Merging upstream version 127.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

12 files changed, 337 insertions, 297 deletions

diff --git a/dom/media/driftcontrol/AudioDriftCorrection.cpp b/dom/media/driftcontrol/AudioDriftCorrection.cpp
index e66c435c36..1b86a99a44 100644
--- a/dom/media/driftcontrol/AudioDriftCorrection.cpp
+++ b/dom/media/driftcontrol/AudioDriftCorrection.cpp
@@ -35,8 +35,8 @@ AudioDriftCorrection::AudioDriftCorrection(
     : mTargetRate(aTargetRate),
       mDriftController(MakeUnique<DriftController>(aSourceRate, aTargetRate,
                                                    mDesiredBuffering)),
-      mResampler(MakeUnique<AudioResampler>(
-          aSourceRate, aTargetRate, mDesiredBuffering, aPrincipalHandle)) {}
+      mResampler(MakeUnique<AudioResampler>(aSourceRate, aTargetRate, 0,
+                                            aPrincipalHandle)) {}
 
 AudioDriftCorrection::~AudioDriftCorrection() = default;
 
@@ -94,7 +94,7 @@ AudioSegment AudioDriftCorrection::RequestFrames(const AudioSegment& aInput,
   mDriftController->UpdateClock(inputDuration, outputDuration,
                                 CurrentBuffering(), BufferSize());
   // Update resampler's rate if there is a new correction.
-  mResampler->UpdateOutRate(mDriftController->GetCorrectedTargetRate());
+  mResampler->UpdateInRate(mDriftController->GetCorrectedSourceRate());
   if (hasUnderrun) {
     if (!mIsHandlingUnderrun) {
       NS_WARNING("Drift-correction: Underrun");
@@ -171,7 +171,8 @@ void AudioDriftCorrection::SetDesiredBuffering(
     media::TimeUnit aDesiredBuffering) {
   mDesiredBuffering = aDesiredBuffering;
   mDriftController->SetDesiredBuffering(mDesiredBuffering);
-  mResampler->SetPreBufferDuration(mDesiredBuffering);
+  mResampler->SetInputPreBufferFrameCount(
+      mDesiredBuffering.ToTicksAtRate(mDriftController->mSourceRate));
 }
 }  // namespace mozilla
 
diff --git a/dom/media/driftcontrol/AudioResampler.cpp b/dom/media/driftcontrol/AudioResampler.cpp
index ecef033a5c..1402fae39e 100644
--- a/dom/media/driftcontrol/AudioResampler.cpp
+++ b/dom/media/driftcontrol/AudioResampler.cpp
@@ -5,12 +5,14 @@
 
 #include "AudioResampler.h"
 
+#include "TimeUnits.h"
+
 namespace mozilla {
 
 AudioResampler::AudioResampler(uint32_t aInRate, uint32_t aOutRate,
-                               media::TimeUnit aPreBufferDuration,
+                               uint32_t aInputPreBufferFrameCount,
                                const PrincipalHandle& aPrincipalHandle)
-    : mResampler(aInRate, aOutRate, aPreBufferDuration),
+    : mResampler(aInRate, aOutRate, aInputPreBufferFrameCount),
       mOutputChunks(aOutRate / 10, STEREO, aPrincipalHandle) {}
 
 void AudioResampler::AppendInput(const AudioSegment& aInSegment) {
@@ -59,11 +61,11 @@ AudioSegment AudioResampler::Resample(uint32_t aOutFrames, bool* aHasUnderrun) {
     return segment;
   }
 
-  media::TimeUnit outDuration(aOutFrames, mResampler.GetOutRate());
+  media::TimeUnit outDuration(aOutFrames, mResampler.mOutRate);
   mResampler.EnsurePreBuffer(outDuration);
 
   const media::TimeUnit chunkCapacity(mOutputChunks.ChunkCapacity(),
-                                      mResampler.GetOutRate());
+                                      mResampler.mOutRate);
 
   while (!outDuration.IsZero()) {
     MOZ_ASSERT(outDuration.IsPositive());
@@ -71,8 +73,7 @@ AudioSegment AudioResampler::Resample(uint32_t aOutFrames, bool* aHasUnderrun) {
     const media::TimeUnit chunkDuration = std::min(outDuration, chunkCapacity);
     outDuration -= chunkDuration;
 
-    const uint32_t outFrames =
-        chunkDuration.ToTicksAtRate(mResampler.GetOutRate());
+    const uint32_t outFrames = chunkDuration.ToTicksAtRate(mResampler.mOutRate);
     for (uint32_t i = 0; i < chunk.ChannelCount(); ++i) {
       if (chunk.mBufferFormat == AUDIO_FORMAT_FLOAT32) {
         *aHasUnderrun |= mResampler.Resample(
@@ -92,8 +93,8 @@ AudioSegment AudioResampler::Resample(uint32_t aOutFrames, bool* aHasUnderrun) {
   return segment;
 }
 
-void AudioResampler::Update(uint32_t aOutRate, uint32_t aChannels) {
-  mResampler.UpdateResampler(aOutRate, aChannels);
+void AudioResampler::Update(uint32_t aInRate, uint32_t aChannels) {
+  mResampler.UpdateResampler(aInRate, aChannels);
   mOutputChunks.Update(aChannels);
 }
 
diff --git a/dom/media/driftcontrol/AudioResampler.h b/dom/media/driftcontrol/AudioResampler.h
index 20e4f1051b..c5982c3a39 100644
--- a/dom/media/driftcontrol/AudioResampler.h
+++ b/dom/media/driftcontrol/AudioResampler.h
@@ -9,12 +9,11 @@
 #include "AudioChunkList.h"
 #include "AudioSegment.h"
 #include "DynamicResampler.h"
-#include "TimeUnits.h"
 
 namespace mozilla {
 
 /**
- * Audio Resampler is a resampler able to change the output rate and channels
+ * Audio Resampler is a resampler able to change the input rate and channels
  * count on the fly. The API is simple and it is based in AudioSegment in order
  * to be used MTG. Memory allocations, for input and output buffers, will happen
  * in the constructor, when channel count changes and if the amount of input
@@ -36,7 +35,7 @@ namespace mozilla {
 class AudioResampler final {
  public:
   AudioResampler(uint32_t aInRate, uint32_t aOutRate,
-                 media::TimeUnit aPreBufferDuration,
+                 uint32_t aInputPreBufferFrameCount,
                  const PrincipalHandle& aPrincipalHandle);
 
   /**
@@ -69,24 +68,24 @@ class AudioResampler final {
   AudioSegment Resample(uint32_t aOutFrames, bool* aHasUnderrun);
 
   /*
-   * Updates the output rate that will be used by the resampler.
+   * Updates the input rate that will be used by the resampler.
    */
-  void UpdateOutRate(uint32_t aOutRate) {
-    Update(aOutRate, mResampler.GetChannels());
+  void UpdateInRate(uint32_t aInRate) {
+    Update(aInRate, mResampler.GetChannels());
   }
 
   /**
-   * Set the duration that should be used for pre-buffering.
+   * Set the number of frames that should be used for input pre-buffering.
    */
-  void SetPreBufferDuration(media::TimeUnit aPreBufferDuration) {
-    mResampler.SetPreBufferDuration(aPreBufferDuration);
+  void SetInputPreBufferFrameCount(uint32_t aInputPreBufferFrameCount) {
+    mResampler.SetInputPreBufferFrameCount(aInputPreBufferFrameCount);
   }
 
  private:
   void UpdateChannels(uint32_t aChannels) {
-    Update(mResampler.GetOutRate(), aChannels);
+    Update(mResampler.GetInRate(), aChannels);
   }
-  void Update(uint32_t aOutRate, uint32_t aChannels);
+  void Update(uint32_t aInRate, uint32_t aChannels);
 
  private:
   DynamicResampler mResampler;
diff --git a/dom/media/driftcontrol/DriftController.cpp b/dom/media/driftcontrol/DriftController.cpp
index b5603f72bb..791bfe9614 100644
--- a/dom/media/driftcontrol/DriftController.cpp
+++ b/dom/media/driftcontrol/DriftController.cpp
@@ -50,7 +50,7 @@ DriftController::DriftController(uint32_t aSourceRate, uint32_t aTargetRate,
       mSourceRate(aSourceRate),
       mTargetRate(aTargetRate),
       mDesiredBuffering(aDesiredBuffering),
-      mCorrectedTargetRate(static_cast<float>(aTargetRate)),
+      mCorrectedSourceRate(static_cast<float>(aSourceRate)),
       mMeasuredSourceLatency(5),
       mMeasuredTargetLatency(5) {
   LOG_CONTROLLER(
@@ -76,8 +76,8 @@ void DriftController::ResetAfterUnderrun() {
   mTargetClock = mAdjustmentInterval;
 }
 
