diff --git a/audio/hermite.c b/audio/hermite.c index 6c96173e33..588c86d29e 100644 --- a/audio/hermite.c +++ b/audio/hermite.c @@ -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; diff --git a/audio/resampler.h b/audio/resampler.h index 2fccc3e0c0..704cdd31f9 100644 --- a/audio/resampler.h +++ b/audio/resampler.h @@ -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 diff --git a/audio/sinc.c b/audio/sinc.c index 029040eae8..040fcb16c5 100644 --- a/audio/sinc.c +++ b/audio/sinc.c @@ -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) diff --git a/audio/test/snr.c b/audio/test/snr.c index 2b7e34b956..c373f95df4 100644 --- a/audio/test/snr.c +++ b/audio/test/snr.c @@ -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);