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interpolate: Interpolate on a frame-by-frame basis
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035716d57b
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933508e2a2
@ -244,17 +244,27 @@ void Source::GenerateFrame() {
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break;
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}
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const size_t size_to_copy =
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std::min(state.current_buffer.size(), current_frame.size() - frame_position);
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std::copy(state.current_buffer.begin(), state.current_buffer.begin() + size_to_copy,
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current_frame.begin() + frame_position);
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state.current_buffer.erase(state.current_buffer.begin(),
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state.current_buffer.begin() + size_to_copy);
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frame_position += size_to_copy;
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state.next_sample_number += static_cast<u32>(size_to_copy);
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switch (state.interpolation_mode) {
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case InterpolationMode::None:
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AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier,
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current_frame, frame_position);
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break;
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case InterpolationMode::Linear:
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AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
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current_frame, frame_position);
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break;
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case InterpolationMode::Polyphase:
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// TODO(merry): Implement polyphase interpolation
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LOG_DEBUG(Audio_DSP, "Polyphase interpolation unimplemented; falling back to linear");
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AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier,
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current_frame, frame_position);
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break;
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default:
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UNIMPLEMENTED();
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break;
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}
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}
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state.next_sample_number += frame_position;
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state.filters.ProcessFrame(current_frame);
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}
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@ -305,25 +315,6 @@ bool Source::DequeueBuffer() {
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return true;
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}
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switch (state.interpolation_mode) {
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case InterpolationMode::None:
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state.current_buffer =
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AudioInterp::None(state.interp_state, state.current_buffer, state.rate_multiplier);
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break;
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case InterpolationMode::Linear:
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state.current_buffer =
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AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
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break;
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case InterpolationMode::Polyphase:
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// TODO(merry): Implement polyphase interpolation
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state.current_buffer =
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AudioInterp::Linear(state.interp_state, state.current_buffer, state.rate_multiplier);
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break;
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default:
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UNIMPLEMENTED();
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break;
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}
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// the first playthrough starts at play_position, loops start at the beginning of the buffer
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state.current_sample_number = (!buf.has_played) ? buf.play_position : 0;
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state.next_sample_number = state.current_sample_number;
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@ -13,74 +13,64 @@ namespace AudioInterp {
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constexpr u64 scale_factor = 1 << 24;
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constexpr u64 scale_mask = scale_factor - 1;
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/// Here we step over the input in steps of rate_multiplier, until we consume all of the input.
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/// Here we step over the input in steps of rate, until we consume all of the input.
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/// Three adjacent samples are passed to fn each step.
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template <typename Function>
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static StereoBuffer16 StepOverSamples(State& state, const StereoBuffer16& input,
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float rate_multiplier, Function fn) {
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ASSERT(rate_multiplier > 0);
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static void StepOverSamples(State& state, StereoBuffer16& input, float rate,
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DSP::HLE::StereoFrame16& output, size_t& outputi, Function fn) {
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ASSERT(rate > 0);
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if (input.size() < 2)
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return {};
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if (input.empty())
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return;
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StereoBuffer16 output;
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output.reserve(static_cast<size_t>(input.size() / rate_multiplier));
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input.insert(input.begin(), {state.xn2, state.xn1});
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u64 step_size = static_cast<u64>(rate_multiplier * scale_factor);
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const u64 step_size = static_cast<u64>(rate * scale_factor);
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u64 fposition = state.fposition;
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size_t inputi = 0;
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u64 fposition = 0;
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const u64 max_fposition = input.size() * scale_factor;
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while (outputi < output.size()) {
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inputi = static_cast<size_t>(fposition / scale_factor);
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if (inputi + 2 >= input.size()) {
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inputi = input.size() - 2;
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break;
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}
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while (fposition < 1 * scale_factor) {
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u64 fraction = fposition & scale_mask;
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output.push_back(fn(fraction, state.xn2, state.xn1, input[0]));
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output[outputi++] = fn(fraction, input[inputi], input[inputi + 1], input[inputi + 2]);
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fposition += step_size;
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}
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while (fposition < 2 * scale_factor) {
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u64 fraction = fposition & scale_mask;
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state.xn2 = input[inputi];
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state.xn1 = input[inputi + 1];
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state.fposition = fposition - inputi * scale_factor;
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output.push_back(fn(fraction, state.xn1, input[0], input[1]));
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fposition += step_size;
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}
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while (fposition < max_fposition) {
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u64 fraction = fposition & scale_mask;
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size_t index = static_cast<size_t>(fposition / scale_factor);
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output.push_back(fn(fraction, input[index - 2], input[index - 1], input[index]));
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fposition += step_size;
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}
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state.xn2 = input[input.size() - 2];
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state.xn1 = input[input.size() - 1];
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return output;
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input.erase(input.begin(), input.begin() + inputi + 2);
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}
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StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier) {
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return StepOverSamples(
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state, input, rate_multiplier,
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void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
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size_t& outputi) {
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StepOverSamples(
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state, input, rate, output, outputi,
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[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) { return x0; });
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}
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StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier) {
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void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
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size_t& outputi) {
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// Note on accuracy: Some values that this produces are +/- 1 from the actual firmware.