-uint32_t DriftController::GetCorrectedTargetRate() const {
-  return std::lround(mCorrectedTargetRate);
+uint32_t DriftController::GetCorrectedSourceRate() const {
+  return std::lround(mCorrectedSourceRate);
 }
 
 void DriftController::UpdateClock(media::TimeUnit aSourceDuration,
@@ -112,14 +112,16 @@ void DriftController::CalculateCorrection(uint32_t aBufferedFrames,
   static constexpr float kDerivativeGain = 0.12;
 
   // Maximum 0.1% change per update.
-  const float cap = static_cast<float>(mTargetRate) / 1000.0f;
+  const float cap = static_cast<float>(mSourceRate) / 1000.0f;
 
   // The integral term can make us grow far outside the cap. Impose a cap on
   // it individually that is roughly equivalent to the final cap.
   const float integralCap = cap / kIntegralGain;
 
-  int32_t error = CheckedInt32(mDesiredBuffering.ToTicksAtRate(mSourceRate) -
-                               aBufferedFrames)
+  // Use nominal (not corrected) source rate when interpreting desired
+  // buffering so that the set point is independent of the control value.
+  int32_t error = CheckedInt32(aBufferedFrames -
+                               mDesiredBuffering.ToTicksAtRate(mSourceRate))
                       .value();
   int32_t proportional = error;
   // targetClockSec is the number of target clock seconds since last
@@ -135,12 +137,12 @@ void DriftController::CalculateCorrection(uint32_t aBufferedFrames,
                         kIntegralGain * mIntegral +
                         kDerivativeGain * derivative;
   float correctedRate =
-      std::clamp(static_cast<float>(mTargetRate) + controlSignal,
-                 mCorrectedTargetRate - cap, mCorrectedTargetRate + cap);
+      std::clamp(static_cast<float>(mSourceRate) + controlSignal,
+                 mCorrectedSourceRate - cap, mCorrectedSourceRate + cap);
 
   // mDesiredBuffering is divided by this to calculate the amount of
   // hysteresis to apply. With a denominator of 5, an error within +/- 20% of
-  // the desired buffering will not make corrections to the target sample
+  // the desired buffering will not make corrections to the source sample
   // rate.
   static constexpr uint32_t kHysteresisDenominator = 5;  // +/- 20%
 
@@ -183,7 +185,7 @@ void DriftController::CalculateCorrection(uint32_t aBufferedFrames,
       return correctedRate;
     }
 
-    return mCorrectedTargetRate;
+    return mCorrectedSourceRate;
   }();
 
   if (mDurationWithinHysteresis > mIntegralCapTimeLimit) {
@@ -201,10 +203,10 @@ void DriftController::CalculateCorrection(uint32_t aBufferedFrames,
   LOG_CONTROLLER(
       LogLevel::Verbose, this,
       "Recalculating Correction: Nominal: %uHz->%uHz, Corrected: "
-      "%uHz->%.2fHz  (diff %.2fHz), error: %.2fms (hysteresisThreshold: "
+      "%.2fHz->%uHz  (diff %.2fHz), error: %.2fms (hysteresisThreshold: "
       "%.2fms), buffering: %.2fms, desired buffering: %.2fms",
-      mSourceRate, mTargetRate, mSourceRate, hysteresisCorrectedRate,
-      hysteresisCorrectedRate - mCorrectedTargetRate,
+      mSourceRate, mTargetRate, hysteresisCorrectedRate, mTargetRate,
+      hysteresisCorrectedRate - mCorrectedSourceRate,
       media::TimeUnit(error, mSourceRate).ToSeconds() * 1000.0,
       media::TimeUnit(hysteresisThreshold, mSourceRate).ToSeconds() * 1000.0,
       media::TimeUnit(aBufferedFrames, mSourceRate).ToSeconds() * 1000.0,
@@ -219,13 +221,13 @@ void DriftController::CalculateCorrection(uint32_t aBufferedFrames,
                   kProportionalGain * proportional, kIntegralGain * mIntegral,
                   kDerivativeGain * derivative, controlSignal);
 
-  if (std::lround(mCorrectedTargetRate) !=
+  if (std::lround(mCorrectedSourceRate) !=
       std::lround(hysteresisCorrectedRate)) {
     ++mNumCorrectionChanges;
   }
 
   mPreviousError = error;
-  mCorrectedTargetRate = hysteresisCorrectedRate;
+  mCorrectedSourceRate = std::max(1.f, hysteresisCorrectedRate);
 
   // Reset the counters to prepare for the next period.
   mTargetClock = media::TimeUnit::Zero();
diff --git a/dom/media/driftcontrol/DriftController.h b/dom/media/driftcontrol/DriftController.h
index 0bd745c737..e8dbc57e0e 100644
--- a/dom/media/driftcontrol/DriftController.h
+++ b/dom/media/driftcontrol/DriftController.h
@@ -22,8 +22,8 @@ namespace mozilla {
  * the calculations.
  *
  * The DriftController looks at how the current buffering level differs from the
- * desired buffering level and sets a corrected target rate. A resampler should
- * be configured to resample from the nominal source rate to the corrected
+ * desired buffering level and sets a corrected source rate. A resampler should
+ * be configured to resample from the corrected source rate to the nominal
  * target rate. It assumes that the resampler is initially configured to
  * resample from the nominal source rate to the nominal target rate.
  *
@@ -53,12 +53,12 @@ class DriftController final {
   void ResetAfterUnderrun();
 
   /**
-   * Returns the drift-corrected target rate.
+   * Returns the drift-corrected source rate.
    */
-  uint32_t GetCorrectedTargetRate() const;
+  uint32_t GetCorrectedSourceRate() const;
 
   /**
-   * The number of times mCorrectedTargetRate has been changed to adjust to
+   * The number of times mCorrectedSourceRate has been changed to adjust to
    * drift.
    */
   uint32_t NumCorrectionChanges() const { return mNumCorrectionChanges; }
@@ -102,9 +102,12 @@ class DriftController final {
   // This implements a simple PID controller with feedback.
   // Set point:     SP = mDesiredBuffering.
   // Process value: PV(t) = aBufferedFrames. This is the feedback.
-  // Error:         e(t) = mDesiredBuffering - aBufferedFrames.
-  // Control value: CV(t) = the number to add to the nominal target rate, i.e.
-  //                the corrected target rate = CV(t) + nominal target rate.
+  // Error:         e(t) = aBufferedFrames - mDesiredBuffering.
+  //                Error is positive when the process value is high, which is
+  //                the opposite of conventional PID controllers because this
+  //                is a reverse-acting system.
+  // Control value: CV(t) = the value to add to the nominal source rate, i.e.
+  //                the corrected source rate = nominal source rate + CV(t).
   //
   // Controller:
   // Proportional part: The error, p(t) = e(t), multiplied by a gain factor, Kp.
@@ -115,13 +118,13 @@ class DriftController final {
   // Control signal:    The sum of the parts' output,
   //                    u(t) = Kp*p(t) + Ki*i(t) + Kd*d(t).
   //
-  // Control action: Converting the control signal to a target sample rate.
+  // Control action: Converting the control signal to a source sample rate.
   //                 Simplified, a positive control signal means the buffer is
-  //                 lower than desired (because the error is positive), so the
-  //                 target sample rate must be increased in order to consume
-  //                 input data slower. We calculate the corrected target rate
-  //                 by simply adding the control signal, u(t), to the nominal
-  //                 target rate.
+  //                 higher than desired (because the error is positive),
+  //                 so the source sample rate must be increased in order to
+  //                 consume input data faster.
+  //                 We calculate the corrected source rate by simply adding
+  //                 the control signal, u(t), to the nominal source rate.
   //
   // Hysteresis: As long as the error is within a threshold of 20% of the set
   //             point (desired buffering level) (up to 10ms for >50ms desired
@@ -144,7 +147,7 @@ class DriftController final {
   int32_t mPreviousError = 0;
   float mIntegral = 0.0;
   Maybe<float> mIntegralCenterForCap;
-  float mCorrectedTargetRate;
+  float mCorrectedSourceRate;
   Maybe<int32_t> mLastHysteresisBoundaryCorrection;
   media::TimeUnit mDurationWithinHysteresis;
   uint32_t mNumCorrectionChanges = 0;
diff --git a/dom/media/driftcontrol/DynamicResampler.cpp b/dom/media/driftcontrol/DynamicResampler.cpp
index e6f230278e..65a2ae3b57 100644
--- a/dom/media/driftcontrol/DynamicResampler.cpp
+++ b/dom/media/driftcontrol/DynamicResampler.cpp
@@ -8,14 +8,13 @@
 namespace mozilla {
 
