dom/media/AudioDriftCorrection.h

   1 /* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
   2 /* This Source Code Form is subject to the terms of the Mozilla Public
   3  * License, v. 2.0. If a copy of the MPL was not distributed with this file,
   4  * You can obtain one at http://mozilla.org/MPL/2.0/. */
   5
   6 #ifndef MOZILLA_AUDIO_DRIFT_CORRECTION_H_
   7 #define MOZILLA_AUDIO_DRIFT_CORRECTION_H_
   8
   9 #include "DynamicResampler.h"
  10 #include "mozilla/Preferences.h"
  11
  12 namespace mozilla {
  13
  14 extern LazyLogModule gMediaTrackGraphLog;
  15
  16 /**
  17  * ClockDrift calculates the diverge of the source clock from the nominal
  18  * (provided) rate compared to the target clock, which is considered the master
  19  * clock. In the case of different sampling rates, it is assumed that resampling
  20  * will take place so the returned correction is estimated after the resampling.
  21  * That means that resampling is taken into account in the calculations but it
  22  * does appear in the correction. The correction must be applied to the top of
  23  * the resampling.
  24  *
  25  * It works by measuring the incoming, the outgoing frames, and the amount of
  26  * buffered data and estimates the correction needed. The correction logic has
  27  * been created with two things in mind. First, not to run out of frames because
  28  * that means the audio will glitch. Second, not to change the correction very
  29  * often because this will result in a change in the resampling ratio. The
  30  * resampler recreates its internal memory when the ratio changes which has a
  31  * performance impact.
  32  *
  33  * The pref `media.clock drift.buffering` can be used to configure the desired
  34  * internal buffering. Right now it is at 50ms. But it can be increased if there
  35  * are audio quality problems.
  36  */
  37 class ClockDrift final {
  38  public:
  39   /**
  40    * Provide the nominal source and the target sample rate.
  41    */
  42   ClockDrift(uint32_t aSourceRate, uint32_t aTargetRate,
  43              uint32_t aDesiredBuffering)
  44       : mSourceRate(aSourceRate),
  45         mTargetRate(aTargetRate),
  46         mDesiredBuffering(aDesiredBuffering) {}
  47
  48   /**
  49    * The correction in the form of a ratio. A correction of 0.98 means that the
  50    * target is 2% slower compared to the source or 1.03 which means that the
  51    * target is 3% faster than the source.
  52    */
  53   float GetCorrection() { return mCorrection; }
  54
  55   /**
  56    * Update the available source frames, target frames, and the current
  57    * buffer, in every iteration. If the conditions are met a new correction is
  58    * calculated. A new correction is calculated in the following cases:
  59    *   1. Every mAdjustmentIntervalMs milliseconds (1000ms).
  60    *   2. Every time we run low on buffered frames (less than 20ms).
  61    * In addition to that, the correction is clamped to 10% to avoid sound
  62    * distortion so the result will be in [0.9, 1.1].
  63    */
  64   void UpdateClock(uint32_t aSourceFrames, uint32_t aTargetFrames,
  65                    uint32_t aBufferedFrames, uint32_t aRemainingFrames) {
  66     if (mSourceClock >= mSourceRate / 10 || mTargetClock >= mTargetRate / 10) {
  67       // Only update the correction if 100ms has passed since last update.
  68       if (aBufferedFrames < mDesiredBuffering * 4 / 10 /*40%*/ ||
  69           aRemainingFrames < mDesiredBuffering * 4 / 10 /*40%*/) {
  70         // We are getting close to the lower or upper bound of the internal
  71         // buffer. Steer clear.
  72         CalculateCorrection(0.9, aBufferedFrames, aRemainingFrames);
  73       } else if ((mTargetClock * 1000 / mTargetRate) >= mAdjustmentIntervalMs ||
  74                  (mSourceClock * 1000 / mSourceRate) >= mAdjustmentIntervalMs) {
  75         // The adjustment interval has passed on one side. Recalculate.
  76         CalculateCorrection(0.6, aBufferedFrames, aRemainingFrames);
  77       }
  78     }
  79     mTargetClock += aTargetFrames;
  80     mSourceClock += aSourceFrames;
  81   }
  82
  83  private:
  84   /**
  85    * aCalculationWeight is a percentage [0, 1] with which the calculated
  86    * correction will be weighted. The existing correction will be weighted with
  87    * 1 - aCalculationWeight. This gives some inertia to the speed at which the
  88    * correction changes, for smoother changes.
