diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
index 712eeb268..d39a1dc7e 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
@@ -302,55 +302,6 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PostdecodingVad)) {
   EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
 }
 
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(FlushBuffer)) {
-  const int id = ACMCodecDB::kISAC;
-  EXPECT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels,
-                                   NULL));
-  const int kNumPackets = 5;
-  const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100);
-  for (int n = 0; n < kNumPackets; ++n)
-     InsertOnePacketOfSilence(id);
-  ACMNetworkStatistics statistics;
-  receiver_->NetworkStatistics(&statistics);
-  ASSERT_EQ(num_10ms_frames * kNumPackets * 10, statistics.currentBufferSize);
-
-  receiver_->FlushBuffers();
-  receiver_->NetworkStatistics(&statistics);
-  ASSERT_EQ(0, statistics.currentBufferSize);
-}
-
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PlayoutTimestamp)) {
-  const int id = ACMCodecDB::kPCM16Bwb;
-  EXPECT_EQ(0, receiver_->AddCodec(id, codecs_[id].pltype, codecs_[id].channels,
-                                   NULL));
-  receiver_->SetPlayoutMode(fax);
-  const int kNumPackets = 5;
-  const int num_10ms_frames = codecs_[id].pacsize / (codecs_[id].plfreq / 100);
-  uint32_t expected_timestamp;
-  AudioFrame frame;
-  int ts_offset = 0;
-  bool first_audio_frame = true;
-  for (int n = 0; n < kNumPackets; ++n) {
-    packet_sent_ = false;
-    InsertOnePacketOfSilence(id);
-    ASSERT_TRUE(packet_sent_);
-    expected_timestamp = last_packet_send_timestamp_;
-    for (int k = 0; k < num_10ms_frames; ++k) {
-      ASSERT_EQ(0, receiver_->GetAudio(codecs_[id].plfreq, &frame));
-      if (first_audio_frame) {
-        // There is an offset in playout timestamps. Perhaps, it is related to
-        // initial delay that NetEq applies
-        ts_offset =  receiver_->PlayoutTimestamp() - expected_timestamp;
-        first_audio_frame = false;
-      } else {
-        EXPECT_EQ(expected_timestamp + ts_offset,
-                  receiver_->PlayoutTimestamp());
-      }
-      expected_timestamp += codecs_[id].plfreq / 100;  // Increment by 10 ms.
-    }
-  }
-}
-
 TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(LastAudioCodec)) {
   const int kCodecId[] = {
       ACMCodecDB::kISAC, ACMCodecDB::kPCMA, ACMCodecDB::kISACSWB,
diff --git a/webrtc/modules/audio_coding/neteq4/audio_classifier.h b/webrtc/modules/audio_coding/neteq4/audio_classifier.h
index e64e110d8..1d041b301 100644
--- a/webrtc/modules/audio_coding/neteq4/audio_classifier.h
+++ b/webrtc/modules/audio_coding/neteq4/audio_classifier.h
@@ -41,7 +41,7 @@ class AudioClassifier {
   bool Analysis(const int16_t* input, int input_length, int channels);
 
   // Gets the current classification : true = music, false = speech.
-  bool is_music() const { return is_music_; }
+  virtual bool is_music() const { return is_music_; }
 
   // Gets the current music probability.
   float music_probability() const { return music_probability_; }
diff --git a/webrtc/modules/audio_coding/neteq4/decision_logic.cc b/webrtc/modules/audio_coding/neteq4/decision_logic.cc
index 04b886a2e..85edbb4dc 100644
--- a/webrtc/modules/audio_coding/neteq4/decision_logic.cc
+++ b/webrtc/modules/audio_coding/neteq4/decision_logic.cc
@@ -145,8 +145,8 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
                                 reset_decoder);
 }
 
-void DecisionLogic::ExpandDecision(bool is_expand_decision) {
-  if (is_expand_decision) {
+void DecisionLogic::ExpandDecision(Operations operation) {
+  if (operation == kExpand) {
     num_consecutive_expands_++;
   } else {
     num_consecutive_expands_ = 0;
diff --git a/webrtc/modules/audio_coding/neteq4/decision_logic.h b/webrtc/modules/audio_coding/neteq4/decision_logic.h
index aca5ca405..e187449b6 100644
--- a/webrtc/modules/audio_coding/neteq4/decision_logic.h
+++ b/webrtc/modules/audio_coding/neteq4/decision_logic.h
@@ -92,7 +92,7 @@ class DecisionLogic {
   // not. Note that this is necessary, since an expand decision can be changed
   // to kNormal in NetEqImpl::GetDecision if there is still enough data in the
   // sync buffer.
-  void ExpandDecision(bool is_expand_decision);
+  virtual void ExpandDecision(Operations operation);
 
   // Adds |value| to |sample_memory_|.
   void AddSampleMemory(int32_t value) {
diff --git a/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h b/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h
index 783b001fc..8ec60349f 100644
--- a/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h
+++ b/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h
@@ -38,6 +38,10 @@ class DecisionLogicNormal : public DecisionLogic {
   virtual ~DecisionLogicNormal() {}
 
  protected:
+  static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
+  static const int kReinitAfterExpands = 100;
+  static const int kMaxWaitForPacket = 10;
+
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples
   // obtained from the last decoded frame. If there is a packet available, the
@@ -54,32 +58,29 @@ class DecisionLogicNormal : public DecisionLogic {
                                             Modes prev_mode, bool play_dtmf,
                                             bool* reset_decoder);
 
- private:
-  static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
-  static const int kReinitAfterExpands = 100;
-  static const int kMaxWaitForPacket = 10;
+  // Returns the operation to do given that the expected packet is not
+  // available, but a packet further into the future is at hand.
+  virtual Operations FuturePacketAvailable(
+      const SyncBuffer& sync_buffer,
+      const Expand& expand,
+      int decoder_frame_length, Modes prev_mode,
+      uint32_t target_timestamp,
+      uint32_t available_timestamp,
+      bool play_dtmf);
 