 DynamicResampler::DynamicResampler(uint32_t aInRate, uint32_t aOutRate,
-                                   media::TimeUnit aPreBufferDuration)
-    : mInRate(aInRate),
-      mPreBufferDuration(aPreBufferDuration),
-      mOutRate(aOutRate) {
+                                   uint32_t aInputPreBufferFrameCount)
+    : mOutRate(aOutRate),
+      mInputPreBufferFrameCount(aInputPreBufferFrameCount),
+      mInRate(aInRate) {
   MOZ_ASSERT(aInRate);
   MOZ_ASSERT(aOutRate);
-  MOZ_ASSERT(aPreBufferDuration.IsPositiveOrZero());
-  UpdateResampler(mOutRate, STEREO);
+  UpdateResampler(mInRate, STEREO);
   mInputStreamFile.Open("DynamicResamplerInFirstChannel", 1, mInRate);
   mOutputStreamFile.Open("DynamicResamplerOutFirstChannel", 1, mOutRate);
 }
@@ -35,7 +34,10 @@ void DynamicResampler::SetSampleFormat(AudioSampleFormat aFormat) {
     b.SetSampleFormat(mSampleFormat);
   }
 
-  EnsureInputBufferDuration(CalculateInputBufferDuration());
+  // Pre-allocate something big.
+  // EnsureInputBufferDuration() adds 50ms for jitter to this first allocation
+  // so the 50ms argument means at least 100ms.
+  EnsureInputBufferSizeInFrames(mInRate / 20);
 }
 
 void DynamicResampler::EnsurePreBuffer(media::TimeUnit aDuration) {
@@ -43,33 +45,36 @@ void DynamicResampler::EnsurePreBuffer(media::TimeUnit aDuration) {
     return;
   }
 
-  media::TimeUnit buffered(mInternalInBuffer[0].AvailableRead(), mInRate);
-  if (buffered.IsZero()) {
+  uint32_t buffered = mInternalInBuffer[0].AvailableRead();
+  if (buffered == 0) {
     // Wait for the first input segment before deciding how much to pre-buffer.
     // If it is large it indicates high-latency, and the buffer would have to
-    // handle that.
+    // handle that.  This also means that the pre-buffer is not set up just
+    // before a large input segment would extend the buffering beyond the
+    // desired level.
     return;
   }
 
   mIsPreBufferSet = true;
 
-  media::TimeUnit needed = aDuration + mPreBufferDuration;
-  EnsureInputBufferDuration(needed);
+  uint32_t needed =
+      aDuration.ToTicksAtRate(mInRate) + mInputPreBufferFrameCount;
+  EnsureInputBufferSizeInFrames(needed);
 
   if (needed > buffered) {
     for (auto& b : mInternalInBuffer) {
-      b.PrependSilence((needed - buffered).ToTicksAtRate(mInRate));
+      b.PrependSilence(needed - buffered);
     }
   } else if (needed < buffered) {
     for (auto& b : mInternalInBuffer) {
-      b.Discard((buffered - needed).ToTicksAtRate(mInRate));
+      b.Discard(buffered - needed);
     }
   }
 }
 
-void DynamicResampler::SetPreBufferDuration(media::TimeUnit aDuration) {
-  MOZ_ASSERT(aDuration.IsPositive());
-  mPreBufferDuration = aDuration;
+void DynamicResampler::SetInputPreBufferFrameCount(
+    uint32_t aInputPreBufferFrameCount) {
+  mInputPreBufferFrameCount = aInputPreBufferFrameCount;
 }
 
 bool DynamicResampler::Resample(float* aOutBuffer, uint32_t aOutFrames,
@@ -93,7 +98,6 @@ void DynamicResampler::ResampleInternal(const float* aInBuffer,
   MOZ_ASSERT(mInRate);
   MOZ_ASSERT(mOutRate);
 
-  MOZ_ASSERT(aInBuffer);
   MOZ_ASSERT(aInFrames);
   MOZ_ASSERT(*aInFrames > 0);
   MOZ_ASSERT(aOutBuffer);
@@ -125,7 +129,6 @@ void DynamicResampler::ResampleInternal(const int16_t* aInBuffer,
   MOZ_ASSERT(mInRate);
   MOZ_ASSERT(mOutRate);
 
-  MOZ_ASSERT(aInBuffer);
   MOZ_ASSERT(aInFrames);
   MOZ_ASSERT(*aInFrames > 0);
   MOZ_ASSERT(aOutBuffer);
@@ -147,19 +150,20 @@ void DynamicResampler::ResampleInternal(const int16_t* aInBuffer,
   }
 }
 
-void DynamicResampler::UpdateResampler(uint32_t aOutRate, uint32_t aChannels) {
-  MOZ_ASSERT(aOutRate);
+void DynamicResampler::UpdateResampler(uint32_t aInRate, uint32_t aChannels) {
+  MOZ_ASSERT(aInRate);
   MOZ_ASSERT(aChannels);
 
   if (mChannels != aChannels) {
+    uint32_t bufferSizeInFrames = InFramesBufferSize();
     if (mResampler) {
       speex_resampler_destroy(mResampler);
     }
-    mResampler = speex_resampler_init(aChannels, mInRate, aOutRate,
+    mResampler = speex_resampler_init(aChannels, aInRate, mOutRate,
                                       SPEEX_RESAMPLER_QUALITY_MIN, nullptr);
     MOZ_ASSERT(mResampler);
     mChannels = aChannels;
-    mOutRate = aOutRate;
+    mInRate = aInRate;
     // Between mono and stereo changes, keep always allocated 2 channels to
     // avoid reallocations in the most common case.
     if ((mChannels == STEREO || mChannels == 1) &&
@@ -192,14 +196,12 @@ void DynamicResampler::UpdateResampler(uint32_t aOutRate, uint32_t aChannels) {
         b->SetSampleFormat(mSampleFormat);
       }
     }
-    media::TimeUnit d = mSetBufferDuration;
-    mSetBufferDuration = media::TimeUnit::Zero();
-    EnsureInputBufferDuration(d);
+    EnsureInputBufferSizeInFrames(bufferSizeInFrames);
     mInputTail.SetLength(mChannels);
     return;
   }
 
-  if (mOutRate != aOutRate) {
+  if (mInRate != aInRate) {
     // If the rates was the same the resampler was not being used so warm up.
     if (mOutRate == mInRate) {
       WarmUpResampler(true);
@@ -208,9 +210,9 @@ void DynamicResampler::UpdateResampler(uint32_t aOutRate, uint32_t aChannels) {
 #ifdef DEBUG
     int rv =
 #endif
-        speex_resampler_set_rate(mResampler, mInRate, aOutRate);
+        speex_resampler_set_rate(mResampler, aInRate, mOutRate);
     MOZ_ASSERT(rv == RESAMPLER_ERR_SUCCESS);
-    mOutRate = aOutRate;
+    mInRate = aInRate;
   }
 }
 
