better resampler

This commit is contained in:
Concedo
2026-03-12 16:49:53 +08:00
parent 796f7bdeff
commit 3092694d2e
3 changed files with 81 additions and 23 deletions
+79 -21
View File
@@ -383,33 +383,91 @@ std::vector<std::vector<int>> split_big_vector_in_two(const std::vector<int>& bi
return result;
}
std::vector<float> resample_wav(const std::vector<float> & input, uint32_t input_rate, uint32_t output_rate) {
if (input.empty() || input_rate == 0 || output_rate == 0)
static double audio_resample_bessel_i0(double x) {
double sum = 1.0;
double term = 1.0;
double y = x * x * 0.25;
for (int k = 1; k < 30; k++) {
term *= y / ((double) k * (double) k);
sum += term;
if (term < sum * 1e-15) {
break;
}
}
return sum;
}
std::vector<float> resample_wav(int num_channels,const std::vector<float>& input,uint32_t input_rate,uint32_t output_rate)
{
if (input.empty() || num_channels <= 0 || input_rate == 0 || output_rate == 0)
return {};
const size_t input_size = input.size();
const double ratio = static_cast<double>(output_rate) / input_rate; // Compute resampling ratio
// Use rounding to avoid systematic truncation error
const size_t output_size = static_cast<size_t>(std::llround(input_size * ratio));
std::vector<float> output(output_size);
const double step = static_cast<double>(input_rate) / output_rate; // Precompute step in source domain
double src_pos = 0.0;
for (size_t i = 0; i < output_size; ++i)
if (input.size() % num_channels != 0)
return {};
const int n_in = input.size() / num_channels;
if (input_rate == output_rate)
return input;
const double ratio = (double)output_rate / (double)input_rate;
const int n_out = (int)std::lround(n_in * ratio);
std::vector<float> output((size_t)n_out * num_channels);
const int half_len = 32;
const double beta = 9.0;
const double inv_i0b = 1.0 / audio_resample_bessel_i0(beta);
const double fc = 0.5 * ((ratio < 1.0) ? ratio : 1.0);
for (int ch = 0; ch < num_channels; ch++)
{
size_t idx = static_cast<size_t>(src_pos);
if (idx >= input_size - 1) // Clamp to valid range (prevents out-of-bounds)
const float* src = input.data() + ch * n_in;
float* dst = output.data() + ch * n_out;
for (int i = 0; i < n_out; i++)
{
output[i] = input[input_size - 1];
double center = (double)i / ratio;
int start = (int)std::floor(center) - half_len + 1;
int end = (int)std::floor(center) + half_len;
double sum = 0.0;
double wgt = 0.0;
for (int j = start; j <= end; j++)
{
double d = center - (double)j;
double sinc_val;
if (std::fabs(d) < 1e-9)
sinc_val = 2.0 * fc;
else
sinc_val = std::sin(2.0 * M_PI * fc * d) / (M_PI * d);
double t = d / (double)half_len;
double win;
if (t < -1.0 || t > 1.0)
win = 0.0;
else
win = audio_resample_bessel_i0(beta * std::sqrt(1.0 - t * t)) * inv_i0b;
double h = sinc_val * win;
int idx = j;
if (idx < 0) idx = 0;
if (idx >= n_in) idx = n_in - 1;
sum += src[idx] * h;
wgt += h;
}
dst[i] = (wgt > 1e-12) ? (float)(sum / wgt) : 0.0f;
}
else
{
const double frac = src_pos - idx;
const float s0 = input[idx];
const float s1 = input[idx + 1];
output[i] = static_cast<float>(s0 + (s1 - s0) * frac);
}
src_pos += step;
}
return output;
}