+  // Returns the operation to do given that the expected packet is available.
+  virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
+
+  // Returns the operation given that no packets are available (except maybe
+  // a DTMF event, flagged by setting |play_dtmf| true).
+  virtual Operations NoPacket(bool play_dtmf);
+
+ private:
   // Returns the operation given that the next available packet is a comfort
   // noise payload (RFC 3389 only, not codec-internal).
   Operations CngOperation(Modes prev_mode, uint32_t target_timestamp,
                           uint32_t available_timestamp);
 
-  // Returns the operation given that no packets are available (except maybe
-  // a DTMF event, flagged by setting |play_dtmf| true).
-  Operations NoPacket(bool play_dtmf);
-
-  // Returns the operation to do given that the expected packet is available.
-  Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
-
-  // Returns the operation to do given that the expected packet is not
-  // available, but a packet further into the future is at hand.
-  Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
-                                   const Expand& expand,
-                                   int decoder_frame_length, Modes prev_mode,
-                                   uint32_t target_timestamp,
-                                   uint32_t available_timestamp,
-                                   bool play_dtmf);
-
   // Checks if enough time has elapsed since the last successful timescale
   // operation was done (i.e., accelerate or preemptive expand).
   bool TimescaleAllowed() const { return timescale_hold_off_ == 0; }
diff --git a/webrtc/modules/audio_coding/neteq4/expand.cc b/webrtc/modules/audio_coding/neteq4/expand.cc
index cba99243d..9b08de15a 100644
--- a/webrtc/modules/audio_coding/neteq4/expand.cc
+++ b/webrtc/modules/audio_coding/neteq4/expand.cc
@@ -56,20 +56,9 @@ int Expand::Process(AudioMultiVector* output) {
     // This is not the first expansion, parameters are already estimated.
     // Extract a noise segment.
     int16_t rand_length = max_lag_;
-    // TODO(hlundin): This if-statement should not be needed. Should be just
-    // as good to generate all of the vector in one call in either case.
-    if (rand_length <= RandomVector::kRandomTableSize) {
-      random_vector_->IncreaseSeedIncrement(2);
-      random_vector_->Generate(rand_length, random_vector);
-    } else {
-      // This only applies to SWB where length could be larger than 256.
-      assert(rand_length <= kMaxSampleRate / 8000 * 120 + 30);
-      random_vector_->IncreaseSeedIncrement(2);
-      random_vector_->Generate(RandomVector::kRandomTableSize, random_vector);
-      random_vector_->IncreaseSeedIncrement(2);
-      random_vector_->Generate(rand_length - RandomVector::kRandomTableSize,
-                               &random_vector[RandomVector::kRandomTableSize]);
-    }
+    // This only applies to SWB where length could be larger than 256.
+    assert(rand_length <= kMaxSampleRate / 8000 * 120 + 30);
+    GenerateRandomVector(2, rand_length, random_vector);
   }
 
 
@@ -262,82 +251,12 @@ int Expand::Process(AudioMultiVector* output) {
     }
 
     // Background noise part.
-    // TODO(hlundin): Move to separate method? In BackgroundNoise class?
-    if (background_noise_->initialized()) {
-      // Use background noise parameters.
-      memcpy(noise_vector - kNoiseLpcOrder,
-             background_noise_->FilterState(channel_ix),
-             sizeof(int16_t) * kNoiseLpcOrder);
-
-      if (background_noise_->ScaleShift(channel_ix) > 1) {
-        add_constant = 1 << (background_noise_->ScaleShift(channel_ix) - 1);
-      } else {
-        add_constant = 0;
-      }
-
-      // Scale random vector to correct energy level.
-      WebRtcSpl_AffineTransformVector(
-          scaled_random_vector, random_vector,
-          background_noise_->Scale(channel_ix), add_constant,
-          background_noise_->ScaleShift(channel_ix),
-          static_cast<int>(current_lag));
-
-      WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_vector,
-                                background_noise_->Filter(channel_ix),
-                                kNoiseLpcOrder + 1,
-                                static_cast<int>(current_lag));
-
-      background_noise_->SetFilterState(
-          channel_ix,
-          &(noise_vector[current_lag - kNoiseLpcOrder]),
-          kNoiseLpcOrder);
-
-      // Unmute the background noise.
-      int16_t bgn_mute_factor = background_noise_->MuteFactor(channel_ix);
-      NetEqBackgroundNoiseMode bgn_mode = background_noise_->mode();
-      if (bgn_mode == kBgnFade &&
-          consecutive_expands_ >= kMaxConsecutiveExpands &&
-          bgn_mute_factor > 0) {
-        // Fade BGN to zero.
-        // Calculate muting slope, approximately -2^18 / fs_hz.
-        int16_t mute_slope;
-        if (fs_hz_ == 8000) {
-          mute_slope = -32;
-        } else if (fs_hz_ == 16000) {
-          mute_slope = -16;
-        } else if (fs_hz_ == 32000) {
-          mute_slope = -8;
-        } else {
-          mute_slope = -5;
-        }
-        // Use UnmuteSignal function with negative slope.
-        // |bgn_mute_factor| is in Q14. |mute_slope| is in Q20.
-        DspHelper::UnmuteSignal(noise_vector, current_lag, &bgn_mute_factor,
-                                mute_slope, noise_vector);
-      } else if (bgn_mute_factor < 16384) {
-        // If mode is kBgnOff, or if kBgnFade has started fading,
-        // Use regular |mute_slope|.
-        if (!stop_muting_ && bgn_mode != kBgnOff &&
-            !(bgn_mode == kBgnFade &&
-                consecutive_expands_ >= kMaxConsecutiveExpands)) {
-          DspHelper::UnmuteSignal(noise_vector, static_cast<int>(current_lag),
-                                  &bgn_mute_factor, parameters.mute_slope,
-                                  noise_vector);
-        } else {
-          // kBgnOn and stop muting, or
-          // kBgnOff (mute factor is always 0), or
-          // kBgnFade has reached 0.
-          WebRtcSpl_AffineTransformVector(noise_vector, noise_vector,
-                                          bgn_mute_factor, 8192, 14,
-                                          static_cast<int>(current_lag));
-        }
-      }
-      // Update mute_factor in BackgroundNoise class.
-      background_noise_->SetMuteFactor(channel_ix, bgn_mute_factor);
-    } else {
-      // BGN parameters have not been initialized; use zero noise.
-      memset(noise_vector, 0, sizeof(int16_t) * current_lag);
-    }
+    GenerateBackgroundNoise(random_vector,
+                            channel_ix,
+                            channel_parameters_[channel_ix].mute_slope,
+                            TooManyExpands(),
+                            current_lag,
+                            unvoiced_array_memory);
 
