зеркало из https://github.com/mozilla/gecko-dev.git
199 строки
8.1 KiB
C++
199 строки
8.1 KiB
C++
/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
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/* This Source Code Form is subject to the terms of the Mozilla Public
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* License, v. 2.0. If a copy of the MPL was not distributed with this file,
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* You can obtain one at http://mozilla.org/MPL/2.0/. */
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#ifndef MOZILLA_AUDIO_DRIFT_CORRECTION_H_
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#define MOZILLA_AUDIO_DRIFT_CORRECTION_H_
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#include "DynamicResampler.h"
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namespace mozilla {
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/**
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* ClockDrift calculates the diverge of the source clock from the nominal
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* (provided) rate compared to the target clock, which is considered the master
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* clock. In the case of different sampling rates, it is assumed that resampling
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* will take place so the returned correction is estimated after the resampling.
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* That means that resampling is taken into account in the calculations but it
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* does appear in the correction. The correction must be applied to the top of
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* the resampling.
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*
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* It works by measuring the incoming, the outgoing frames, and the amount of
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* buffered data and estimates the correction needed. The correction logic has
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* been created with two things in mind. First, not to run out of frames because
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* that means the audio will glitch. Second, not to change the correction very
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* often because this will result in a change in the resampling ratio. The
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* resampler recreates its internal memory when the ratio changes which has a
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* performance impact.
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*
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* The pref `media.clock drift.buffering` can be used to configure the desired
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* internal buffering. Right now it is at 5ms. But it can be increased if there
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* are audio quality problems.
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*/
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class ClockDrift final {
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public:
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/**
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* Provide the nominal source and the target sample rate.
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*/
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ClockDrift(int32_t aSourceRate, int32_t aTargetRate)
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: mSourceRate(aSourceRate),
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mTargetRate(aTargetRate),
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mDesiredBuffering(5 * mSourceRate / 100) {
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if (Preferences::HasUserValue("media.clockdrift.buffering")) {
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int msecs = Preferences::GetInt("media.clockdrift.buffering");
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mDesiredBuffering = msecs * mSourceRate / 100;
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}
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}
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/**
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* The correction in the form of a ratio. A correction of 0.98 means that the
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* target is 2% slower compared to the source or 1.03 which means that the
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* target is 3% faster than the source.
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*/
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float GetCorrection() { return mCorrection; }
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/**
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* Update the available source frames, target frames, and the current
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* buffering, in every iteration. If the condition are met a new correction is
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* calculated. A new correction is calculated in the following cases:
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* 1. Every 100 iterations which mean every 100 calls of this method.
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* 2. Every time we run out of buffered frames (less than 2ms).
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* In addition to that, the correction is clamped to 10% to avoid sound
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* distortion so the result will be in [0.9, 1.1].
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*/
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void UpdateClock(int aSourceClock, int aTargetClock, int aBufferedFrames) {
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if (mIterations == mAdjustementWindow) {
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CalculateCorrection(aBufferedFrames);
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} else if (aBufferedFrames < 2 * mSourceRate / 100 /*20ms*/) {
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BufferedFramesCorrection(aBufferedFrames);
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}
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mTargetClock += aTargetClock;
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mSourceClock += aSourceClock;
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++mIterations;
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}
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private:
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void CalculateCorrection(int aBufferedFrames) {
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// We want to maintain 4 ms buffered
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int32_t bufferedFramesDiff = aBufferedFrames - mDesiredBuffering;
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int32_t resampledSourceClock = mSourceClock + bufferedFramesDiff;
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if (mTargetRate != mSourceRate) {
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resampledSourceClock =
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resampledSourceClock *
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(static_cast<float>(mTargetRate) / static_cast<float>(mSourceRate));
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}
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mCorrection = (float)mTargetClock / resampledSourceClock;
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// Clamp to ragnge [0.9, 1.1] to avoid distortion
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mCorrection = std::min(std::max(mCorrection, 0.9f), 1.1f);
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// If previous correction slightly smaller (-1%) ignore it to avoid
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// recalculations. Don't do it when is greater (+1%) to avoid risking
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// running out of frames.