@@ -236,13 +238,9 @@ void DynamicResampler::WarmUpResampler(bool aSkipLatency) {
     }
   }
   if (aSkipLatency) {
-    int inputLatency = speex_resampler_get_input_latency(mResampler);
-    MOZ_ASSERT(inputLatency > 0);
-    uint32_t ratioNum, ratioDen;
-    speex_resampler_get_ratio(mResampler, &ratioNum, &ratioDen);
-    // Ratio at this point is one so only skip the input latency. No special
-    // calculations are needed.
-    speex_resampler_set_skip_frac_num(mResampler, inputLatency * ratioDen);
+    // Don't generate output frames corresponding to times before the next
+    // input sample.
+    speex_resampler_skip_zeros(mResampler);
   }
   mIsWarmingUp = false;
 }
@@ -268,7 +266,16 @@ void DynamicResampler::AppendInputSilence(const uint32_t aInFrames) {
 }
 
 uint32_t DynamicResampler::InFramesBufferSize() const {
-  return mSetBufferDuration.ToTicksAtRate(mInRate);
+  if (mSampleFormat == AUDIO_FORMAT_SILENCE) {
+    return 0;
+  }
+  // Buffers may have different capacities if a memory allocation has failed.
+  MOZ_ASSERT(!mInternalInBuffer.IsEmpty());
+  uint32_t min = std::numeric_limits<uint32_t>::max();
+  for (const auto& b : mInternalInBuffer) {
+    min = std::min(min, b.Capacity());
+  }
+  return min;
 }
 
 uint32_t DynamicResampler::InFramesBuffered(uint32_t aChannelIndex) const {
@@ -276,7 +283,7 @@ uint32_t DynamicResampler::InFramesBuffered(uint32_t aChannelIndex) const {
   MOZ_ASSERT(aChannelIndex <= mChannels);
   MOZ_ASSERT(aChannelIndex <= mInternalInBuffer.Length());
   if (!mIsPreBufferSet) {
-    return mPreBufferDuration.ToTicksAtRate(mInRate);
+    return mInputPreBufferFrameCount;
   }
   return mInternalInBuffer[aChannelIndex].AvailableRead();
 }
diff --git a/dom/media/driftcontrol/DynamicResampler.h b/dom/media/driftcontrol/DynamicResampler.h
index c1b9000aa0..1f601c898b 100644
--- a/dom/media/driftcontrol/DynamicResampler.h
+++ b/dom/media/driftcontrol/DynamicResampler.h
@@ -27,15 +27,13 @@ const uint32_t STEREO = 2;
  * to allow the requested to be resampled and returned.
  *
  * Input data buffering makes use of the AudioRingBuffer. The capacity of the
- * buffer is initially 100ms of float audio and it is pre-allocated at the
- * constructor. Should the input data grow beyond that, the input buffer is
- * re-allocated on the fly. In addition to that, due to special feature of
+ * buffer is initially 100ms of audio and it is pre-allocated during
+ * SetSampleFormat(). Should the input data grow beyond that, the input buffer
+ * is re-allocated on the fly. In addition to that, due to special feature of
  * AudioRingBuffer, no extra copies take place when the input data is fed to the
  * resampler.
  *
- * The sample format must be set before using any method. If the provided sample
- * format is of type short the pre-allocated capacity of the input buffer
- * becomes 200ms of short audio.
+ * The sample format must be set before using any method.
  *
  * The DynamicResampler is not thread-safe, so all the methods appart from the
  * constructor must be called on the same thread.
@@ -47,16 +45,15 @@ class DynamicResampler final {
    * The channel count will be set to stereo. Memory allocation will take
    * place. The input buffer is non-interleaved.
    */
-  DynamicResampler(
-      uint32_t aInRate, uint32_t aOutRate,
-      media::TimeUnit aPreBufferDuration = media::TimeUnit::Zero());
+  DynamicResampler(uint32_t aInRate, uint32_t aOutRate,
+                   uint32_t aInputPreBufferFrameCount = 0);
   ~DynamicResampler();
 
   /**
    * Set the sample format type to float or short.
    */
   void SetSampleFormat(AudioSampleFormat aFormat);
-  uint32_t GetOutRate() const { return mOutRate; }
+  uint32_t GetInRate() const { return mInRate; }
   uint32_t GetChannels() const { return mChannels; }
 
   /**
@@ -81,16 +78,16 @@ class DynamicResampler final {
 
   /**
    * Prepends existing input data with a silent pre-buffer if not already done.
-   * Data will be prepended so that after resampling aOutFrames worth of output
-   * data, the buffering level will be as close as possible to
-   * mPreBufferDuration, which is the desired buffering level.
+   * Data will be prepended so that after resampling aDuration of data,
+   * the buffering level will be as close as possible to
+   * mInputPreBufferFrameCount, which is the desired buffering level.
    */
   void EnsurePreBuffer(media::TimeUnit aDuration);
 
   /**
-   * Set the duration that should be used for pre-buffering.
+   * Set the number of frames that should be used for input pre-buffering.
    */
-  void SetPreBufferDuration(media::TimeUnit aDuration);
+  void SetInputPreBufferFrameCount(uint32_t aInputPreBufferFrameCount);
 
   /*
    * Resample as much frames as needed from the internal input buffer to the
@@ -114,14 +111,14 @@ class DynamicResampler final {
 
   /**
    * Update the output rate or/and the channel count. If a value is not updated
-   * compared to the current one nothing happens. Changing the `aOutRate`
+   * compared to the current one nothing happens. Changing the `aInRate`
    * results in recalculation in the resampler. Changing `aChannels` results in
    * the reallocation of the internal input buffer with the exception of
    * changes between mono to stereo and vice versa where no reallocation takes
    * place. A stereo internal input buffer is always maintained even if the
    * sound is mono.
    */
-  void UpdateResampler(uint32_t aOutRate, uint32_t aChannels);
+  void UpdateResampler(uint32_t aInRate, uint32_t aChannels);
 
  private:
   template <typename T>
@@ -174,24 +171,24 @@ class DynamicResampler final {
     }
 
     uint32_t totalOutFramesNeeded = aOutFrames;
-    auto resample = [&] {
-      mInternalInBuffer[aChannelIndex].ReadNoCopy(
-          [&](const Span<const T>& aInBuffer) -> uint32_t {
-            if (!totalOutFramesNeeded) {
-              return 0;
-            }
-            uint32_t outFramesResampled = totalOutFramesNeeded;
-            uint32_t inFrames = aInBuffer.Length();
-            ResampleInternal(aInBuffer.data(), &inFrames, aOutBuffer,
-                             &outFramesResampled, aChannelIndex);
-            aOutBuffer += outFramesResampled;
-            totalOutFramesNeeded -= outFramesResampled;
-            mInputTail[aChannelIndex].StoreTail<T>(aInBuffer.To(inFrames));
-            return inFrames;
-          });
+    auto resample = [&](const T* aInBuffer, uint32_t aInLength) -> uint32_t {
+      uint32_t outFramesResampled = totalOutFramesNeeded;
+      uint32_t inFrames = aInLength;
+      ResampleInternal(aInBuffer, &inFrames, aOutBuffer, &outFramesResampled,
+                       aChannelIndex);
+      aOutBuffer += outFramesResampled;
+      totalOutFramesNeeded -= outFramesResampled;
+      mInputTail[aChannelIndex].StoreTail<T>(aInBuffer, inFrames);
+      return inFrames;
     };
 
-    resample();
+    mInternalInBuffer[aChannelIndex].ReadNoCopy(
+        [&](const Span<const T>& aInBuffer) -> uint32_t {
+          if (!totalOutFramesNeeded) {
+            return 0;
+          }
+          return resample(aInBuffer.Elements(), aInBuffer.Length());
+        });
 
     if (totalOutFramesNeeded == 0) {
       return false;
@@ -204,8 +201,7 @@ class DynamicResampler final {
           ((CheckedUint32(totalOutFramesNeeded) * mInRate + mOutRate - 1) /
            mOutRate)
               .value();
-      mInternalInBuffer[aChannelIndex].WriteSilence(totalInFramesNeeded);
-      resample();
+      resample(nullptr, totalInFramesNeeded);
     }
     mIsPreBufferSet = false;
     return true;
@@ -219,33 +215,14 @@ class DynamicResampler final {
     MOZ_ASSERT(mChannels);
     MOZ_ASSERT(aChannelIndex < mChannels);
     MOZ_ASSERT(aChannelIndex < mInternalInBuffer.Length());
-    EnsureInputBufferDuration(media::TimeUnit(
-        CheckedInt64(mInternalInBuffer[aChannelIndex].AvailableRead()) +
-            aInFrames,
-        mInRate));
+    EnsureInputBufferSizeInFrames(
+        mInternalInBuffer[aChannelIndex].AvailableRead() + aInFrames);
     mInternalInBuffer[aChannelIndex].Write(Span(aInBuffer, aInFrames));
   }
 
   void WarmUpResampler(bool aSkipLatency);
 
-  media::TimeUnit CalculateInputBufferDuration() const {
-    // Pre-allocate something big, twice the pre-buffer, or at least 100ms.
-    return std::max(mPreBufferDuration * 2, media::TimeUnit::FromSeconds(0.1));
-  }
-
-  bool EnsureInputBufferDuration(media::TimeUnit aDuration) {
-    if (aDuration <= mSetBufferDuration) {
-      // Buffer size is sufficient.
-      return true;
-    }
-
-    // 5 second cap.
-    const media::TimeUnit cap = media::TimeUnit::FromSeconds(5);
-    if (mSetBufferDuration == cap) {
-      // Already at the cap.
-      return false;
-    }
-
+  bool EnsureInputBufferSizeInFrames(uint32_t aSizeInFrames) {
     uint32_t sampleSize = 0;
     if (mSampleFormat == AUDIO_FORMAT_FLOAT32) {
       sampleSize = sizeof(float);
@@ -258,53 +235,62 @@ class DynamicResampler final {
       return true;
     }
 
