time_stretch: Simplify audio stretcher

This commit is contained in:
MerryMage 2018-09-08 21:28:19 +01:00
parent f34711219a
commit eed55a813e
4 changed files with 88 additions and 162 deletions

View File

@ -15,6 +15,7 @@ DspInterface::DspInterface() = default;
DspInterface::~DspInterface() = default;
void DspInterface::SetSink(const std::string& sink_id, const std::string& audio_device) {
sink.reset();
const SinkDetails& sink_details = GetSinkDetails(sink_id);
sink = sink_details.factory(audio_device);
sink->SetCallback(
@ -32,7 +33,7 @@ void DspInterface::EnableStretching(bool enable) {
return;
if (!enable) {
FlushResidualStretcherAudio();
flushing_time_stretcher = true;
}
perform_time_stretching = enable;
}
@ -51,17 +52,27 @@ void DspInterface::OutputFrame(StereoFrame16& frame) {
fifo.Push(frame.data(), frame.size());
}
void DspInterface::FlushResidualStretcherAudio() {}
void DspInterface::OutputCallback(s16* buffer, size_t num_frames) {
const size_t frames_written = fifo.Pop(buffer, num_frames);
void DspInterface::OutputCallback(s16* buffer, std::size_t num_frames) {
std::size_t frames_written;
if (perform_time_stretching) {
const std::vector<s16> in{fifo.Pop()};
const std::size_t num_in{in.size() / 2};
frames_written = time_stretcher.Process(in.data(), num_in, buffer, num_frames);
} else if (flushing_time_stretcher) {
time_stretcher.Flush();
frames_written = time_stretcher.Process(nullptr, 0, buffer, num_frames);
frames_written += fifo.Pop(buffer, num_frames - frames_written);
flushing_time_stretcher = false;
} else {
frames_written = fifo.Pop(buffer, num_frames);
}
if (frames_written > 0) {
std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16));
}
// Hold last emitted frame; this prevents popping.
for (size_t i = frames_written; i < num_frames; i++) {
for (std::size_t i = frames_written; i < num_frames; i++) {
std::memcpy(buffer + 2 * i, &last_frame[0], 2 * sizeof(s16));
}
}

View File

@ -85,7 +85,8 @@ private:
void OutputCallback(s16* buffer, std::size_t num_frames);
std::unique_ptr<Sink> sink;
bool perform_time_stretching = false;
std::atomic<bool> perform_time_stretching = false;
std::atomic<bool> flushing_time_stretcher = false;
Common::RingBuffer<s16, 0x2000, 2> fifo;
std::array<s16, 2> last_frame{};
TimeStretcher time_stretcher;