     // Add background noise to the combined voiced-unvoiced signal.
     for (size_t i = 0; i < current_lag; i++) {
@@ -353,11 +272,8 @@ int Expand::Process(AudioMultiVector* output) {
   }
 
   // Increase call number and cap it.
-  ++consecutive_expands_;
-  if (consecutive_expands_ > kMaxConsecutiveExpands) {
-    consecutive_expands_ = kMaxConsecutiveExpands;
-  }
-
+  consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ?
+      kMaxConsecutiveExpands : consecutive_expands_ + 1;
   return 0;
 }
 
@@ -373,6 +289,24 @@ void Expand::SetParametersForMergeAfterExpand() {
   stop_muting_ = true;
 }
 
+void Expand::InitializeForAnExpandPeriod() {
+  lag_index_direction_ = 1;
+  current_lag_index_ = -1;
+  stop_muting_ = false;
+  random_vector_->set_seed_increment(1);
+  consecutive_expands_ = 0;
+  for (size_t ix = 0; ix < num_channels_; ++ix) {
+    channel_parameters_[ix].current_voice_mix_factor = 16384;  // 1.0 in Q14.
+    channel_parameters_[ix].mute_factor = 16384;  // 1.0 in Q14.
+    // Start with 0 gain for background noise.
+    background_noise_->SetMuteFactor(ix, 0);
+  }
+}
+
+bool Expand::TooManyExpands() {
+  return consecutive_expands_ >= kMaxConsecutiveExpands;
+}
+
 void Expand::AnalyzeSignal(int16_t* random_vector) {
   int32_t auto_correlation[kUnvoicedLpcOrder + 1];
   int16_t reflection_coeff[kUnvoicedLpcOrder];
@@ -400,18 +334,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   const int16_t* audio_history =
       &(*sync_buffer_)[0][sync_buffer_->Size() - signal_length];
 
-  // Initialize some member variables.
-  lag_index_direction_ = 1;
-  current_lag_index_ = -1;
-  stop_muting_ = false;
-  random_vector_->set_seed_increment(1);
-  consecutive_expands_ = 0;
-  for (size_t ix = 0; ix < num_channels_; ++ix) {
-    channel_parameters_[ix].current_voice_mix_factor = 16384;  // 1.0 in Q14.
-    channel_parameters_[ix].mute_factor = 16384;  // 1.0 in Q14.
-    // Start with 0 gain for background noise.
-    background_noise_->SetMuteFactor(ix, 0);
-  }
+  // Initialize.
+  InitializeForAnExpandPeriod();
 
   // Calculate correlation in downsampled domain (4 kHz sample rate).
   int16_t correlation_scale;
@@ -873,5 +797,108 @@ Expand* ExpandFactory::Create(BackgroundNoise* background_noise,
                     num_channels);
 }
 
