAMPLER_IDENT, out_rate / in_rate))
{
fprintf(stderr, "Failed to allocate resampler ...\n");
return 1;
}
for (;;)
{
size_t output_samples;
struct resampler_data data;
double uniform, rate_mod;
if (fread(input_i, sizeof(int16_t), 1024, stdin) != 1024)
break;
uniform = (2.0 * rand()) / RAND_MAX - 1.0;
rate_mod = 1.0 + ratio_max_deviation * uniform;
convert_s16_to_float(input_f, input_i, 1024, 1.0f);
data.data_in = input_f;
data.data_out = output_f;
data.input_frames = sizeof(input_f) / (2 * sizeof(float));
data.ratio = ratio * rate_mod;
resampler->process(re, &data);
output_samples = data.output_frames * 2;
convert_float_to_s16(output_i, output_f, output_samples);
if (fwrite(output_i, sizeof(int16_t), output_samples, stdout) != output_samples)
break;
}
if (resampler && re)
resampler->free(re);
resampler = NULL;
re = NULL;
}

@ -1,21 +0,0 @@
function val = modified_bessel(x)
% Mirrors operation as done in RetroArch. Verify accuracy against Matlab's
% implementation.
sum = zeros(size(x));
factorial = ones(size(x));
factorial_mult = zeros(size(x));
x_pow = ones(size(x));
two_div_pow = ones(size(x));
x_sqr = x .* x;
for i = 0 : 17
sum = sum + x_pow .* two_div_pow ./ (factorial .* factorial);
factorial_mult = factorial_mult + 1.0;
x_pow = x_pow .* x_sqr;
two_div_pow = two_div_pow * 0.25;
factorial = factorial .* factorial_mult;
end
val = sum;

@ -1,51 +0,0 @@
% MATLAB test case for RetroArch SINC upsampler.
close all;
%%
% Test RetroArch's kaiser function.
real_kaiser = kaiser(1024, 10.0)';
rarch_kaiser = kaiser_window(1024, 10.0);
figure('name', 'Bessel function test');
subplot(2, 1, 1), plot(rarch_kaiser), title('RetroArch kaiser');
subplot(2, 1, 2), plot(rarch_kaiser - real_kaiser), title('Error');
%%
% 4-tap and 8-tap are Lanczos windowed, but include here for completeness.
phases = 256;
ratio = 2.0;
bw = min(1.0, ratio);
downsample = round(phases / ratio);
cutoffs = bw * [0.65 0.75 0.825 0.90 0.962];
betas = [2.0 3.0 5.5 10.5 14.5];
sidelobes = round([2 4 8 32 128] / bw);
taps = sidelobes * 2;
freqs = 0.05 : 0.02 : 0.99;
%filters = length(taps);
%for i = 1 : filters
for i = 5
filter_length = taps(i) * phases;
% Generate SINC.
sinc_indices = 2 * ((0 : (filter_length - 1)) / filter_length) - 1;
s = cutoffs(i) * sinc(cutoffs(i) * sinc_indices * sidelobes(i));
win = kaiser(filter_length, betas(i))';
filter = s .* win;
impulse_response_half = upfirdn(1, filter, phases, downsample) / bw;
figure('name', sprintf('Response SINC: %d taps', taps(i)));
freqz(impulse_response_half);
ylim([-200 0]);
signal = zeros(1, 80001);
for freq = freqs
signal = signal + sin(pi * freq * (0 : 80000));
end
resampled = upfirdn(signal, filter, phases, downsample);
figure('name', sprintf('Kaiser SINC: %d taps, w = %.f', taps(i), freq));
freqz(resampled .* kaiser(length(resampled), 40.0)', 1, 16 * 1024);
ylim([-180 100]);
end

@ -1,358 +0,0 @@
/* RetroArch - A frontend for libretro.
* Copyright (C) 2010-2014 - Hans-Kristian Arntzen
*
* RetroArch is free software: you can redistribute it and/or modify it under the terms
* of the GNU General Public License as published by the Free Software Found-
* ation, either version 3 of the License, or (at your option) any later version.
*
* RetroArch 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License along with RetroArch.
* If not, see <http://www.gnu.org/licenses/>.
*/
#include <stdio.h>
#include <stdlib.h>
#include <math.h>
#include <complex.h>
#include <string.h>
#include <assert.h>
#include <stdbool.h>
#include <retro_assert.h>
#include <retro_miscellaneous.h>
#include "../audio_resampler_driver.h"
#include "../audio_utils.h"
#ifndef RESAMPLER_IDENT
#define RESAMPLER_IDENT "sinc"
#endif
static void gen_signal(float *out, double omega, double bias_samples, size_t samples)
{
size_t i;
for (i = 0; i < samples; i += 2)
{
out[i + 0] = cos(((i >> 1) + bias_samples) * omega);
out[i + 1] = out[i + 0];
}
}
struct snr_result
{
double snr;
double gain;
unsigned alias_freq[3];
double alias_power[3];
};
static unsigned bitrange(unsigned len)
{
unsigned ret = 0;
while ((len >>= 1))
ret++;
return ret;
}
static unsigned bitswap(unsigned i, unsigned range)
{
unsigned shifts;
unsigned ret = 0;
for (shifts = 0; shifts < range; shifts++)
ret |= (i & (1 << (range - shifts - 1))) ? (1 << shifts) : 0;
return ret;
}
/* When interleaving the butterfly buffer, addressing puts bits in reverse.