View File

@ -3,143 +3,75 @@
// Refer to the license.txt file included.
#include <algorithm>
#include <chrono>
#include <cmath>
#include <vector>
#include <cstddef>
#include <memory>
#include <SoundTouch.h>
#include "audio_core/audio_types.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/logging/log.h"
using steady_clock = std::chrono::steady_clock;
namespace AudioCore {
constexpr double MIN_RATIO = 0.1;
constexpr double MAX_RATIO = 100.0;
static double ClampRatio(double ratio) {
return std::clamp(ratio, MIN_RATIO, MAX_RATIO);
TimeStretcher::TimeStretcher()
: sample_rate(native_sample_rate), sound_touch(std::make_unique<soundtouch::SoundTouch>()) {
sound_touch->setChannels(2);
sound_touch->setSampleRate(native_sample_rate);
sound_touch->setPitch(1.0);
sound_touch->setTempo(1.0);
}
constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
constexpr std::size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
constexpr double SMOOTHING_FACTOR = 0.007;
struct TimeStretcher::Impl {
soundtouch::SoundTouch soundtouch;
steady_clock::time_point frame_timer = steady_clock::now();
std::size_t samples_queued = 0;
double smoothed_ratio = 1.0;
double sample_rate = static_cast<double>(native_sample_rate);
};
std::vector<s16> TimeStretcher::Process(std::size_t samples_in_queue) {
// This is a very simple algorithm without any fancy control theory. It works and is stable.
double ratio = CalculateCurrentRatio();
ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
impl->smoothed_ratio =
(1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
// SoundTouch's tempo definition the inverse of our ratio definition.
impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio);
std::vector<s16> samples = GetSamples();
if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) {
samples.clear();
LOG_DEBUG(Audio, "Dropping frames!");
}
return samples;
}
TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) {
impl->soundtouch.setPitch(1.0);
impl->soundtouch.setChannels(2);
impl->soundtouch.setSampleRate(native_sample_rate);
Reset();
}
TimeStretcher::~TimeStretcher() {
impl->soundtouch.clear();
}
TimeStretcher::~TimeStretcher() = default;
void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) {
impl->sample_rate = static_cast<double>(sample_rate);
impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate);
sound_touch->setSampleRate(sample_rate);
sample_rate = native_sample_rate;
}
void TimeStretcher::AddSamples(const s16* buffer, std::size_t num_samples) {
impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
impl->samples_queued += num_samples;
std::size_t TimeStretcher::Process(const s16* in, std::size_t num_in, s16* out,
std::size_t num_out) {
const double time_delta = static_cast<double>(num_out) / sample_rate; // seconds
double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
const double max_latency = 0.25; // seconds
const double max_backlog = sample_rate * max_latency;
const double backlog_fullness = sound_touch->numSamples() / max_backlog;
if (backlog_fullness > 4.0) {
// Too many samples in backlog: Don't push anymore on
num_in = 0;
}
// We ideally want the backlog to be about 50% full.
// This gives some headroom both ways to prevent underflow and overflow.
// We tweak current_ratio to encourage this.
constexpr double tweak_time_scale = 0.050; // seconds
const double tweak_correction = (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
current_ratio *= std::pow(1.0 + 2.0 * tweak_correction, tweak_correction < 0 ? 3.0 : 1.0);
// This low-pass filter smoothes out variance in the calculated stretch ratio.
// The time-scale determines how responsive this filter is.
constexpr double lpf_time_scale = 0.712; // seconds
const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
stretch_ratio += lpf_gain * (current_ratio - stretch_ratio);
// Place a lower limit of 5% speed. When a game boots up, there will be
// many silence samples. These do not need to be timestretched.
stretch_ratio = std::max(stretch_ratio, 0.05);
sound_touch->setTempo(stretch_ratio);
LOG_DEBUG(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, stretch_ratio,
backlog_fullness);
sound_touch->putSamples(in, num_in);
return sound_touch->receiveSamples(out, num_out);
}
void TimeStretcher::Clear() {
sound_touch->clear();
}
void TimeStretcher::Flush() {
impl->soundtouch.flush();
}
void TimeStretcher::Reset() {
impl->soundtouch.setTempo(1.0);
impl->soundtouch.clear();
impl->smoothed_ratio = 1.0;
impl->frame_timer = steady_clock::now();
impl->samples_queued = 0;
SetOutputSampleRate(native_sample_rate);
}
double TimeStretcher::CalculateCurrentRatio() {
const steady_clock::time_point now = steady_clock::now();
const std::chrono::duration<double> duration = now - impl->frame_timer;
const double expected_time =
static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
const double actual_time = duration.count();
double ratio;
if (expected_time != 0) {
ratio = ClampRatio(actual_time / expected_time);
} else {
ratio = impl->smoothed_ratio;
}
impl->frame_timer = now;
impl->samples_queued = 0;
return ratio;
}
double TimeStretcher::CorrectForUnderAndOverflow(double ratio, std::size_t sample_delay) const {
const std::size_t min_sample_delay =
static_cast<std::size_t>(MIN_DELAY_TIME * impl->sample_rate);
const std::size_t max_sample_delay =
static_cast<std::size_t>(MAX_DELAY_TIME * impl->sample_rate);
if (sample_delay < min_sample_delay) {
// Make the ratio bigger.
ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio);
} else if (sample_delay > max_sample_delay) {
// Make the ratio smaller.
ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio;
}
return ClampRatio(ratio);
}
std::vector<s16> TimeStretcher::GetSamples() {
uint available = impl->soundtouch.numSamples();
std::vector<s16> output(static_cast<std::size_t>(available) * 2);
impl->soundtouch.receiveSamples(output.data(), available);
return output;
sound_touch->flush();
}
} // namespace AudioCore

View File

@ -4,57 +4,39 @@
#pragma once
#include <array>
#include <cstddef>
#include <memory>
#include <vector>
#include "common/common_types.h"
namespace soundtouch {
class SoundTouch;
}
namespace AudioCore {
class TimeStretcher final {
class TimeStretcher {
public:
TimeStretcher();
~TimeStretcher();
/**
* Set sample rate for the samples that Process returns.
* @param sample_rate The sample rate.
*/
void SetOutputSampleRate(unsigned int sample_rate);
/**
* Add samples to be processed.
* @param sample_buffer Buffer of samples in interleaved stereo PCM16 format.
* @param num_samples Number of samples.
*/
void AddSamples(const s16* sample_buffer, std::size_t num_samples);
/// @param in Input sample buffer
/// @param num_in Number of input frames in `in`
/// @param out Output sample buffer
/// @param num_out Desired number of output frames in `out`
/// @returns Actual number of frames written to `out`
std::size_t Process(const s16* in, std::size_t num_in, s16* out, std::size_t num_out);
void Clear();
/// Flush audio remaining in internal buffers.
void Flush();
/// Resets internal state and clears buffers.
void Reset();
/**
* Does audio stretching and produces the time-stretched samples.
* Timer calculations use sample_delay to determine how much of a margin we have.
* @param sample_delay How many samples are buffered downstream of this module and haven't been
* played yet.
* @return Samples to play in interleaved stereo PCM16 format.
*/
std::vector<s16> Process(std::size_t sample_delay);
private:
struct Impl;
std::unique_ptr<Impl> impl;
/// INTERNAL: ratio = wallclock time / emulated time
double CalculateCurrentRatio();
/// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate
/// direction.
double CorrectForUnderAndOverflow(double ratio, std::size_t sample_delay) const;
/// INTERNAL: Gets the time-stretched samples from SoundTouch.
std::vector<s16> GetSamples();
unsigned int sample_rate;
std::unique_ptr<soundtouch::SoundTouch> sound_touch;
double stretch_ratio = 1.0;
};
} // namespace AudioCore