+// TODO(turajs): This can be moved to BackgroundNoise class.
+void Expand::GenerateBackgroundNoise(int16_t* random_vector,
+                                     size_t channel,
+                                     int16_t mute_slope,
+                                     bool too_many_expands,
+                                     size_t num_noise_samples,
+                                     int16_t* buffer) {
+  static const int kNoiseLpcOrder = BackgroundNoise::kMaxLpcOrder;
+  int16_t scaled_random_vector[kMaxSampleRate / 8000 * 125];
+  assert(kMaxSampleRate / 8000 * 125 >= (int)num_noise_samples);
+  int16_t* noise_samples = &buffer[kNoiseLpcOrder];
+  if (background_noise_->initialized()) {
+    // Use background noise parameters.
+    memcpy(noise_samples - kNoiseLpcOrder,
+           background_noise_->FilterState(channel),
+           sizeof(int16_t) * kNoiseLpcOrder);
+
+    int dc_offset = 0;
+    if (background_noise_->ScaleShift(channel) > 1) {
+      dc_offset = 1 << (background_noise_->ScaleShift(channel) - 1);
+    }
+
+    // Scale random vector to correct energy level.
+    WebRtcSpl_AffineTransformVector(
+        scaled_random_vector, random_vector,
+        background_noise_->Scale(channel), dc_offset,
+        background_noise_->ScaleShift(channel),
+        static_cast<int>(num_noise_samples));
+
+    WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples,
+                              background_noise_->Filter(channel),
+                              kNoiseLpcOrder + 1,
+                              static_cast<int>(num_noise_samples));
+
+    background_noise_->SetFilterState(
+        channel,
+        &(noise_samples[num_noise_samples - kNoiseLpcOrder]),
+        kNoiseLpcOrder);
+
+    // Unmute the background noise.
+    int16_t bgn_mute_factor = background_noise_->MuteFactor(channel);
+    NetEqBackgroundNoiseMode bgn_mode = background_noise_->mode();
+    if (bgn_mode == kBgnFade && too_many_expands && bgn_mute_factor > 0) {
+      // Fade BGN to zero.
+      // Calculate muting slope, approximately -2^18 / fs_hz.
+      int16_t mute_slope;
+      if (fs_hz_ == 8000) {
+        mute_slope = -32;
+      } else if (fs_hz_ == 16000) {
+        mute_slope = -16;
+      } else if (fs_hz_ == 32000) {
+        mute_slope = -8;
+      } else {
+        mute_slope = -5;
+      }
+      // Use UnmuteSignal function with negative slope.
+      // |bgn_mute_factor| is in Q14. |mute_slope| is in Q20.
+      DspHelper::UnmuteSignal(noise_samples,
+                              num_noise_samples,
+                              &bgn_mute_factor,
+                              mute_slope,
+                              noise_samples);
+    } else if (bgn_mute_factor < 16384) {
+      // If mode is kBgnOff, or if kBgnFade has started fading,
+      // Use regular |mute_slope|.
+      if (!stop_muting_ && bgn_mode != kBgnOff &&
+          !(bgn_mode == kBgnFade && too_many_expands)) {
+        DspHelper::UnmuteSignal(noise_samples,
+                                static_cast<int>(num_noise_samples),
+                                &bgn_mute_factor,
+                                mute_slope,
+                                noise_samples);
+      } else {
+        // kBgnOn and stop muting, or
+        // kBgnOff (mute factor is always 0), or
+        // kBgnFade has reached 0.
+        WebRtcSpl_AffineTransformVector(noise_samples, noise_samples,
+                                        bgn_mute_factor, 8192, 14,
+                                        static_cast<int>(num_noise_samples));
+      }
+    }
+    // Update mute_factor in BackgroundNoise class.
+    background_noise_->SetMuteFactor(channel, bgn_mute_factor);
+  } else {
+    // BGN parameters have not been initialized; use zero noise.
+    memset(noise_samples, 0, sizeof(int16_t) * num_noise_samples);
+  }
+}
+
+void Expand::GenerateRandomVector(int seed_increment,
+                                  size_t length,
+                                  int16_t* random_vector) {
+  // TODO(turajs): According to hlundin The loop should not be needed. Should be
+  // just as good to generate all of the vector in one call.
+  size_t samples_generated = 0;
+  const size_t kMaxRandSamples = RandomVector::kRandomTableSize;
+  while(samples_generated < length) {
+    size_t rand_length = std::min(length - samples_generated, kMaxRandSamples);
+    random_vector_->IncreaseSeedIncrement(seed_increment);
+    random_vector_->Generate(rand_length, &random_vector[samples_generated]);
+    samples_generated += rand_length;
+  }
+}
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/expand.h b/webrtc/modules/audio_coding/neteq4/expand.h
index 4de8d7c55..aec1cd9b6 100644
--- a/webrtc/modules/audio_coding/neteq4/expand.h
+++ b/webrtc/modules/audio_coding/neteq4/expand.h
@@ -36,12 +36,13 @@ class Expand {
          RandomVector* random_vector,
          int fs,
          size_t num_channels)
-      : background_noise_(background_noise),
+      : random_vector_(random_vector),
         sync_buffer_(sync_buffer),
-        random_vector_(random_vector),
         first_expand_(true),
         fs_hz_(fs),
         num_channels_(num_channels),
+        consecutive_expands_(0),
+        background_noise_(background_noise),
         overlap_length_(5 * fs / 8000),
         lag_index_direction_(0),
         current_lag_index_(0),
@@ -57,19 +58,19 @@ class Expand {
   virtual ~Expand() {}
 
   // Resets the object.
-  void Reset();
+  virtual void Reset();
 
   // The main method to produce concealment data. The data is appended to the
   // end of |output|.
-  int Process(AudioMultiVector* output);
+  virtual int Process(AudioMultiVector* output);
 
   // Prepare the object to do extra expansion during normal operation following
   // a period of expands.
-  void SetParametersForNormalAfterExpand();
+  virtual void SetParametersForNormalAfterExpand();
 
   // Prepare the object to do extra expansion during merge operation following
   // a period of expands.
-  void SetParametersForMergeAfterExpand();
+  virtual void SetParametersForMergeAfterExpand();
 
   // Sets the mute factor for |channel| to |value|.
   void SetMuteFactor(int16_t value, size_t channel) {
@@ -84,9 +85,38 @@ class Expand {
   }
 
   // Accessors and mutators.
-  size_t overlap_length() const { return overlap_length_; }
+  virtual size_t overlap_length() const { return overlap_length_; }
   int16_t max_lag() const { return max_lag_; }
 
+ protected:
+  static const int kMaxConsecutiveExpands = 200;
+  void GenerateRandomVector(int seed_increment,
+                            size_t length,
+                            int16_t* random_vector);
+
+  void GenerateBackgroundNoise(int16_t* random_vector,
+                               size_t channel,
+                               int16_t mute_slope,
+                               bool too_many_expands,
+                               size_t num_noise_samples,
+                               int16_t* buffer);
+
+  // Initializes member variables at the beginning of an expand period.
+  void InitializeForAnExpandPeriod();
+
+  bool TooManyExpands();
+
+  // Analyzes the signal history in |sync_buffer_|, and set up all parameters
+  // necessary to produce concealment data.
+  void AnalyzeSignal(int16_t* random_vector);
+
+  RandomVector* random_vector_;
+  SyncBuffer* sync_buffer_;
+  bool first_expand_;
+  const int fs_hz_;
+  const size_t num_channels_;
+  int consecutive_expands_;
+
  private:
   static const int kUnvoicedLpcOrder = 6;
   static const int kNumCorrelationCandidates = 3;
@@ -94,7 +124,6 @@ class Expand {
   static const int kLpcAnalysisLength = 160;
   static const int kMaxSampleRate = 48000;
   static const int kNumLags = 3;
-  static const int kMaxConsecutiveExpands = 200;
 
   struct ChannelParameters {
     // Constructor.
@@ -122,10 +151,6 @@ class Expand {
     int16_t mute_slope; /* Q20 */
   };
 
