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-rw-r--r--src/audio/resample.cpp116
1 files changed, 59 insertions, 57 deletions
diff --git a/src/audio/resample.cpp b/src/audio/resample.cpp
index 7accd0a1..430a6a26 100644
--- a/src/audio/resample.cpp
+++ b/src/audio/resample.cpp
@@ -1,4 +1,17 @@
+/*
+ * Copyright 2023 jacqueline <me@jacqueline.id.au>
+ *
+ * SPDX-License-Identifier: GPL-3.0-only
+ */
#include "resample.hpp"
+/*
+ * This file contains the implementation for a 32-bit floating point resampler.
+ * It is largely based on David Bryant's ART resampler, which is BSD-licensed,
+ * and available at https://github.com/dbry/audio-resampler/.
+ *
+ * This resampler uses windowed sinc interpolation filters, with an additional
+ * lowpass filter to reduce aliasing.
+ */
#include <stdint.h>
#include <stdlib.h>
@@ -14,13 +27,11 @@
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;
+static constexpr size_t kFilterSize = 16;
-typedef std::array<float, kTapsPerFilter> Filter;
+typedef std::array<float, kFilterSize> Filter;
static std::array<Filter, kNumFilters + 1> sFilters{};
static bool sFiltersInitialised = false;
@@ -35,15 +46,15 @@ Resampler::Resampler(uint32_t source_sample_rate,
static_cast<double>(source_sample_rate)),
num_channels_(num_channels) {
channel_buffers_.resize(num_channels);
- channel_buffer_size_ = kTapsPerFilter * 16;
+ channel_buffer_size_ = kFilterSize * 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;
+ output_offset_ = kFilterSize / 2.0f;
+ input_index_ = kFilterSize;
if (!sFiltersInitialised) {
sFiltersInitialised = true;
@@ -64,7 +75,7 @@ auto Resampler::Process(cpp::span<const sample::Sample> input,
size_t input_frames = input.size() / num_channels_;
size_t output_frames = output.size() / num_channels_;
- int half_taps = kTapsPerFilter / 2;
+ int half_taps = kFilterSize / 2;
while (output_frames > 0) {
if (output_offset_ >= input_index_ - half_taps) {
if (input_frames > 0) {
@@ -74,12 +85,12 @@ auto Resampler::Process(cpp::span<const sample::Sample> input,
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));
+ channel_buffers_[i] + channel_buffer_size_ - kFilterSize,
+ kFilterSize * sizeof(float));
}
- output_offset_ -= channel_buffer_size_ - kTapsPerFilter;
- input_index_ -= channel_buffer_size_ - kTapsPerFilter;
+ output_offset_ -= channel_buffer_size_ - kFilterSize;
+ input_index_ -= channel_buffer_size_ - kFilterSize;
}
for (int i = 0; i < num_channels_; ++i) {
@@ -97,7 +108,11 @@ auto Resampler::Process(cpp::span<const sample::Sample> input,
output[samples_produced++] = sample::FromFloat(Subsample(i));
}
- output_offset_ += (1.0f / factor_);
+ // NOTE: floating point division here is potentially slow due to FPU
+ // limitations. Consider explicitly bunding the xtensa libgcc divsion via
+ // reciprocal implementation if we care about portability between
+ // compilers.
+ output_offset_ += 1.0f / factor_;
output_frames--;
}
}
@@ -105,36 +120,34 @@ auto Resampler::Process(cpp::span<const sample::Sample> input,
return {samples_used, samples_produced};
}
+/*
+ * Constructs the filter in-place for the given index of sFilters. This only
+ * needs to be done once, per-filter. 64-bit math is okay here, because filters
+ * will not be initialised within a performance critical path.
+ */
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;
+ Filter& filter = sFilters[index];
+ std::array<double, kFilterSize> working_buffer{};
- double fraction =
- static_cast<double>(index) / static_cast<double>(kNumFilters);
+ double fraction = 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.
+ for (int i = 0; i < kFilterSize; ++i) {
+ // "dist" is the absolute distance from the sinc maximum to the filter tap
+ // to be calculated, in radians.
+ double dist = fabs((kFilterSize / 2.0 - 1.0) + fraction - i) * M_PI;
+ // "ratio" is that distance divided by half the tap count such that it
+ // reaches π at the window extremes
+ double ratio = dist / (kFilterSize / 2.0);
- 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);
+ // Hann window. We could alternatively use a Blackman Harris window,
+ // however our unusually small filter size makes the Hann window's
+ // steeper cutoff more important.
+ value *= 0.5 * (1.0 + cos(ratio));
} else {
value = 1.0;
}
@@ -143,50 +156,39 @@ auto InitFilter(int index) -> void {
filter_sum += value;
}
- // filter should have unity DC gain
-
+ // 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)) {
+ for (int i = kFilterSize / 2; i < kFilterSize;
+ i = kFilterSize - i - (i >= kFilterSize / 2)) {
working_buffer[i] *= scaler;
filter[i] = working_buffer[i] - error;
error += static_cast<double>(filter[i]) - working_buffer[i];
}
}
+/*
+ * Performs sub-sampling with interpolation for the given channel. Assumes that
+ * the channel buffer has already been filled with samples.
+ */
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});
+ float sum1 = ApplyFilter(sFilters[filter_index],
+ {source.data() - kFilterSize / 2 + 1, kFilterSize});
offset_fractional -= filter_index;
- sum2 = ApplyFilter(sFilters[filter_index + 1],
- {source.data() - kTapsPerFilter / 2 + 1, kTapsPerFilter});
+ float sum2 = ApplyFilter(sFilters[filter_index + 1],
+ {source.data() - kFilterSize / 2 + 1, kFilterSize});
return (sum2 * offset_fractional) + (sum1 * (1.0f - offset_fractional));
}
@@ -194,7 +196,7 @@ return ApplyFilter(
auto Resampler::ApplyFilter(cpp::span<float> filter, cpp::span<float> input)
-> float {
float sum = 0.0;
- for (int i = 0; i < kTapsPerFilter; i++) {
+ for (int i = 0; i < kFilterSize; i++) {
sum += filter[i] * input[i];
}
return sum;
generated by cgit v1.2.3 (git 2.25.1) at 2026-04-30 00:37:54 +0000