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#include "resample.hpp"
#include <stdint.h>
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <cmath>
#include <numeric>
#include "esp_log.h"
#include "sample.hpp"
#include "stream_info.hpp"
namespace audio {
static constexpr char kTag[] = "resample";
static constexpr double kLowPassRatio = 0.5;
static constexpr size_t kNumFilters = 64;
static constexpr size_t kTapsPerFilter = 16;
typedef std::array<float, kTapsPerFilter> Filter;
static std::array<Filter, kNumFilters + 1> sFilters{};
static bool sFiltersInitialised = false;
auto InitFilter(int index) -> void;
Resampler::Resampler(uint32_t source_sample_rate,
uint32_t target_sample_rate,
uint8_t num_channels)
: source_sample_rate_(source_sample_rate),
target_sample_rate_(target_sample_rate),
factor_(static_cast<double>(target_sample_rate) /
static_cast<double>(source_sample_rate)),
num_channels_(num_channels) {
channel_buffers_.resize(num_channels);
channel_buffer_size_ = kTapsPerFilter * 16;
for (int i = 0; i < num_channels; i++) {
channel_buffers_[i] =
static_cast<float*>(calloc(sizeof(float), channel_buffer_size_));
}
output_offset_ = kTapsPerFilter / 2.0f;
input_index_ = kTapsPerFilter;
if (!sFiltersInitialised) {
sFiltersInitialised = true;
for (int i = 0; i < kNumFilters + 1; i++) {
InitFilter(i);
}
}
}
Resampler::~Resampler() {}
auto Resampler::Process(cpp::span<const sample::Sample> input,
cpp::span<sample::Sample> output,
bool end_of_data) -> std::pair<size_t, size_t> {
size_t samples_used = 0;
size_t samples_produced = 0;
size_t input_frames = input.size() / num_channels_;
size_t output_frames = output.size() / num_channels_;
int half_taps = kTapsPerFilter / 2;
while (output_frames > 0) {
if (output_offset_ >= input_index_ - half_taps) {
if (input_frames > 0) {
// Check whether the channel buffers will overflow with the addition of
// this sample. If so, we need to move the remaining contents back to
// the beginning of the buffer.
if (input_index_ == channel_buffer_size_) {
for (int i = 0; i < num_channels_; ++i) {
memmove(channel_buffers_[i],
channel_buffers_[i] + channel_buffer_size_ - kTapsPerFilter,
kTapsPerFilter * sizeof(float));
}
output_offset_ -= channel_buffer_size_ - kTapsPerFilter;
input_index_ -= channel_buffer_size_ - kTapsPerFilter;
}
for (int i = 0; i < num_channels_; ++i) {
channel_buffers_[i][input_index_] =
sample::ToFloat(input[samples_used++]);
}
input_index_++;
input_frames--;
} else {
break;
}
} else {
for (int i = 0; i < num_channels_; i++) {
output[samples_produced++] = sample::FromFloat(Subsample(i));
}
output_offset_ += (1.0f / factor_);
output_frames--;
}
}
return {samples_used, samples_produced};
}
auto InitFilter(int index) -> void {
const double a0 = 0.35875;
const double a1 = 0.48829;
const double a2 = 0.14128;
const double a3 = 0.01168;
double fraction =
static_cast<double>(index) / static_cast<double>(kNumFilters);
double filter_sum = 0.0;
// "dist" is the absolute distance from the sinc maximum to the filter tap to
// be calculated, in radians "ratio" is that distance divided by half the tap
// count such that it reaches π at the window extremes
// Note that with this scaling, the odd terms of the Blackman-Harris
// calculation appear to be negated with respect to the reference formula
// version.
Filter& filter = sFilters[index];
std::array<double, kTapsPerFilter> working_buffer{};
for (int i = 0; i < kTapsPerFilter; ++i) {
double dist = fabs((kTapsPerFilter / 2.0 - 1.0) + fraction - i) * M_PI;
double ratio = dist / (kTapsPerFilter / 2.0);
double value;
if (dist != 0.0) {
value = sin(dist * kLowPassRatio) / (dist * kLowPassRatio);
// Blackman-Harris window
value *= a0 + a1 * cos(ratio) + a2 * cos(2 * ratio) + a3 * cos(3 * ratio);
} else {
value = 1.0;
}
working_buffer[i] = value;
filter_sum += value;
}
// filter should have unity DC gain
double scaler = 1.0 / filter_sum;
double error = 0.0;
for (int i = kTapsPerFilter / 2; i < kTapsPerFilter;
i = kTapsPerFilter - i - (i >= kTapsPerFilter / 2)) {
working_buffer[i] *= scaler;
filter[i] = working_buffer[i] - error;
error += static_cast<double>(filter[i]) - working_buffer[i];
}
}
auto Resampler::Subsample(int channel) -> float {
float sum1, sum2;
cpp::span<float> source{channel_buffers_[channel], channel_buffer_size_};
int offset_integral = std::floor(output_offset_);
source = source.subspan(offset_integral);
float offset_fractional = output_offset_ - offset_integral;
/*
// no interpolate
size_t filter_index = std::floor(offset_fractional * kNumFilters + 0.5f);
//ESP_LOGI(kTag, "selected filter %u of %u", filter_index, kNumFilters);
int start_offset = kTapsPerFilter / 2 + 1;
//ESP_LOGI(kTag, "using offset of %i, length %u", start_offset, kTapsPerFilter);
return ApplyFilter(
sFilters[filter_index],
{source.data() - start_offset, kTapsPerFilter});
*/
offset_fractional *= kNumFilters;
int filter_index = std::floor(offset_fractional);
sum1 = ApplyFilter(sFilters[filter_index],
{source.data() - kTapsPerFilter / 2 + 1, kTapsPerFilter});
offset_fractional -= filter_index;
sum2 = ApplyFilter(sFilters[filter_index + 1],
{source.data() - kTapsPerFilter / 2 + 1, kTapsPerFilter});
return (sum2 * offset_fractional) + (sum1 * (1.0f - offset_fractional));
}
auto Resampler::ApplyFilter(cpp::span<float> filter, cpp::span<float> input)
-> float {
float sum = 0.0;
for (int i = 0; i < kTapsPerFilter; i++) {
sum += filter[i] * input[i];
}
return sum;
}
} // namespace audio
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