-  // Analyze the signal history in |sync_buffer_|, and set up all parameters
-  // necessary to produce concealment data.
-  void AnalyzeSignal(int16_t* random_vector);
-
   // Calculate the auto-correlation of |input|, with length |input_length|
   // samples. The correlation is calculated from a downsampled version of
   // |input|, and is written to |output|. The scale factor is written to
@@ -136,13 +161,7 @@ class Expand {
   void UpdateLagIndex();
 
   BackgroundNoise* background_noise_;
-  SyncBuffer* sync_buffer_;
-  RandomVector* random_vector_;
-  bool first_expand_;
-  const int fs_hz_;
-  const size_t num_channels_;
   const size_t overlap_length_;
-  int consecutive_expands_;
   int16_t max_lag_;
   size_t expand_lags_[kNumLags];
   int lag_index_direction_;
diff --git a/webrtc/modules/audio_coding/neteq4/interface/neteq.h b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
index 466882a5f..c0f7fd69b 100644
--- a/webrtc/modules/audio_coding/neteq4/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq4/interface/neteq.h
@@ -108,7 +108,8 @@ class NetEq {
   // Creates a new NetEq object, starting at the sample rate |sample_rate_hz|.
   // (Note that it will still change the sample rate depending on what payloads
   // are being inserted; |sample_rate_hz| is just for startup configuration.)
-  static NetEq* Create(int sample_rate_hz);
+  static NetEq* Create(int sample_rate_hz,
+                       bool enable_audio_classifier = false);
 
   virtual ~NetEq() {}
 
diff --git a/webrtc/modules/audio_coding/neteq4/merge.cc b/webrtc/modules/audio_coding/neteq4/merge.cc
index 4b5601657..90068bb4a 100644
--- a/webrtc/modules/audio_coding/neteq4/merge.cc
+++ b/webrtc/modules/audio_coding/neteq4/merge.cc
@@ -20,6 +20,7 @@
 #include "webrtc/modules/audio_coding/neteq4/dsp_helper.h"
 #include "webrtc/modules/audio_coding/neteq4/expand.h"
 #include "webrtc/modules/audio_coding/neteq4/sync_buffer.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
 
 namespace webrtc {
 
@@ -307,9 +308,11 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
                              stop_position_downsamp, correlation_shift, 1);
 
   // Normalize correlation to 14 bits and copy to a 16-bit array.
-  static const int kPadLength = 4;
-  int16_t correlation16[kPadLength + kMaxCorrelationLength + kPadLength] = {0};
-  int16_t* correlation_ptr = &correlation16[kPadLength];
+  const int pad_length = static_cast<int>(expand_->overlap_length() - 1);
+  const int correlation_buffer_size = 2 * pad_length + kMaxCorrelationLength;
+  scoped_ptr<int16_t[]> correlation16(new int16_t[correlation_buffer_size]);
+  memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
+  int16_t* correlation_ptr = &correlation16[pad_length];
   int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
                                                      stop_position_downsamp);
   int16_t norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
@@ -332,7 +335,7 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   // start index |start_index_downsamp| and the effective array length.
   int modified_stop_pos =
       std::min(stop_position_downsamp,
-               kMaxCorrelationLength + kPadLength - start_index_downsamp);
+               kMaxCorrelationLength + pad_length - start_index_downsamp);
   int best_correlation_index;
   int16_t best_correlation;
   static const int kNumCorrelationCandidates = 1;
@@ -355,4 +358,9 @@ int16_t Merge::CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
   return best_correlation_index;
 }
 
+int Merge::RequiredFutureSamples() {
+  return static_cast<int>(fs_hz_ / 100 * num_channels_);  // 10 ms.
+}
+
+
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/merge.h b/webrtc/modules/audio_coding/neteq4/merge.h
index f1f64e6c5..213b48716 100644
--- a/webrtc/modules/audio_coding/neteq4/merge.h
+++ b/webrtc/modules/audio_coding/neteq4/merge.h
@@ -35,8 +35,8 @@ class Merge {
  public:
   Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
       : fs_hz_(fs_hz),
-        fs_mult_(fs_hz_ / 8000),
         num_channels_(num_channels),
+        fs_mult_(fs_hz_ / 8000),
         timestamps_per_call_(fs_hz_ / 100),
         expand_(expand),
         sync_buffer_(sync_buffer),
@@ -44,6 +44,8 @@ class Merge {
     assert(num_channels_ > 0);
   }
 
+  virtual ~Merge() {}
+
   // The main method to produce the audio data. The decoded data is supplied in
   // |input|, having |input_length| samples in total for all channels
   // (interleaved). The result is written to |output|. The number of channels
@@ -51,9 +53,15 @@ class Merge {
   // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
   // will be used to scale the audio, and is updated in the process. The array
   // must have |num_channels_| elements.
-  int Process(int16_t* input, size_t input_length,
-              int16_t* external_mute_factor_array,
-              AudioMultiVector* output);
+  virtual int Process(int16_t* input, size_t input_length,
+                      int16_t* external_mute_factor_array,
+                      AudioMultiVector* output);
+
+  virtual int RequiredFutureSamples();
+
+ protected:
+  const int fs_hz_;
+  const size_t num_channels_;
 
  private:
   static const int kMaxSampleRate = 48000;
@@ -87,9 +95,7 @@ class Merge {
                                  int start_position, int input_length,
                                  int expand_period) const;
 
-  const int fs_hz_;
   const int fs_mult_;  // fs_hz_ / 8000.
-  const size_t num_channels_;
   const int timestamps_per_call_;
   Expand* expand_;
   SyncBuffer* sync_buffer_;
diff --git a/webrtc/modules/audio_coding/neteq4/neteq.cc b/webrtc/modules/audio_coding/neteq4/neteq.cc
index a64f01b25..9f36a9655 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq.cc
@@ -28,7 +28,7 @@ namespace webrtc {
 