+    uint32_t sizeInFrames = InFramesBufferSize();
+    if (aSizeInFrames <= sizeInFrames) {
+      // Buffer size is sufficient.
+      return true;  // no reallocation necessary
+    }
+
+    // 5 second cap.
+    const uint32_t cap = 5 * mInRate;
+    if (sizeInFrames >= cap) {
+      // Already at the cap.
+      return false;
+    }
+
     // As a backoff strategy, at least double the previous size.
-    media::TimeUnit duration = mSetBufferDuration * 2;
+    sizeInFrames *= 2;
 
-    if (aDuration > duration) {
+    if (aSizeInFrames > sizeInFrames) {
       // A larger buffer than the normal backoff strategy provides is needed, or
-      // this is the first time setting the buffer size. Round up to the nearest
-      // 100ms, some jitter is expected.
-      duration = aDuration.ToBase<media::TimeUnit::CeilingPolicy>(10);
+      // this is the first time setting the buffer size. Add another 50ms, as
+      // some jitter is expected.
+      sizeInFrames = aSizeInFrames + mInRate / 20;
     }
 
-    duration = std::min(cap, duration);
+    // mInputPreBufferFrameCount is an indication of the desired average
+    // buffering.  Provide for at least twice this.
+    sizeInFrames = std::max(sizeInFrames, mInputPreBufferFrameCount * 2);
+
+    sizeInFrames = std::min(cap, sizeInFrames);
 
     bool success = true;
     for (auto& b : mInternalInBuffer) {
-      success = success &&
-                b.SetLengthBytes(sampleSize * duration.ToTicksAtRate(mInRate));
+      success = success && b.EnsureLengthBytes(sampleSize * sizeInFrames);
     }
 
     if (success) {
       // All buffers have the new size.
-      mSetBufferDuration = duration;
       return true;
     }
 
-    const uint32_t sizeInFrames =
-        static_cast<uint32_t>(mSetBufferDuration.ToTicksAtRate(mInRate));
     // Allocating an input buffer failed. We stick with the old buffer size.
     NS_WARNING(nsPrintfCString("Failed to allocate a buffer of %u bytes (%u "
                                "frames). Expect glitches.",
                                sampleSize * sizeInFrames, sizeInFrames)
                    .get());
-    for (auto& b : mInternalInBuffer) {
-      MOZ_ALWAYS_TRUE(b.SetLengthBytes(sampleSize * sizeInFrames));
-    }
     return false;
   }
 
  public:
-  const uint32_t mInRate;
+  const uint32_t mOutRate;
 
  private:
   bool mIsPreBufferSet = false;
   bool mIsWarmingUp = false;
-  media::TimeUnit mPreBufferDuration;
-  media::TimeUnit mSetBufferDuration = media::TimeUnit::Zero();
+  uint32_t mInputPreBufferFrameCount;
   uint32_t mChannels = 0;
-  uint32_t mOutRate;
+  uint32_t mInRate;
 
   AutoTArray<AudioRingBuffer, STEREO> mInternalInBuffer;
 
@@ -324,16 +310,16 @@ class DynamicResampler final {
     }
     template <typename T>
     void StoreTail(const T* aInBuffer, uint32_t aInFrames) {
-      if (aInFrames >= MAXSIZE) {
-        PodCopy(Buffer<T>(), aInBuffer + aInFrames - MAXSIZE, MAXSIZE);
-        mSize = MAXSIZE;
+      const T* inBuffer = aInBuffer;
+      mSize = std::min(aInFrames, MAXSIZE);
+      if (inBuffer) {
+        PodCopy(Buffer<T>(), inBuffer + aInFrames - mSize, mSize);
       } else {
-        PodCopy(Buffer<T>(), aInBuffer, aInFrames);
-        mSize = aInFrames;
+        std::fill_n(Buffer<T>(), mSize, static_cast<T>(0));
       }
     }
     uint32_t Length() { return mSize; }
-    static const uint32_t MAXSIZE = 20;
+    static constexpr uint32_t MAXSIZE = 20;
 
    private:
     float mBuffer[MAXSIZE] = {};
diff --git a/dom/media/driftcontrol/gtest/TestAudioDriftCorrection.cpp b/dom/media/driftcontrol/gtest/TestAudioDriftCorrection.cpp
index c13f443d37..9d3f0f091a 100644
--- a/dom/media/driftcontrol/gtest/TestAudioDriftCorrection.cpp
+++ b/dom/media/driftcontrol/gtest/TestAudioDriftCorrection.cpp
@@ -260,10 +260,10 @@ TEST(TestAudioDriftCorrection, LargerTransmitterBlockSizeThanDesiredBuffering)
 
   // Input is stable so no corrections should occur.
   EXPECT_EQ(ad.NumCorrectionChanges(), 0U);
-  // The drift correction buffer size had to be larger than the desired (the
-  // buffer size is twice the initial buffering level), to accomodate the large
-  // input block size.
-  EXPECT_EQ(ad.BufferSize(), 9600U);
+  // The desired buffering and pre-buffering level was
+  // transmitterBlockSize * 11 / 10 to accomodate the large input block size.
+  // The buffer size was twice the pre-buffering level.
+  EXPECT_EQ(ad.BufferSize(), transmitterBlockSize * 11 / 10 * 2);
 }
 
 TEST(TestAudioDriftCorrection, LargerReceiverBlockSizeThanDesiredBuffering)
@@ -275,9 +275,9 @@ TEST(TestAudioDriftCorrection, LargerReceiverBlockSizeThanDesiredBuffering)
       MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
   AudioDriftCorrection ad(sampleRate, sampleRate, testPrincipal);
 
+  AudioSegment inSegment;
   for (uint32_t i = 0; i < (sampleRate / 1000) * 500;
        i += transmitterBlockSize) {
-    AudioSegment inSegment;
     AudioChunk chunk =
         CreateAudioChunk<float>(transmitterBlockSize, 1, AUDIO_FORMAT_FLOAT32);
     inSegment.AppendAndConsumeChunk(std::move(chunk));
@@ -285,6 +285,7 @@ TEST(TestAudioDriftCorrection, LargerReceiverBlockSizeThanDesiredBuffering)
     if (i % receiverBlockSize == 0) {
       AudioSegment outSegment = ad.RequestFrames(inSegment, receiverBlockSize);
       EXPECT_EQ(outSegment.GetDuration(), receiverBlockSize);
+      inSegment.Clear();
     }
 
     if (i >= receiverBlockSize) {
@@ -294,11 +295,12 @@ TEST(TestAudioDriftCorrection, LargerReceiverBlockSizeThanDesiredBuffering)
 
   // Input is stable so no corrections should occur.
   EXPECT_EQ(ad.NumCorrectionChanges(), 0U);
+  EXPECT_EQ(ad.NumUnderruns(), 0U);
   // The drift correction buffer size had to be larger than the desired (the
   // buffer size is twice the initial buffering level), to accomodate the large
   // input block size that gets buffered in the resampler only when processing
   // output.
-  EXPECT_EQ(ad.BufferSize(), 19200U);
+  EXPECT_EQ(ad.BufferSize(), 9600U);
 }
 