  89    */
  90   void CalculateCorrection(float aCalculationWeight, uint32_t aBufferedFrames,
  91                            uint32_t aRemainingFrames) {
  92     // We want to maintain the desired buffer
  93     uint32_t bufferedFramesDiff = aBufferedFrames - mDesiredBuffering;
  94     uint32_t resampledSourceClock =
  95         std::max(1u, mSourceClock + bufferedFramesDiff);
  96     if (mTargetRate != mSourceRate) {
  97       resampledSourceClock *= static_cast<float>(mTargetRate) / mSourceRate;
  98     }
  99
 100     MOZ_LOG(gMediaTrackGraphLog, LogLevel::Verbose,
 101             ("ClockDrift %p Calculated correction %.3f (with weight: %.1f -> "
 102              "%.3f) (buffer: %u, desired: %u, remaining: %u)",
 103              this, static_cast<float>(mTargetClock) / resampledSourceClock,
 104              aCalculationWeight,
 105              (1 - aCalculationWeight) * mCorrection +
 106                  aCalculationWeight * mTargetClock / resampledSourceClock,
 107              aBufferedFrames, mDesiredBuffering, aRemainingFrames));
 108
 109     mCorrection = (1 - aCalculationWeight) * mCorrection +
 110                   aCalculationWeight * mTargetClock / resampledSourceClock;
 111
 112     // Clamp to range [0.9, 1.1] to avoid distortion
 113     mCorrection = std::min(std::max(mCorrection, 0.9f), 1.1f);
 114
 115     // Reset the counters to prepare for the next period.
 116     mTargetClock = 0;
 117     mSourceClock = 0;
 118   }
 119
 120  public:
 121   const uint32_t mSourceRate;
 122   const uint32_t mTargetRate;
 123   const uint32_t mAdjustmentIntervalMs = 1000;
 124   const uint32_t mDesiredBuffering;
 125
 126  private:
 127   float mCorrection = 1.0;
 128
 129   uint32_t mSourceClock = 0;
 130   uint32_t mTargetClock = 0;
 131 };
 132
 133 /**
 134  * Correct the drift between two independent clocks, the source, and the target
 135  * clock. The target clock is the master clock so the correction syncs the drift
 136  * of the source clock to the target. The nominal sampling rates of source and
 137  * target must be provided. If the source and the target operate in different
 138  * sample rate the drift correction will be performed on the top of resampling
 139  * from the source rate to the target rate.
 140  *
 141  * It works with AudioSegment in order to be able to be used from the
 142  * MediaTrackGraph/MediaTrack. The audio buffers are pre-allocated so there is
 143  * no new allocation takes place during operation. The preallocation capacity is
 144  * 100ms for input and 100ms for output. The class consists of ClockDrift and
 145  * AudioResampler check there for more details.
 146  *
 147  * The class is not thread-safe. The construction can happen in any thread but
 148  * the member method must be used in a single thread that can be different than
 149  * the construction thread. Appropriate for being used in the high priority
 150  * audio thread.
 151  */
 152 class AudioDriftCorrection final {
 153  public:
 154   AudioDriftCorrection(uint32_t aSourceRate, uint32_t aTargetRate)
 155       : mDesiredBuffering(
 156             std::max(5, Preferences::GetInt("media.clockdrift.buffering", 50)) *
 157             aSourceRate / 1000),
 158         mTargetRate(aTargetRate),
 159         mClockDrift(aSourceRate, aTargetRate, mDesiredBuffering),
 160         mResampler(aSourceRate, aTargetRate, mDesiredBuffering) {}
 161
 162   /**
 163    * The source audio frames and request the number of target audio frames must
 164    * be provided. The duration of the source and the output is considered as the
 165    * source clock and the target clock. The input is buffered internally so some
 166    * latency exists. The returned AudioSegment must be cleaned up because the
 167    * internal buffer will be reused after 100ms. If the drift correction (and
 168    * possible resampling) is not possible due to lack of input data an empty
 169    * AudioSegment will be returned. Not thread-safe.
 170    */
 171   AudioSegment RequestFrames(const AudioSegment& aInput,
 172                              uint32_t aOutputFrames) {
 173     // Very important to go first since the Dynamic will get the sample format
 174     // from the chunk.
 175     if (aInput.GetDuration()) {
 176       // Always go through the resampler because the clock might shift later.
 177       mResampler.AppendInput(aInput);
 178     }
 179     mClockDrift.UpdateClock(aInput.GetDuration(), aOutputFrames,
 180                             mResampler.InputReadableFrames(),
 181                             mResampler.InputWritableFrames());
 182     TrackRate receivingRate = mTargetRate * mClockDrift.GetCorrection();
 183     // Update resampler's rate if there is a new correction.
 184     mResampler.UpdateOutRate(receivingRate);
 185     // If it does not have enough frames the result will be an empty segment.
 186     AudioSegment output = mResampler.Resample(aOutputFrames);
 187     if (output.IsEmpty()) {
 188       NS_WARNING("Got nothing from the resampler");
 189       output.AppendNullData(aOutputFrames);
 190     }
 191     return output;
 192   }
 193
 194   // Only accessible from the same thread that is driving RequestFrames().
 195   uint32_t CurrentBuffering() const { return mResampler.InputReadableFrames(); }
 196
 197   const uint32_t mDesiredBuffering;
 198   const uint32_t mTargetRate;
 199
 200  private:
 201   ClockDrift mClockDrift;
 202   AudioResampler mResampler;
 203 };
 204
 205 };     // namespace mozilla
 206 #endif /* MOZILLA_AUDIO_DRIFT_CORRECTION_H_ */