* [0, 1, 2, 3, 4, 5, 6, 7] => [0, 4, 2, 6, 1, 5, 3, 7] */
static void interleave(complex double *butterfly_buf, size_t samples)
{
unsigned i;
unsigned range = bitrange(samples);
for (i = 0; i < samples; i++)
{
unsigned target = bitswap(i, range);
if (target > i)
{
complex double tmp = butterfly_buf[target];
butterfly_buf[target] = butterfly_buf[i];
butterfly_buf[i] = tmp;
}
}
}
static complex double gen_phase(double index)
{
return cexp(M_PI * I * index);
}
static void butterfly(complex double *a, complex double *b, complex double mod)
{
complex double a_;
complex double b_;
mod *= *b;
a_ = *a + mod;
b_ = *a - mod;
*a = a_;
*b = b_;
}
static void butterflies(complex double *butterfly_buf, double phase_dir, size_t step_size, size_t samples)
{
unsigned i, j;
for (i = 0; i < samples; i += 2 * step_size)
for (j = i; j < i + step_size; j++)
butterfly(&butterfly_buf[j], &butterfly_buf[j + step_size], gen_phase((phase_dir * (j - i)) / step_size));
}
static void calculate_fft(const float *data, complex double *butterfly_buf, size_t samples)
{
unsigned i;
unsigned step_size;
/* Enforce POT. */
retro_assert((samples & (samples - 1)) == 0);
for (i = 0; i < samples; i++)
butterfly_buf[i] = data[2 * i];
/* Interleave buffer to work with FFT. */
interleave(butterfly_buf, samples);
/* Fly, lovely butterflies! :D */
for (step_size = 1; step_size < samples; step_size *= 2)
butterflies(butterfly_buf, -1.0, step_size, samples);
}
static void calculate_fft_adjust(complex double *butterfly_buf, double gain, bool merge_high, size_t samples)
{
unsigned i;
if (merge_high)
{
for (i = 1; i < samples / 2; i++)
butterfly_buf[i] *= 2.0;
}
/* Normalize amplitudes. */
for (i = 0; i < samples; i++)
butterfly_buf[i] *= gain;
}
static void calculate_ifft(complex double *butterfly_buf, size_t samples, bool normalize)
{
unsigned step_size;
/* Enforce POT. */
retro_assert((samples & (samples - 1)) == 0);
interleave(butterfly_buf, samples);
/* Fly, lovely butterflies! In opposite direction! :D */
for (step_size = 1; step_size < samples; step_size *= 2)
butterflies(butterfly_buf, 1.0, step_size, samples);
if (normalize)
calculate_fft_adjust(butterfly_buf, 1.0 / samples, false, samples);
}
static void test_fft(void)
{
unsigned i, j;
float signal[32];
complex double butterfly_buf[16];
complex double buf_tmp[16];
const float cos_freqs[] = {
1.0, 4.0, 6.0,
};
const float sin_freqs[] = {
-2.0, 5.0, 7.0,
};
fprintf(stderr, "Sanity checking FFT ...\n");
for (i = 0; i < 16; i++)
{
signal[2 * i] = 0.0;
for (j = 0; j < sizeof(cos_freqs) / sizeof(cos_freqs[0]); j++)
signal[2 * i] += cos(2.0 * M_PI * i * cos_freqs[j] / 16.0);
for ( j = 0; j < sizeof(sin_freqs) / sizeof(sin_freqs[0]); j++)
signal[2 * i] += sin(2.0 * M_PI * i * sin_freqs[j] / 16.0);
}
calculate_fft(signal, butterfly_buf, 16);
memcpy(buf_tmp, butterfly_buf, sizeof(buf_tmp));
calculate_fft_adjust(buf_tmp, 1.0 / 16, true, 16);
fprintf(stderr, "FFT: { ");
for (i = 0; i < 7; i++)
fprintf(stderr, "(%4.2lf, %4.2lf), ", creal(buf_tmp[i]), cimag(buf_tmp[i]));
fprintf(stderr, "(%4.2lf, %4.2lf) }\n", creal(buf_tmp[7]), cimag(buf_tmp[7]));
calculate_ifft(butterfly_buf, 16, true);
fprintf(stderr, "Original: { ");
for (i = 0; i < 15; i++)
fprintf(stderr, "%5.2f, ", signal[2 * i]);
fprintf(stderr, "%5.2f }\n", signal[2 * 15]);
fprintf(stderr, "FFT => IFFT: { ");
for ( i = 0; i < 15; i++)
fprintf(stderr, "%5.2lf, ", creal(butterfly_buf[i]));
fprintf(stderr, "%5.2lf }\n", creal(butterfly_buf[15]));
}
static void set_alias_power(struct snr_result *res, unsigned freq, double power)
{
unsigned i;
for (i = 0; i < 3; i++)
{