 TEST(TestAudioDriftCorrection, DynamicInputBufferSizeChanges)
@@ -329,9 +331,9 @@ TEST(TestAudioDriftCorrection, DynamicInputBufferSizeChanges)
       if (((receivedFramesStart - transmittedFramesStart + i) /
            aTransmitterBlockSize) > numBlocksTransmitted) {
         tone.Generate(inSegment, aTransmitterBlockSize);
-        MOZ_ASSERT(!inSegment.IsNull());
+        MOZ_RELEASE_ASSERT(!inSegment.IsNull());
         inToneVerifier.AppendData(inSegment);
-        MOZ_ASSERT(!inSegment.IsNull());
+        MOZ_RELEASE_ASSERT(!inSegment.IsNull());
         ++numBlocksTransmitted;
         totalFramesTransmitted += aTransmitterBlockSize;
       }
@@ -459,29 +461,50 @@ TEST(TestAudioDriftCorrection, DriftStepResponseUnderrunHighLatencyInput)
   constexpr uint32_t iterations = 200;
   const PrincipalHandle testPrincipal =
       MakePrincipalHandle(nsContentUtils::GetSystemPrincipal());
-  uint32_t inputRate = nominalRate * 1005 / 1000;  // 0.5% drift
-  uint32_t inputInterval = inputRate;
+  uint32_t inputRate1 = nominalRate * 1005 / 1000;  // 0.5% drift
+  uint32_t inputInterval1 = inputRate1;
   AudioGenerator<AudioDataValue> tone(1, nominalRate, 440);
   AudioDriftCorrection ad(nominalRate, nominalRate, testPrincipal);
   for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
     AudioSegment inSegment;
     if (i > 0 && i % interval == 0) {
-      tone.Generate(inSegment, inputInterval);
+      tone.Generate(inSegment, inputInterval1);
     }
     ad.RequestFrames(inSegment, interval / 100);
   }
 
-  inputRate = nominalRate * 995 / 1000;  // -0.5% drift
-  inputInterval = inputRate;
+  uint32_t inputRate2 = nominalRate * 995 / 1000;  // -0.5% drift
+  uint32_t inputInterval2 = inputRate2;
   for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
     AudioSegment inSegment;
+    // The first segment is skipped to cause an underrun.
     if (i > 0 && i % interval == 0) {
-      tone.Generate(inSegment, inputInterval);
+      tone.Generate(inSegment, inputInterval2);
     }
     ad.RequestFrames(inSegment, interval / 100);
+    if (i >= interval / 10 && i < interval) {
+      // While the DynamicResampler has not set its pre-buffer after the
+      // underrun, InFramesBuffered() reports the pre-buffer size.
+      // The initial desired buffer and pre-buffer size was
+      // inputInterval1 * 11 / 10 to accomodate the large input block size.
+      // This was doubled when the underrun occurred.
+      EXPECT_EQ(ad.CurrentBuffering(), inputInterval1 * 11 / 10 * 2)
+          << "for i=" << i;
+    } else if (i == interval) {
+      // After the pre-buffer was set and used to generate the first output
+      // block, the actual number of frames buffered almost matches the
+      // pre-buffer size, with some rounding from output to input frame count
+      // conversion.
+      EXPECT_EQ(ad.CurrentBuffering(), inputInterval1 * 11 / 10 * 2 - 1)
+          << "after first input after underrun";
+    }
   }
 
-  EXPECT_EQ(ad.BufferSize(), 220800U);
+  // The initial desired buffering and pre-buffering level was
+  // inputInterval1 * 11 / 10 to accomodate the large input block size.
+  // The buffer size was initially twice the pre-buffering level, and then
+  // doubled when the underrun occurred.
+  EXPECT_EQ(ad.BufferSize(), inputInterval1 * 11 / 10 * 2 * 2);
   EXPECT_EQ(ad.NumUnderruns(), 1u);
 }
 
@@ -511,7 +534,7 @@ TEST(TestAudioDriftCorrection, DriftStepResponseOverrun)
     ad.RequestFrames(inSegment, interval / 100);
   }
 
-  // Change input callbacks to 2000ms (+0.5% drift) = 48200 frames, which will
+  // Change input callbacks to 1000ms (+0.5% drift) = 48200 frames, which will
   // overrun the ring buffer.
   for (uint32_t i = 0; i < interval * iterations; i += interval / 100) {
     AudioSegment inSegment;
@@ -524,6 +547,9 @@ TEST(TestAudioDriftCorrection, DriftStepResponseOverrun)
     ad.RequestFrames(inSegment, interval / 100);
   }
 
-  EXPECT_EQ(ad.BufferSize(), 105600U);
+  // The desired buffering and pre-buffering levels were increased to
+  // inputInterval * 11 / 10 to accomodate the large input block size.
+  // The buffer size was increased to twice the pre-buffering level.
+  EXPECT_EQ(ad.BufferSize(), inputInterval * 11 / 10 * 2);
   EXPECT_EQ(ad.NumUnderruns(), 1u);
 }
diff --git a/dom/media/driftcontrol/gtest/TestAudioResampler.cpp b/dom/media/driftcontrol/gtest/TestAudioResampler.cpp
index f04bc87314..7122b60a1a 100644
--- a/dom/media/driftcontrol/gtest/TestAudioResampler.cpp
+++ b/dom/media/driftcontrol/gtest/TestAudioResampler.cpp
@@ -64,8 +64,7 @@ TEST(TestAudioResampler, OutAudioSegment_Float)
 
   uint32_t pre_buffer = 21;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioSegment inSegment =
       CreateAudioSegment<float>(in_frames, channels, AUDIO_FORMAT_FLOAT32);
@@ -91,9 +90,9 @@ TEST(TestAudioResampler, OutAudioSegment_Float)
     }
   }
 
-  // Update out rate
-  out_rate = 44100;
-  dr.UpdateOutRate(out_rate);
+  // Update in rate
+  in_rate = 26122;
+  dr.UpdateInRate(in_rate);
   out_frames = in_frames * out_rate / in_rate;
   EXPECT_EQ(out_frames, 18u);
   // Even if we provide no input if we have enough buffered input, we can create
@@ -121,8 +120,7 @@ TEST(TestAudioResampler, OutAudioSegment_Short)
 
   uint32_t pre_buffer = 21;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioSegment inSegment =
       CreateAudioSegment<short>(in_frames, channels, AUDIO_FORMAT_S16);
@@ -148,9 +146,9 @@ TEST(TestAudioResampler, OutAudioSegment_Short)
     }
   }
 
-  // Update out rate
-  out_rate = 44100;
-  dr.UpdateOutRate(out_rate);
+  // Update in rate
+  in_rate = 26122;
+  dr.UpdateInRate(out_rate);
   out_frames = in_frames * out_rate / in_rate;
   EXPECT_EQ(out_frames, 18u);
   // Even if we provide no input if we have enough buffered input, we can create
@@ -175,8 +173,7 @@ TEST(TestAudioResampler, OutAudioSegmentLargerThanResampledInput_Float)
 
   uint32_t pre_buffer = 5;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    PRINCIPAL_HANDLE_NONE);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, PRINCIPAL_HANDLE_NONE);
 
   AudioSegment inSegment =
       CreateAudioSegment<float>(in_frames, channels, AUDIO_FORMAT_FLOAT32);
@@ -209,8 +206,7 @@ TEST(TestAudioResampler, InAudioSegment_Float)
   uint32_t out_rate = 48000;
 
   uint32_t pre_buffer = 10;
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioSegment inSegment;
 
@@ -275,8 +271,7 @@ TEST(TestAudioResampler, InAudioSegment_Short)
   uint32_t out_rate = 48000;
 
   uint32_t pre_buffer = 10;
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioSegment inSegment;
 
@@ -342,8 +337,7 @@ TEST(TestAudioResampler, ChannelChange_MonoToStereo)
 
   uint32_t pre_buffer = 0;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioChunk monoChunk =
       CreateAudioChunk<float>(in_frames, 1, AUDIO_FORMAT_FLOAT32);
@@ -378,8 +372,7 @@ TEST(TestAudioResampler, ChannelChange_StereoToMono)
 
   uint32_t pre_buffer = 0;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioChunk monoChunk =
       CreateAudioChunk<float>(in_frames, 1, AUDIO_FORMAT_FLOAT32);
@@ -414,8 +407,7 @@ TEST(TestAudioResampler, ChannelChange_StereoToQuad)
 
   uint32_t pre_buffer = 0;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, pre_buffer, testPrincipal);
 
   AudioChunk stereoChunk =
       CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32);
@@ -452,7 +444,7 @@ TEST(TestAudioResampler, ChannelChange_QuadToStereo)
   uint32_t in_rate = 24000;
   uint32_t out_rate = 48000;
 
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit::Zero(), testPrincipal);
+  AudioResampler dr(in_rate, out_rate, 0, testPrincipal);
 
   AudioChunk stereoChunk =
       CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32);
@@ -497,7 +489,7 @@ TEST(TestAudioResampler, ChannelChange_Discontinuity)
 
   uint32_t in_frames = in_rate / 100;
   uint32_t out_frames = out_rate / 100;
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit::Zero(), testPrincipal);
+  AudioResampler dr(in_rate, out_rate, 0, testPrincipal);
 