 // Creates all classes needed and inject them into a new NetEqImpl object.
 // Return the new object.
-NetEq* NetEq::Create(int sample_rate_hz) {
+NetEq* NetEq::Create(int sample_rate_hz, bool enable_audio_classifier) {
   BufferLevelFilter* buffer_level_filter = new BufferLevelFilter;
   DecoderDatabase* decoder_database = new DecoderDatabase;
   DelayPeakDetector* delay_peak_detector = new DelayPeakDetector;
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
index 8ab6205ec..e407ee83e 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.cc
@@ -61,7 +61,8 @@ NetEqImpl::NetEqImpl(int fs,
                      TimestampScaler* timestamp_scaler,
                      AccelerateFactory* accelerate_factory,
                      ExpandFactory* expand_factory,
-                     PreemptiveExpandFactory* preemptive_expand_factory)
+                     PreemptiveExpandFactory* preemptive_expand_factory,
+                     bool create_components)
     : buffer_level_filter_(buffer_level_filter),
       decoder_database_(decoder_database),
       delay_manager_(delay_manager),
@@ -103,13 +104,9 @@ NetEqImpl::NetEqImpl(int fs,
   output_size_samples_ = kOutputSizeMs * 8 * fs_mult_;
   decoder_frame_length_ = 3 * output_size_samples_;
   WebRtcSpl_Init();
-  decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
-                                              kPlayoutOn,
-                                              decoder_database_.get(),
-                                              *packet_buffer_.get(),
-                                              delay_manager_.get(),
-                                              buffer_level_filter_.get()));
-  SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
+  if (create_components) {
+    SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
+  }
 }
 
 NetEqImpl::~NetEqImpl() {
@@ -284,12 +281,7 @@ void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
   CriticalSectionScoped lock(crit_sect_.get());
   if (!decision_logic_.get() || mode != decision_logic_->playout_mode()) {
     // The reset() method calls delete for the old object.
-    decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
-                                                mode,
-                                                decoder_database_.get(),
-                                                *packet_buffer_.get(),
-                                                delay_manager_.get(),
-                                                buffer_level_filter_.get()));
+    CreateDecisionLogic(mode);
   }
 }
 
@@ -948,7 +940,7 @@ int NetEqImpl::GetDecision(Operations* operation,
     return 0;
   }
 
-  decision_logic_->ExpandDecision(*operation == kExpand);
+  decision_logic_->ExpandDecision(*operation);
 
   // Check conditions for reset.
   if (new_codec_ || *operation == kUndefined) {
@@ -1067,6 +1059,11 @@ int NetEqImpl::GetDecision(Operations* operation,
       // Move on with the preemptive expand decision.
       break;
     }
+    case kMerge: {
+      required_samples =
+          std::max(merge_->RequiredFutureSamples(), required_samples);
+      break;
+    }
     default: {
       // Do nothing.
     }
@@ -1834,6 +1831,14 @@ int NetEqImpl::ExtractPackets(int required_samples, PacketList* packet_list) {
   return extracted_samples;
 }
 
+void NetEqImpl::UpdatePlcComponents(int fs_hz, size_t channels) {
+  // Delete objects and create new ones.
+  expand_.reset(expand_factory_->Create(background_noise_.get(),
+                                        sync_buffer_.get(), &random_vector_,
+                                        fs_hz, channels));
+  merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
+}
+
 void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
   LOG_API2(fs_hz, channels);
   // TODO(hlundin): Change to an enumerator and skip assert.
@@ -1881,21 +1886,20 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
   // Reset random vector.
   random_vector_.Reset();
 
-  // Delete Expand object and create a new one.
-  expand_.reset(expand_factory_->Create(background_noise_.get(),
-                                        sync_buffer_.get(), &random_vector_,
-                                        fs_hz, channels));
+  UpdatePlcComponents(fs_hz, channels);
+
   // Move index so that we create a small set of future samples (all 0).
   sync_buffer_->set_next_index(sync_buffer_->next_index() -
-                               expand_->overlap_length());
+      expand_->overlap_length());
 
   normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_,
                            expand_.get()));
-  merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
   accelerate_.reset(
       accelerate_factory_->Create(fs_hz, channels, *background_noise_));
-  preemptive_expand_.reset(
-      preemptive_expand_factory_->Create(fs_hz, channels, *background_noise_));
+  preemptive_expand_.reset(preemptive_expand_factory_->Create(
+      fs_hz, channels,
+      *background_noise_,
+      static_cast<int>(expand_->overlap_length())));
 
   // Delete ComfortNoise object and create a new one.
   comfort_noise_.reset(new ComfortNoise(fs_hz, decoder_database_.get(),
@@ -1908,8 +1912,11 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
     decoded_buffer_.reset(new int16_t[decoded_buffer_length_]);
   }
 
-  // Communicate new sample rate and output size to DecisionLogic object.
-  assert(decision_logic_.get());
+  // Create DecisionLogic if it is not created yet, then communicate new sample
+  // rate and output size to DecisionLogic object.
+  if (!decision_logic_.get()) {
+    CreateDecisionLogic(kPlayoutOn);
+  }
   decision_logic_->SetSampleRate(fs_hz_, output_size_samples_);
 }
 
@@ -1930,4 +1937,12 @@ NetEqOutputType NetEqImpl::LastOutputType() {
   }
 }
 
+void NetEqImpl::CreateDecisionLogic(NetEqPlayoutMode mode) {
+  decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
+                                              mode,
+                                              decoder_database_.get(),
+                                              *packet_buffer_.get(),
+                                              delay_manager_.get(),
+                                              buffer_level_filter_.get()));
+}
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_impl.h b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
index dabf2d6cb..3d883a014 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq4/neteq_impl.h
@@ -70,7 +70,8 @@ class NetEqImpl : public webrtc::NetEq {
             TimestampScaler* timestamp_scaler,
             AccelerateFactory* accelerate_factory,
             ExpandFactory* expand_factory,
-            PreemptiveExpandFactory* preemptive_expand_factory);
+            PreemptiveExpandFactory* preemptive_expand_factory,
+            bool create_components = true);
 
   virtual ~NetEqImpl();
 
