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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
#include "AudioDriftCorrection.h"
#include <cmath>
#include "AudioResampler.h"
#include "DriftController.h"
namespace mozilla {
extern LazyLogModule gMediaTrackGraphLog;
#define LOG_CONTROLLER(level, controller, format, ...) \
MOZ_LOG(gMediaTrackGraphLog, level, \
("DriftController %p: (plot-id %u) " format, controller, \
(controller)->mPlotId, ##__VA_ARGS__))
static media::TimeUnit DesiredBuffering(media::TimeUnit aSourceLatency) {
constexpr media::TimeUnit kMinBuffer(10, MSECS_PER_S);
constexpr media::TimeUnit kMaxBuffer(2500, MSECS_PER_S);
const auto clamped = std::clamp(aSourceLatency, kMinBuffer, kMaxBuffer);
// Ensure the base is the source's sampling rate.
return clamped.ToBase(aSourceLatency);
}
AudioDriftCorrection::AudioDriftCorrection(
uint32_t aSourceRate, uint32_t aTargetRate,
const PrincipalHandle& aPrincipalHandle)
: mTargetRate(aTargetRate),
mDriftController(MakeUnique<DriftController>(aSourceRate, aTargetRate,
mDesiredBuffering)),
mResampler(MakeUnique<AudioResampler>(aSourceRate, aTargetRate, 0,
aPrincipalHandle)) {}
AudioDriftCorrection::~AudioDriftCorrection() = default;
AudioSegment AudioDriftCorrection::RequestFrames(const AudioSegment& aInput,
uint32_t aOutputFrames) {
const media::TimeUnit inputDuration(aInput.GetDuration(),
mDriftController->mSourceRate);
const media::TimeUnit outputDuration(aOutputFrames, mTargetRate);
if (inputDuration.IsPositive()) {
if (mDesiredBuffering.IsZero()) {
// Start with the desired buffering at at least 50ms, since the drift is
// still unknown. It may be adjust downward later on, when we have adapted
// to the drift more.
const media::TimeUnit desiredBuffering = DesiredBuffering(std::max(
inputDuration * 11 / 10, media::TimeUnit::FromSeconds(0.05)));
LOG_CONTROLLER(LogLevel::Info, mDriftController.get(),
"Initial desired buffering %.2fms",
desiredBuffering.ToSeconds() * 1000.0);
SetDesiredBuffering(desiredBuffering);
} else if (inputDuration > mDesiredBuffering) {
// Input latency is higher than the desired buffering. Increase the
// desired buffering to try to avoid underruns.
if (inputDuration > mSourceLatency) {
const media::TimeUnit desiredBuffering =
DesiredBuffering(inputDuration * 11 / 10);
LOG_CONTROLLER(
LogLevel::Info, mDriftController.get(),
"High observed input latency %.2fms (%" PRId64
" frames). Increasing desired buffering %.2fms->%.2fms frames",
inputDuration.ToSeconds() * 1000.0, aInput.GetDuration(),
mDesiredBuffering.ToSeconds() * 1000.0,
desiredBuffering.ToSeconds() * 1000.0);
SetDesiredBuffering(desiredBuffering);
} else {
const media::TimeUnit desiredBuffering =
DesiredBuffering(mSourceLatency * 11 / 10);
LOG_CONTROLLER(LogLevel::Info, mDriftController.get(),
"Increasing desired buffering %.2fms->%.2fms, "
"based on reported input-latency %.2fms.",
mDesiredBuffering.ToSeconds() * 1000.0,
desiredBuffering.ToSeconds() * 1000.0,
mSourceLatency.ToSeconds() * 1000.0);
SetDesiredBuffering(desiredBuffering);
}
}
mIsHandlingUnderrun = false;
// Very important to go first since DynamicResampler will get the sample
// format from the chunk.
mResampler->AppendInput(aInput);
}
bool hasUnderrun = false;
AudioSegment output = mResampler->Resample(aOutputFrames, &hasUnderrun);
mDriftController->UpdateClock(inputDuration, outputDuration,
CurrentBuffering(), BufferSize());
// Update resampler's rate if there is a new correction.
mResampler->UpdateInRate(mDriftController->GetCorrectedSourceRate());
if (hasUnderrun) {
if (!mIsHandlingUnderrun) {
NS_WARNING("Drift-correction: Underrun");
LOG_CONTROLLER(LogLevel::Info, mDriftController.get(),
"Underrun. Doubling the desired buffering %.2fms->%.2fms",
mDesiredBuffering.ToSeconds() * 1000.0,
(mDesiredBuffering * 2).ToSeconds() * 1000.0);
mIsHandlingUnderrun = true;
++mNumUnderruns;
SetDesiredBuffering(DesiredBuffering(mDesiredBuffering * 2));
mDriftController->ResetAfterUnderrun();
}
}
if (mDriftController->DurationWithinHysteresis() >
mLatencyReductionTimeLimit &&
mDriftController->DurationSinceDesiredBufferingChange() >
mLatencyReductionTimeLimit) {
// We have been stable within hysteresis for a while. Let's reduce the
// desired buffering if we can.
const media::TimeUnit sourceLatency =
mDriftController->MeasuredSourceLatency();
// We target 30% over the measured source latency, a bit higher than how we
// adapt to high source latency.
const media::TimeUnit targetDesiredBuffering =
DesiredBuffering(sourceLatency * 13 / 10);
if (targetDesiredBuffering < mDesiredBuffering) {
// The new target is lower than the current desired buffering. Proceed by
// reducing the difference by 10%, but do it in 10ms-steps so there is a
// chance of reaching the target (by truncation).
const media::TimeUnit diff =
(mDesiredBuffering - targetDesiredBuffering) / 10;
// Apply the 10%-diff and 2ms-steps, but don't go lower than the
// already-decided desired target.
const media::TimeUnit target = std::max(
targetDesiredBuffering, (mDesiredBuffering - diff).ToBase(500));
if (target < mDesiredBuffering) {
LOG_CONTROLLER(
LogLevel::Info, mDriftController.get(),
"Reducing desired buffering because the buffering level is stable. "
"%.2fms->%.2fms. Measured source latency is %.2fms, ideal target "
"is %.2fms.",
mDesiredBuffering.ToSeconds() * 1000.0, target.ToSeconds() * 1000.0,
sourceLatency.ToSeconds() * 1000.0,
targetDesiredBuffering.ToSeconds() * 1000.0);
SetDesiredBuffering(target);
}
}
}
return output;
}
uint32_t AudioDriftCorrection::CurrentBuffering() const {
return mResampler->InputReadableFrames();
}
uint32_t AudioDriftCorrection::BufferSize() const {
return mResampler->InputCapacityFrames();
}
uint32_t AudioDriftCorrection::NumCorrectionChanges() const {
return mDriftController->NumCorrectionChanges();
}
void AudioDriftCorrection::SetSourceLatency(media::TimeUnit aSourceLatency) {
LOG_CONTROLLER(
LogLevel::Info, mDriftController.get(), "SetSourceLatency %.2fms->%.2fms",
mSourceLatency.ToSeconds() * 1000.0, aSourceLatency.ToSeconds() * 1000.0);
mSourceLatency = aSourceLatency;
}
void AudioDriftCorrection::SetDesiredBuffering(
media::TimeUnit aDesiredBuffering) {
mDesiredBuffering = aDesiredBuffering;
mDriftController->SetDesiredBuffering(mDesiredBuffering);
mResampler->SetInputPreBufferFrameCount(
mDesiredBuffering.ToTicksAtRate(mDriftController->mSourceRate));
}
} // namespace mozilla
#undef LOG_CONTROLLER