   AudioChunk monoChunk =
       CreateAudioChunk<float>(in_frames, 1, AUDIO_FORMAT_FLOAT32);
@@ -560,8 +552,7 @@ TEST(TestAudioResampler, ChannelChange_Discontinuity2)
 
   uint32_t in_frames = in_rate / 100;
   uint32_t out_frames = out_rate / 100;
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(10, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, 10, testPrincipal);
 
   AudioChunk monoChunk =
       CreateAudioChunk<float>(in_frames / 2, 1, AUDIO_FORMAT_FLOAT32);
@@ -630,8 +621,7 @@ TEST(TestAudioResampler, ChannelChange_Discontinuity3)
 
   uint32_t in_frames = in_rate / 100;
   uint32_t out_frames = out_rate / 100;
-  AudioResampler dr(in_rate, out_rate, media::TimeUnit(10, in_rate),
-                    testPrincipal);
+  AudioResampler dr(in_rate, out_rate, 10, testPrincipal);
 
   AudioChunk stereoChunk =
       CreateAudioChunk<float>(in_frames, 2, AUDIO_FORMAT_FLOAT32);
@@ -660,9 +650,9 @@ TEST(TestAudioResampler, ChannelChange_Discontinuity3)
 
   // The resampler here is updated due to the rate change. This is because the
   // in and out rate was the same so a pass through logic was used. By updating
-  // the out rate to something different than the in rate, the resampler will
+  // the in rate to something different than the out rate, the resampler will
   // start being used and discontinuity will exist.
-  dr.UpdateOutRate(out_rate + 400);
+  dr.UpdateInRate(in_rate - 400);
   dr.AppendInput(inSegment);
   AudioSegment s2 = dr.Resample(out_frames, &hasUnderrun);
   EXPECT_FALSE(hasUnderrun);
diff --git a/dom/media/driftcontrol/gtest/TestDriftController.cpp b/dom/media/driftcontrol/gtest/TestDriftController.cpp
index 33486f945f..132577e44a 100644
--- a/dom/media/driftcontrol/gtest/TestDriftController.cpp
+++ b/dom/media/driftcontrol/gtest/TestDriftController.cpp
@@ -18,50 +18,50 @@ TEST(TestDriftController, Basic)
   constexpr uint32_t bufferedHigh = 7 * 480;
 
   DriftController c(48000, 48000, media::TimeUnit::FromSeconds(0.05));
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000U);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000U);
 
   // The adjustment interval is 1s.
   const auto oneSec = media::TimeUnit(48000, 48000);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, bufferedLow, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48048u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 47952u);
 
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 47952u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48048u);
 }
 
 TEST(TestDriftController, BasicResampler)
 {
   // The buffer level is the only input to the controller logic.
-  constexpr uint32_t buffered = 5 * 240;
-  constexpr uint32_t bufferedLow = 3 * 240;
-  constexpr uint32_t bufferedHigh = 7 * 240;
+  constexpr uint32_t buffered = 5 * 480;
+  constexpr uint32_t bufferedLow = 3 * 480;
+  constexpr uint32_t bufferedHigh = 7 * 480;
 
-  DriftController c(24000, 48000, media::TimeUnit::FromSeconds(0.05));
+  DriftController c(48000, 24000, media::TimeUnit::FromSeconds(0.05));
 
   // The adjustment interval is 1s.
   const auto oneSec = media::TimeUnit(48000, 48000);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // low
   c.UpdateClock(oneSec, oneSec, bufferedLow, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48048u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 47952u);
 
   // high
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // high
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 47964u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48048u);
 }
 
 TEST(TestDriftController, BufferedInput)
@@ -72,56 +72,56 @@ TEST(TestDriftController, BufferedInput)
   constexpr uint32_t bufferedHigh = 7 * 480;
 
   DriftController c(48000, 48000, media::TimeUnit::FromSeconds(0.05));
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // The adjustment interval is 1s.
   const auto oneSec = media::TimeUnit(48000, 48000);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // 0 buffered when updating correction
   c.UpdateClock(oneSec, oneSec, 0, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48048u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 47952u);
 
   c.UpdateClock(oneSec, oneSec, bufferedLow, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 47952u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48048u);
 }
 
 TEST(TestDriftController, BufferedInputWithResampling)
 {
   // The buffer level is the only input to the controller logic.
-  constexpr uint32_t buffered = 5 * 240;
-  constexpr uint32_t bufferedLow = 3 * 240;
-  constexpr uint32_t bufferedHigh = 7 * 240;
+  constexpr uint32_t buffered = 5 * 480;
+  constexpr uint32_t bufferedLow = 3 * 480;
+  constexpr uint32_t bufferedHigh = 7 * 480;
 
-  DriftController c(24000, 48000, media::TimeUnit::FromSeconds(0.05));
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  DriftController c(48000, 24000, media::TimeUnit::FromSeconds(0.05));
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // The adjustment interval is 1s.
   const auto oneSec = media::TimeUnit(24000, 24000);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // 0 buffered when updating correction
   c.UpdateClock(oneSec, oneSec, 0, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48048u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 47952u);
 
   c.UpdateClock(oneSec, oneSec, bufferedLow, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 47952u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48048u);
 }
 
 TEST(TestDriftController, SmallError)
@@ -132,21 +132,21 @@ TEST(TestDriftController, SmallError)
   constexpr uint32_t bufferedHigh = buffered + 48;
 
   DriftController c(48000, 48000, media::TimeUnit::FromSeconds(0.05));
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   // The adjustment interval is 1s.
   const auto oneSec = media::TimeUnit(48000, 48000);
 
   c.UpdateClock(oneSec, oneSec, buffered, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, bufferedLow, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
   c.UpdateClock(oneSec, oneSec, bufferedHigh, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000u);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000u);
 }
 
 TEST(TestDriftController, SmallBufferedFrames)
@@ -158,11 +158,34 @@ TEST(TestDriftController, SmallBufferedFrames)
   media::TimeUnit oneSec = media::TimeUnit::FromSeconds(1);
   media::TimeUnit hundredMillis = oneSec / 10;
 
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000U);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000U);
   for (uint32_t i = 0; i < 9; ++i) {
     c.UpdateClock(hundredMillis, hundredMillis, bufferedLow, 0);
   }
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48000U);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 48000U);
   c.UpdateClock(hundredMillis, hundredMillis, bufferedLow, 0);
-  EXPECT_EQ(c.GetCorrectedTargetRate(), 48048U);
+  EXPECT_EQ(c.GetCorrectedSourceRate(), 47952U);
+}
+
+TEST(TestDriftController, VerySmallBufferedFrames)
+{
+  // The buffer level is the only input to the controller logic.
+  uint32_t bufferedLow = 1;
+  uint32_t nominalRate = 48000;
+
+  DriftController c(nominalRate, nominalRate, media::TimeUnit::FromSeconds(1));
+  media::TimeUnit oneSec = media::TimeUnit::FromSeconds(1);
+  media::TimeUnit sourceDuration(1, nominalRate);
+
+  EXPECT_EQ(c.GetCorrectedSourceRate(), nominalRate);
+  uint32_t previousCorrected = nominalRate;
+  // Steps are limited to nominalRate/1000.
+  // Perform 1001 steps to check the corrected rate does not underflow zero.
+  for (uint32_t i = 0; i < 1001; ++i) {
+    c.UpdateClock(sourceDuration, oneSec, bufferedLow, 0);
+    uint32_t correctedRate = c.GetCorrectedSourceRate();
+    EXPECT_LE(correctedRate, previousCorrected) << "for i=" << i;
+    EXPECT_GT(correctedRate, 0u) << "for i=" << i;
+    previousCorrected = correctedRate;
+  }
 }
diff --git a/dom/media/driftcontrol/gtest/TestDynamicResampler.cpp b/dom/media/driftcontrol/gtest/TestDynamicResampler.cpp
index fb8ac52ae4..539dfbfbea 100644
--- a/dom/media/driftcontrol/gtest/TestDynamicResampler.cpp
+++ b/dom/media/driftcontrol/gtest/TestDynamicResampler.cpp
@@ -19,7 +19,7 @@ TEST(TestDynamicResampler, SameRates_Float1)
 
   DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
 
   // float in_ch1[] = {.1, .2, .3, .4, .5, .6, .7, .8, .9, 1.0};
@@ -76,7 +76,7 @@ TEST(TestDynamicResampler, SameRates_Short1)
 
   DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_S16);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
 
   short in_ch1[] = {1, 2, 3};
@@ -298,9 +298,9 @@ TEST(TestDynamicResampler, UpdateOutRate_Float)
 
   uint32_t pre_buffer = 20;
 
-  DynamicResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate));
+  DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
 
   float in_ch1[10] = {};
@@ -329,10 +329,10 @@ TEST(TestDynamicResampler, UpdateOutRate_Float)
     EXPECT_FLOAT_EQ(out_ch2[i], 0.0);
   }
 
-  // Update out rate
-  out_rate = 44100;
-  dr.UpdateResampler(out_rate, channels);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  // Update in rate
+  in_rate = 26122;
+  dr.UpdateResampler(in_rate, channels);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
   out_frames = in_frames * out_rate / in_rate;
   EXPECT_EQ(out_frames, 18u);
@@ -354,9 +354,9 @@ TEST(TestDynamicResampler, UpdateOutRate_Short)
 
   uint32_t pre_buffer = 20;
 
-  DynamicResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate));
+  DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_S16);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
 
   short in_ch1[10] = {};
@@ -385,10 +385,10 @@ TEST(TestDynamicResampler, UpdateOutRate_Short)
     EXPECT_EQ(out_ch2[i], 0.0);
   }
 
-  // Update out rate
-  out_rate = 44100;
-  dr.UpdateResampler(out_rate, channels);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  // Update in rate
+  in_rate = 26122;
+  dr.UpdateResampler(in_rate, channels);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
   out_frames = in_frames * out_rate / in_rate;
   EXPECT_EQ(out_frames, 18u);
@@ -400,16 +400,15 @@ TEST(TestDynamicResampler, UpdateOutRate_Short)
   EXPECT_FALSE(hasUnderrun);
 }
 
-TEST(TestDynamicResampler, BigRangeOutRates_Float)
+TEST(TestDynamicResampler, BigRangeInRates_Float)
 {
   uint32_t in_frames = 10;
   uint32_t out_frames = 10;
   uint32_t channels = 2;
   uint32_t in_rate = 44100;
   uint32_t out_rate = 44100;
-  uint32_t pre_buffer = 20;
 
-  DynamicResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate));
+  DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32);
 
   const uint32_t in_capacity = 40;
@@ -427,10 +426,14 @@ TEST(TestDynamicResampler, BigRangeOutRates_Float)
   float out_ch1[out_capacity] = {};
   float out_ch2[out_capacity] = {};
 
-  for (uint32_t rate = 10000; rate < 90000; ++rate) {
-    out_rate = rate;
-    dr.UpdateResampler(out_rate, channels);
-    EXPECT_EQ(dr.GetOutRate(), out_rate);
+  // Downsampling at a high enough ratio happens to have enough excess
+  // in_frames from rounding in the out_frames calculation to cover the
+  // skipped input latency when switching from zero-latency 44100->44100 to a
+  // non-1:1 ratio.
+  for (uint32_t rate = 100000; rate >= 10000; rate -= 2) {
+    in_rate = rate;
+    dr.UpdateResampler(in_rate, channels);
+    EXPECT_EQ(dr.GetInRate(), in_rate);
     EXPECT_EQ(dr.GetChannels(), channels);
     in_frames = 20;  // more than we need
     out_frames = in_frames * out_rate / in_rate;
@@ -444,16 +447,15 @@ TEST(TestDynamicResampler, BigRangeOutRates_Float)
   }
 }
 
-TEST(TestDynamicResampler, BigRangeOutRates_Short)
+TEST(TestDynamicResampler, BigRangeInRates_Short)
 {
   uint32_t in_frames = 10;
   uint32_t out_frames = 10;
   uint32_t channels = 2;
   uint32_t in_rate = 44100;
   uint32_t out_rate = 44100;
-  uint32_t pre_buffer = 20;
 
-  DynamicResampler dr(in_rate, out_rate, media::TimeUnit(pre_buffer, in_rate));
+  DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_S16);
 
   const uint32_t in_capacity = 40;
@@ -471,9 +473,9 @@ TEST(TestDynamicResampler, BigRangeOutRates_Short)
   short out_ch1[out_capacity] = {};
   short out_ch2[out_capacity] = {};
 
-  for (uint32_t rate = 10000; rate < 90000; ++rate) {
-    out_rate = rate;
-    dr.UpdateResampler(out_rate, channels);
+  for (uint32_t rate = 100000; rate >= 10000; rate -= 2) {
+    in_rate = rate;
+    dr.UpdateResampler(in_rate, channels);
     in_frames = 20;  // more than we need
     out_frames = in_frames * out_rate / in_rate;
     for (uint32_t y = 0; y < 2; ++y) {
@@ -517,8 +519,8 @@ TEST(TestDynamicResampler, UpdateChannels_Float)
   EXPECT_FALSE(hasUnderrun);
 
   // Add 3rd channel
-  dr.UpdateResampler(out_rate, 3);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  dr.UpdateResampler(in_rate, 3);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), 3u);
 
   float in_ch3[10] = {};
@@ -546,8 +548,8 @@ TEST(TestDynamicResampler, UpdateChannels_Float)
   in_buffer[3] = in_ch4;
   float out_ch4[10] = {};
 
-  dr.UpdateResampler(out_rate, 4);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  dr.UpdateResampler(in_rate, 4);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), 4u);
   dr.AppendInput(in_buffer, in_frames);
 
@@ -592,8 +594,8 @@ TEST(TestDynamicResampler, UpdateChannels_Short)
   EXPECT_FALSE(hasUnderrun);
 
   // Add 3rd channel
-  dr.UpdateResampler(out_rate, 3);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  dr.UpdateResampler(in_rate, 3);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), 3u);
 
   short in_ch3[10] = {};
@@ -622,8 +624,8 @@ TEST(TestDynamicResampler, UpdateChannels_Short)
   in_buffer[3] = in_ch4;
   short out_ch4[10] = {};
 
-  dr.UpdateResampler(out_rate, 4);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  dr.UpdateResampler(in_rate, 4);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), 4u);
   dr.AppendInput(in_buffer, in_frames);
 
@@ -647,7 +649,7 @@ TEST(TestDynamicResampler, Underrun)
 
   DynamicResampler dr(in_rate, out_rate);
   dr.SetSampleFormat(AUDIO_FORMAT_FLOAT32);
-  EXPECT_EQ(dr.GetOutRate(), out_rate);
+  EXPECT_EQ(dr.GetInRate(), in_rate);
   EXPECT_EQ(dr.GetChannels(), channels);
 
   float in_ch1[in_frames] = {};
@@ -689,7 +691,7 @@ TEST(TestDynamicResampler, Underrun)
   }
 
   // Now try with resampling.
-  dr.UpdateResampler(out_rate / 2, channels);
+  dr.UpdateResampler(in_rate * 2, channels);
   dr.AppendInput(in_buffer, in_frames);
   hasUnderrun = dr.Resample(out_ch1, out_frames, 0);
   EXPECT_TRUE(hasUnderrun);
diff --git a/dom/media/driftcontrol/plot.py b/dom/media/driftcontrol/plot.py
index d55c0f7de0..c3685ead7c 100755
--- a/dom/media/driftcontrol/plot.py
+++ b/dom/media/driftcontrol/plot.py
@@ -86,23 +86,23 @@ MOZ_LOG_FILE=/tmp/driftcontrol.csv       \
             [d + h for (d, h) in zip(desired, hysteresisthreshold)],
             alpha=0.2,
             color="goldenrod",
-            legend_label="Hysteresis Threshold (won't correct out rate within area)",
+            legend_label="Hysteresis Threshold (won't correct in rate within area)",
         )
 
         fig2 = figure(x_range=fig1.x_range)
         fig2.line(t, inrate, color="hotpink", legend_label="Nominal in sample rate")
         fig2.line(t, outrate, color="firebrick", legend_label="Nominal out sample rate")
         fig2.line(
-            t, corrected, color="dodgerblue", legend_label="Corrected out sample rate"
+            t, corrected, color="dodgerblue", legend_label="Corrected in sample rate"
         )
         fig2.line(
             t,
             hysteresiscorrected,
             color="seagreen",
-            legend_label="Hysteresis-corrected out sample rate",
+            legend_label="Hysteresis-corrected in sample rate",
         )
         fig2.line(
-            t, configured, color="goldenrod", legend_label="Configured out sample rate"
+            t, configured, color="goldenrod", legend_label="Configured in sample rate"
         )
 
         fig3 = figure(x_range=fig1.x_range)