@@ -203,7 +204,7 @@ class NetEqImpl : public webrtc::NetEq {
   // This accessor method is only intended for testing purposes.
   virtual const SyncBuffer* sync_buffer_for_test() const;
 
- private:
+ protected:
   static const int kOutputSizeMs = 10;
   static const int kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
@@ -331,6 +332,14 @@ class NetEqImpl : public webrtc::NetEq {
   // GetAudio().
   NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
+  // Updates Expand and Merge.
+  virtual void UpdatePlcComponents(int fs_hz, size_t channels)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
+  // Creates DecisionLogic object for the given mode.
+  void CreateDecisionLogic(NetEqPlayoutMode mode)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+
   const scoped_ptr<BufferLevelFilter> buffer_level_filter_;
   const scoped_ptr<DecoderDatabase> decoder_database_;
   const scoped_ptr<DelayManager> delay_manager_;
@@ -388,6 +397,7 @@ class NetEqImpl : public webrtc::NetEq {
   int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
   uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
 
+ private:
   DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi b/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi
index 9d0aa4233..dda2a1ce8 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi
+++ b/webrtc/modules/audio_coding/neteq4/neteq_tests.gypi
@@ -49,6 +49,7 @@
         'CODEC_PCM16B_WB',
         'CODEC_ISAC_SWB',
         'CODEC_PCM16B_32KHZ',
+        'CODEC_PCM16B_48KHZ',
         'CODEC_CNGCODEC8',
         'CODEC_CNGCODEC16',
         'CODEC_CNGCODEC32',
diff --git a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
index 365e233e6..1e8c1288a 100644
--- a/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
@@ -18,6 +18,7 @@
 #include <stdlib.h>
 #include <string.h>  // memset
 
+#include <algorithm>
 #include <set>
 #include <string>
 #include <vector>
@@ -232,6 +233,7 @@ class NetEqDecodingTest : public ::testing::Test {
   unsigned int sim_clock_;
   int16_t out_data_[kMaxBlockSize];
   int output_sample_rate_;
+  int algorithmic_delay_ms_;
 };
 
 // Allocating the static const so that it can be passed by reference.
@@ -246,12 +248,16 @@ NetEqDecodingTest::NetEqDecodingTest()
     : neteq_(NULL),
       rtp_fp_(NULL),
       sim_clock_(0),
-      output_sample_rate_(kInitSampleRateHz) {
+      output_sample_rate_(kInitSampleRateHz),
+      algorithmic_delay_ms_(0) {
   memset(out_data_, 0, sizeof(out_data_));
 }
 
 void NetEqDecodingTest::SetUp() {
   neteq_ = NetEq::Create(kInitSampleRateHz);
+  NetEqNetworkStatistics stat;
+  ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
+  algorithmic_delay_ms_ = stat.current_buffer_size_ms;
   ASSERT_TRUE(neteq_);
   LoadDecoders();
 }
@@ -483,8 +489,8 @@ void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz,
   ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
 
   // To be able to test the fading of background noise we need at lease to pull
-  // 610 frames.
-  const int kFadingThreshold = 610;
+  // 611 frames.
+  const int kFadingThreshold = 611;
 
   // Test several CNG-to-PLC packet for the expected behavior. The number 20 is
   // arbitrary, but sufficiently large to test enough number of frames.
@@ -1110,12 +1116,16 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketInsert)) {
 // First insert several noise like packets, then sync-packets. Decoding all
 // packets should not produce error, statistics should not show any packet loss
 // and sync-packets should decode to zero.
+// TODO(turajs) we will have a better test if we have a referece NetEq, and
+// when Sync packets are inserted in "test" NetEq we insert all-zero payload
+// in reference NetEq and compare the output of those two.
 TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
   uint8_t payload[kPayloadBytes];
   int16_t decoded[kBlockSize16kHz];
+  int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
   for (int n = 0; n < kPayloadBytes; ++n) {
     payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
   }
@@ -1125,7 +1135,6 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
   int num_channels;
   int samples_per_channel;
   uint32_t receive_timestamp = 0;
-  int delay_samples = 0;
   for (int n = 0; n < 100; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                       receive_timestamp));
@@ -1135,16 +1144,15 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
     ASSERT_EQ(1, num_channels);
 
-    // Even if there is RTP packet in NetEq's buffer, the first frame pulled
-    // from NetEq starts with few zero samples. Here we measure this delay.
-    if (n == 0) {
-      while (decoded[delay_samples] == 0) delay_samples++;
-    }
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
   }
   const int kNumSyncPackets = 10;
+
+  // Make sure sufficient number of sync packets are inserted that we can
+  // conduct a test.
+  ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay);
   // Insert sync-packets, the decoded sequence should be all-zero.
   for (int n = 0; n < kNumSyncPackets; ++n) {
     ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
@@ -1153,30 +1161,37 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
     ASSERT_EQ(1, num_channels);
-    EXPECT_TRUE(IsAllZero(&decoded[delay_samples],
-                          samples_per_channel * num_channels - delay_samples));
-    delay_samples = 0;  // Delay only matters in the first frame.
+    if (n > algorithmic_frame_delay) {
+      EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
+    }
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
   }
-  // We insert a regular packet, if sync packet are not correctly buffered then
+
+  // We insert regular packets, if sync packet are not correctly buffered then
   // network statistics would show some packet loss.
-  ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
-                                    receive_timestamp));
-  ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
-                                &samples_per_channel, &num_channels,
-                                &output_type));
-  // Make sure the last inserted packet is decoded and there are non-zero
-  // samples.
-  EXPECT_FALSE(IsAllZero(decoded, samples_per_channel * num_channels));
+  for (int n = 0; n <= algorithmic_frame_delay + 10; ++n) {
+    ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
+                                      receive_timestamp));
+    ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
+                                  &samples_per_channel, &num_channels,
+                                  &output_type));
+    if (n >= algorithmic_frame_delay + 1) {
+      // Expect that this frame contain samples from regular RTP.
+      EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
+    }
+    rtp_info.header.sequenceNumber++;
+    rtp_info.header.timestamp += kBlockSize16kHz;
+    receive_timestamp += kBlockSize16kHz;
+  }
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
   // Expecting a "clean" network.
   EXPECT_EQ(0, network_stats.packet_loss_rate);
   EXPECT_EQ(0, network_stats.expand_rate);
   EXPECT_EQ(0, network_stats.accelerate_rate);
-  EXPECT_EQ(0, network_stats.preemptive_rate);
+  EXPECT_LE(network_stats.preemptive_rate, 150);
 }
 