if (power >= res->alias_power[i])
{
memmove(res->alias_freq + i + 1, res->alias_freq + i, (2 - i) * sizeof(res->alias_freq[0]));
memmove(res->alias_power + i + 1, res->alias_power + i, (2 - i) * sizeof(res->alias_power[0]));
res->alias_power[i] = power;
res->alias_freq[i] = freq;
break;
}
}
}
static void calculate_snr(struct snr_result *res,
unsigned in_rate, unsigned max_rate,
const float *resamp, complex double *butterfly_buf,
size_t samples)
{
unsigned i;
double signal;
double noise = 0.0;
samples >>= 1;
calculate_fft(resamp, butterfly_buf, samples);
calculate_fft_adjust(butterfly_buf, 1.0 / samples, true, samples);
memset(res, 0, sizeof(*res));
signal = cabs(butterfly_buf[in_rate] * butterfly_buf[in_rate]);
butterfly_buf[in_rate] = 0.0;
/* Aliased frequencies above half the original sampling rate are not considered. */
for (i = 0; i <= max_rate; i++)
{
double power = cabs(butterfly_buf[i] * butterfly_buf[i]);
set_alias_power(res, i, power);
noise += power;
}
res->snr = 10.0 * log10(signal / noise);
res->gain = 10.0 * log10(signal);
for (i = 0; i < 3; i++)
res->alias_power[i] = 10.0 * log10(res->alias_power[i]);
}
int main(int argc, char *argv[])
{
static const float freq_list[] = {
0.001, 0.002, 0.003, 0.004, 0.005, 0.006, 0.007, 0.008, 0.009,
0.010, 0.015, 0.020, 0.025, 0.030, 0.035, 0.040, 0.045, 0.050,
0.060, 0.070, 0.080, 0.090,
0.10, 0.15, 0.20, 0.25, 0.30, 0.35,
0.40, 0.41, 0.42, 0.43, 0.44, 0.45,
0.46, 0.47, 0.48, 0.49,
0.495, 0.496, 0.497, 0.498, 0.499,
};
unsigned out_rate, in_rate, samples, i;
double ratio;
float *input, *output;
complex double *butterfly_buf;
const rarch_resampler_t *resampler = NULL;
const unsigned fft_samples = 1024 * 128;
void *re = NULL;
if (argc != 2)
{
fprintf(stderr, "Usage: %s <ratio> (out-rate is fixed for FFT).\n", argv[0]);
return 1;
}
ratio = strtod(argv[1], NULL);
out_rate = fft_samples / 2;
in_rate = round(out_rate / ratio);
ratio = (double)out_rate / in_rate;
samples = in_rate * 4;
input = calloc(sizeof(float), samples);
output = calloc(sizeof(float), (fft_samples + 16) * 2);
butterfly_buf = calloc(sizeof(complex double), fft_samples / 2);
retro_assert(input);
retro_assert(output);
if (!rarch_resampler_realloc(&re, &resampler, RESAMPLER_IDENT, ratio))
{
free(input);
free(output);
free(butterfly_buf);
return 1;
}
test_fft();
for (i = 0; i < sizeof(freq_list) / sizeof(freq_list[0]); i++)
{
struct resampler_data data;
unsigned max_freq;
struct snr_result res = {0};
unsigned freq = freq_list[i] * in_rate;
double omega = 2.0 * M_PI * freq / in_rate;
gen_signal(input, omega, 0, samples);
data.data_in = input;
data.data_out = output;
data.input_frames = in_rate * 2;
data.ratio = ratio;
resampler->process(re, &data);
/* We generate 2 seconds worth of audio, however,
* only the last second is considered so phase has stabilized. */
max_freq = min(in_rate, out_rate) / 2;
if (freq > max_freq)
continue;
calculate_snr(&res, freq, max_freq, output + fft_samples - 2048, butterfly_buf, fft_samples);
printf("SNR @ w = %5.3f : %6.2lf dB, Gain: %6.1lf dB\n",
freq_list[i], res.snr, res.gain);
printf("\tAliases: #1 (w = %5.3f, %6.2lf dB), #2 (w = %5.3f, %6.2lf dB), #3 (w = %5.3f, %6.2lf dB)\n",
res.alias_freq[0] / (float)in_rate, res.alias_power[0],
res.alias_freq[1] / (float)in_rate, res.alias_power[1],
res.alias_freq[2] / (float)in_rate, res.alias_power[2]);
}
if (resampler && re)
resampler->free(re);
resampler = NULL;
re = NULL;
free(input);
free(output);
free(butterfly_buf);
}

@ -1,3 +0,0 @@
#!/bin/sh
ffmpeg -i "$1" -f s16le - | ./test-sinc-highest 44100 48000 $3 | ffmpeg -y -ar 48000 -f s16le -ac 2 -i - "$2"