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return StepOverSamples(state, input, rate_multiplier,
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[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
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// This is a saturated subtraction. (Verified by black-box fuzzing.)
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s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
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s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
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StepOverSamples(state, input, rate, output, outputi,
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[](u64 fraction, const auto& x0, const auto& x1, const auto& x2) {
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// This is a saturated subtraction. (Verified by black-box fuzzing.)
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s64 delta0 = MathUtil::Clamp<s64>(x1[0] - x0[0], -32768, 32767);
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s64 delta1 = MathUtil::Clamp<s64>(x1[1] - x0[1], -32768, 32767);
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return std::array<s16, 2>{
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static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
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static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
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};
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});
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return std::array<s16, 2>{
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static_cast<s16>(x0[0] + fraction * delta0 / scale_factor),
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static_cast<s16>(x0[1] + fraction * delta1 / scale_factor),
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};
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});
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}
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} // namespace AudioInterp
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@ -6,6 +6,7 @@
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#include <array>
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#include <vector>
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#include "audio_core/hle/common.h"
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#include "common/common_types.h"
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namespace AudioInterp {
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@ -14,31 +15,35 @@ namespace AudioInterp {
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using StereoBuffer16 = std::vector<std::array<s16, 2>>;
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struct State {
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// Two historical samples.
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/// Two historical samples.
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std::array<s16, 2> xn1 = {}; ///< x[n-1]
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std::array<s16, 2> xn2 = {}; ///< x[n-2]
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/// Current fractional position.
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u64 fposition = 0;
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};
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/**
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* No interpolation. This is equivalent to a zero-order hold. There is a two-sample predelay.
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* @param state Interpolation state.
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* @param input Input buffer.
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* @param rate_multiplier Stretch factor. Must be a positive non-zero value.
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* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
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* performs upsampling.
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* @return The resampled audio buffer.
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* @param rate Stretch factor. Must be a positive non-zero value.
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* rate > 1.0 performs decimation and rate < 1.0 performs upsampling.
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* @param output The resampled audio buffer.
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* @param outputi The index of output to start writing to.
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*/
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StereoBuffer16 None(State& state, const StereoBuffer16& input, float rate_multiplier);
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void None(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
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size_t& outputi);
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/**
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* Linear interpolation. This is equivalent to a first-order hold. There is a two-sample predelay.
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* @param state Interpolation state.
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* @param input Input buffer.
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* @param rate_multiplier Stretch factor. Must be a positive non-zero value.
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* rate_multiplier > 1.0 performs decimation and rate_multipler < 1.0
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* performs upsampling.
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* @return The resampled audio buffer.
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* @param rate Stretch factor. Must be a positive non-zero value.
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* rate > 1.0 performs decimation and rate < 1.0 performs upsampling.
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* @param output The resampled audio buffer.
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* @param outputi The index of output to start writing to.
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*/
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StereoBuffer16 Linear(State& state, const StereoBuffer16& input, float rate_multiplier);
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void Linear(State& state, StereoBuffer16& input, float rate, DSP::HLE::StereoFrame16& output,
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size_t& outputi);
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} // namespace AudioInterp
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