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if (mPreviousCorrection - mCorrection <= 0.01 &&
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mPreviousCorrection - mCorrection > 0) {
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mCorrection = mPreviousCorrection;
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}
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mPreviousCorrection = mCorrection;
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// Reset the counters to preper for the new period.
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mIterations = 0;
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mTargetClock = 0;
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mSourceClock = 0;
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}
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void BufferedFramesCorrection(int aBufferedFrames) {
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int32_t bufferedFramesDiff = aBufferedFrames - mDesiredBuffering;
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int32_t resampledSourceClock = mSourceRate + bufferedFramesDiff;
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if (mTargetRate != mSourceRate) {
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resampledSourceClock = resampledSourceClock *
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(static_cast<float>(mTargetRate) / mSourceRate);
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}
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MOZ_ASSERT(mTargetRate > resampledSourceClock);
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mPreviousCorrection = mCorrection;
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mCorrection +=
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static_cast<float>(mTargetRate) / resampledSourceClock - 1.0f;
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// Clamp to range [0.9, 1.1] to avoid distortion
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mCorrection = std::min(std::max(mCorrection, 0.9f), 1.1f);
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}
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private:
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const int32_t mSourceRate;
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const int32_t mTargetRate;
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float mCorrection = 1.0;
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float mPreviousCorrection = 1.0;
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const int32_t mAdjustementWindow = 100;
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int32_t mDesiredBuffering; // defult: 5ms
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int32_t mSourceClock = 0;
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int32_t mTargetClock = 0;
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int32_t mIterations = 0;
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};
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/**
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* Correct the drift between two independent clocks, the source, and the target
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* clock. The target clock is the master clock so the correction syncs the drift
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* of the source clock to the target. The nominal sampling rates of source and
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* target must be provided. If the source and the target operate in different
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* sample rate the drift correction will be performed on the top of resampling
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* from the source rate to the target rate.
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*
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* It works with AudioSegment in order to be able to be used from the
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* MediaTrackGraph/MediaTrack. The audio buffers are pre-allocated so there is
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* no new allocation takes place during operation. The preallocation capacity is
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* 100ms for input and 100ms for output. The class consists of ClockDrift and
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* AudioResampler check there for more details.
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*
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* The class is not thread-safe. The construction can happen in any thread but
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* the member method must be used in a single thread that can be different than
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* the construction thread. Appropriate for being used in the high priority
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* audio thread.
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*/
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class AudioDriftCorrection final {
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public:
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AudioDriftCorrection(int32_t aSourceRate, int32_t aTargetRate)
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: mClockDrift(aSourceRate, aTargetRate),
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mResampler(aSourceRate, aTargetRate, aTargetRate / 20 /*50ms*/),
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mTargetRate(aTargetRate) {}
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/**
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* The source audio frames and request the number of target audio frames must
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* be provided. The duration of the source and the output is considered as the
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* source clock and the target clock. The input is buffered internally so some
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* latency exists. The returned AudioSegment must be cleaned up because the
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* internal buffer will be reused after 100ms. If the drift correction (and
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* possible resampling) is not possible due to lack of input data an empty
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* AudioSegment will be returned. Not thread-safe.
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*/
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AudioSegment RequestFrames(const AudioSegment& aInput,
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int32_t aOutputFrames) {
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// Very important to go first since the Dynamic will get the sample format
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// from the chunk.
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if (aInput.GetDuration()) {
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// Always go through the resampler because the clock might shift later.
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mResampler.AppendInput(aInput);
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}
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mClockDrift.UpdateClock(aInput.GetDuration(), aOutputFrames,
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mResampler.InputDuration());
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TrackRate receivingRate = mTargetRate * mClockDrift.GetCorrection();
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// Update resampler's rate if there is a new correction.
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mResampler.UpdateOutRate(receivingRate);
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// If it does not have enough frames the result will be an empty segment.
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AudioSegment output = mResampler.Resample(aOutputFrames);
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if (output.IsEmpty()) {
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output.AppendNullData(aOutputFrames);
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}
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return output;
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}
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private:
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ClockDrift mClockDrift;
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AudioResampler mResampler;
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const int32_t mTargetRate;
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};
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}; // namespace mozilla
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#endif /* MOZILLA_AUDIO_DRIFT_CORRECTION_H_ */
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