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Improve SNR measurement.

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This commit is contained in:
Themaister 2012-05-08 12:11:56 +02:00
parent 16e7355314
commit 3878e81ce4
4 changed files with 53 additions and 52 deletions
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12
audio/hermite.c
View File

@ -28,18 +28,6 @@ struct rarch_resampler
double r_frac;
};
void resampler_preinit(rarch_resampler_t *re, double omega, double *samples_offset)
{
*samples_offset = 2.0;
for (int i = 0; i < 4; i++)
{
re->chan_data[0][i] = (float)cos((i - 2) * omega);
re->chan_data[1][i] = re->chan_data[0][i];
}
re->r_frac = 0.0;
}
static inline float hermite_kernel(float mu1, float a, float b, float c, float d)
{
float mu2, mu3, m0, m1, a0, a1, a2, a3;

3
audio/resampler.h
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@ -42,8 +42,5 @@ rarch_resampler_t *resampler_new(void);
void resampler_process(rarch_resampler_t *re, struct resampler_data *data);
void resampler_free(rarch_resampler_t *re);
// Generate a starting cosine pulse with given frequency for testing (SNR, etc) purposes.
void resampler_preinit(rarch_resampler_t *re, double omega, double *samples_offset);
#endif

17
audio/sinc.c
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@ -43,9 +43,9 @@
#define PHASES_WRAP (1 << (PHASE_BITS + SUBPHASE_BITS))
#define FRAMES_SHIFT (PHASE_BITS + SUBPHASE_BITS)
#define SIDELOBES 16
#define SIDELOBES 8
#define TAPS (SIDELOBES * 2)
#define CUTOFF 0.9
#define CUTOFF 1.0
#define PHASE_INDEX 0
#define DELTA_INDEX 1
@ -61,19 +61,6 @@ struct rarch_resampler
uint32_t time;
};
void resampler_preinit(rarch_resampler_t *re, double omega, double *samples_offset)
{
*samples_offset = SIDELOBES + 1;
for (int i = 0; i < 2 * SIDELOBES; i++)
{
re->buffer_l[i] = re->buffer_l[i + TAPS] = cos((i - (SIDELOBES - 1)) * omega);
re->buffer_r[i] = re->buffer_r[i + TAPS] = re->buffer_l[i];
}
re->time = 0;
re->ptr = 0;
}
static inline double sinc(double val)
{
if (fabs(val) < 0.00001)

73
audio/test/snr.c
View File

@ -36,6 +36,9 @@ struct snr_result
{
double snr;
double gain;
unsigned alias_freq[3];
double alias_power[3];
};
static unsigned bitrange(unsigned len)
@ -115,7 +118,7 @@ static void calculate_fft_adjust(complex double *butterfly_buf, double gain, boo
if (merge_high)
{
for (unsigned i = 1; i < samples / 2; i++)
butterfly_buf[i] += conj(butterfly_buf[samples - i]);
butterfly_buf[i] *= 2.0;
}
// Normalize amplitudes.
@ -184,25 +187,51 @@ static void test_fft(void)
printf("%5.2lf }\n", creal(butterfly_buf[15]));
}
static void set_alias_power(struct snr_result *res, unsigned freq, double power)
{
for (unsigned 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;
}
}
}
// This doesn't yet take account for slight phase distortions,
// so reported SNR is lower than reality.
static void calculate_snr(struct snr_result *res,
unsigned in_rate,
unsigned in_rate, unsigned max_rate,
const float *resamp, complex double *butterfly_buf, size_t samples)
{
samples >>= 1;
calculate_fft(resamp, butterfly_buf, samples);
calculate_fft_adjust(butterfly_buf, 1.0 / samples, true, samples);
memset(res, 0, sizeof(*res));
double signal = cabs(butterfly_buf[in_rate] * butterfly_buf[in_rate]);
butterfly_buf[in_rate] = 0.0;
double noise = 0.0;
for (unsigned i = 0; i < samples / 2; i++)
noise += cabs(butterfly_buf[i] * butterfly_buf[i]);
// Aliased frequencies above half the original sampling rate are not considered.
for (unsigned 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 (unsigned i = 0; i < 3; i++)
res->alias_power[i] = 10.0 * log10(res->alias_power[i]);
}
int main(int argc, char *argv[])
@ -216,7 +245,7 @@ int main(int argc, char *argv[])
double ratio = strtod(argv[1], NULL);
const unsigned fft_samples = 1024 * 128;
unsigned out_rate = fft_samples;
unsigned out_rate = fft_samples / 2;
unsigned in_rate = out_rate / ratio;
ratio = (double)out_rate / in_rate;
@ -227,17 +256,19 @@ int main(int argc, char *argv[])
}
static const float freq_list[] = {
0.001, 0.002, 0.003, 0.004, 0.005, 0.008,
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,
0.10, 0.15, 0.20, 0.25, 0.30, 0.35, 0.40, 0.45,
0.46, 0.47, 0.48, 0.49, 0.495,
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 samples = in_rate * 2;
unsigned samples = in_rate * 4;
float *input = calloc(sizeof(float), samples);
float *output = calloc(sizeof(float), (fft_samples + 1) * 2);
complex double *butterfly_buf = calloc(sizeof(complex double), fft_samples);
bool warned = false;
complex double *butterfly_buf = calloc(sizeof(complex double), fft_samples / 2);
assert(input);
assert(output);
@ -250,32 +281,30 @@ int main(int argc, char *argv[])
{
unsigned freq = freq_list[i] * in_rate;
double omega = 2.0 * M_PI * freq / in_rate;
double sample_offset;
resampler_preinit(re, omega, &sample_offset);
gen_signal(input, omega, sample_offset, samples);
gen_signal(input, omega, 0, samples);
struct resampler_data data = {
.data_in = input,
.data_out = output,
.input_frames = in_rate,
.input_frames = in_rate * 2,
.ratio = ratio,
};
resampler_process(re, &data);
unsigned out_samples = data.output_frames * 2;
assert(out_samples >= fft_samples * 2);
if (out_samples != fft_samples * 2 && !warned)
{
fprintf(stderr, "Out samples != fft_samples ... %u / %u\n", out_samples, fft_samples * 2);
warned = true;
}
// We generate 2 seconds worth of audio, however, only the last second is considered so phase has stabilized.
struct snr_result res;
calculate_snr(&res, freq, output, butterfly_buf, fft_samples * 2);
calculate_snr(&res, freq, in_rate / 2, output + fft_samples, 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]);
}
resampler_free(re);