 // Test if the size of the packet buffer reported correctly when containing
@@ -1199,7 +1214,8 @@ TEST_F(NetEqDecodingTest,
   int num_channels;
   int samples_per_channel;
   uint32_t receive_timestamp = 0;
-  for (int n = 0; n < 1; ++n) {
+  int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
+  for (int n = 0; n < algorithmic_frame_delay; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
                                       receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
@@ -1225,7 +1241,8 @@ TEST_F(NetEqDecodingTest,
   }
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
-  EXPECT_EQ(kNumSyncPackets * 10, network_stats.current_buffer_size_ms);
+  EXPECT_EQ(kNumSyncPackets * 10 + algorithmic_delay_ms_,
+            network_stats.current_buffer_size_ms);
 
   // Rewind |rtp_info| to that of the first sync packet.
   memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info));
@@ -1298,7 +1315,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
       if (packets_inserted > 4) {
         // Expect preferred and actual buffer size to be no more than 2 frames.
         EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
-        EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2);
+        EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 +
+                  algorithmic_delay_ms_);
       }
       last_seq_no = seq_no;
       last_timestamp = timestamp;
@@ -1362,6 +1380,8 @@ void NetEqDecodingTest::DuplicateCng() {
   const int kSamples = kFrameSizeMs * kSampleRateKhz;
   const int kPayloadBytes = kSamples * 2;
 
+  const int algorithmic_delay_samples = std::max(
+      algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
   // Insert three speech packet. Three are needed to get the frame length
   // correct.
   int out_len;
@@ -1398,7 +1418,7 @@ void NetEqDecodingTest::DuplicateCng() {
                 kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
   ASSERT_EQ(kBlockSize16kHz, out_len);
   EXPECT_EQ(kOutputCNG, type);
-  EXPECT_EQ(timestamp - 10, neteq_->PlayoutTimestamp());
+  EXPECT_EQ(timestamp - algorithmic_delay_samples, neteq_->PlayoutTimestamp());
 
   // Insert the same CNG packet again. Note that at this point it is old, since
   // we have already decoded the first copy of it.
@@ -1412,7 +1432,8 @@ void NetEqDecodingTest::DuplicateCng() {
                   kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
     EXPECT_EQ(kOutputCNG, type);
-    EXPECT_EQ(timestamp - 10, neteq_->PlayoutTimestamp());
+    EXPECT_EQ(timestamp - algorithmic_delay_samples,
+              neteq_->PlayoutTimestamp());
   }
 
   // Insert speech again.
@@ -1427,7 +1448,8 @@ void NetEqDecodingTest::DuplicateCng() {
                 kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
   ASSERT_EQ(kBlockSize16kHz, out_len);
   EXPECT_EQ(kOutputNormal, type);
-  EXPECT_EQ(timestamp + kSamples - 10, neteq_->PlayoutTimestamp());
+  EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
+            neteq_->PlayoutTimestamp());
 }
 
 TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
diff --git a/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc b/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc
index c7ce31040..dace45eb9 100644
--- a/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc
+++ b/webrtc/modules/audio_coding/neteq4/preemptive_expand.cc
@@ -101,8 +101,10 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
 PreemptiveExpand* PreemptiveExpandFactory::Create(
     int sample_rate_hz,
     size_t num_channels,
-    const BackgroundNoise& background_noise) const {
-  return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise);
+    const BackgroundNoise& background_noise,
+    int overlap_samples) const {
+  return new PreemptiveExpand(
+      sample_rate_hz, num_channels, background_noise, overlap_samples);
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/preemptive_expand.h b/webrtc/modules/audio_coding/neteq4/preemptive_expand.h
index 241425e81..983f64d54 100644
--- a/webrtc/modules/audio_coding/neteq4/preemptive_expand.h
+++ b/webrtc/modules/audio_coding/neteq4/preemptive_expand.h
@@ -29,11 +29,13 @@ class BackgroundNoise;
 // PreemptiveExpand are implemented.
 class PreemptiveExpand : public TimeStretch {
  public:
-  PreemptiveExpand(int sample_rate_hz, size_t num_channels,
-                   const BackgroundNoise& background_noise)
+  PreemptiveExpand(int sample_rate_hz,
+                   size_t num_channels,
+                   const BackgroundNoise& background_noise,
+                   int overlap_samples)
       : TimeStretch(sample_rate_hz, num_channels, background_noise),
         old_data_length_per_channel_(-1),
-        overlap_samples_(5 * sample_rate_hz / 8000) {
+        overlap_samples_(overlap_samples) {
   }
 
   virtual ~PreemptiveExpand() {}
@@ -77,7 +79,8 @@ struct PreemptiveExpandFactory {
   virtual PreemptiveExpand* Create(
       int sample_rate_hz,
       size_t num_channels,
-      const BackgroundNoise& background_noise) const;
+      const BackgroundNoise& background_noise,
+      int overlap_samples) const;
 };
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq4/sync_buffer.h b/webrtc/modules/audio_coding/neteq4/sync_buffer.h
index e1e5daf1b..0659f199d 100644
--- a/webrtc/modules/audio_coding/neteq4/sync_buffer.h
+++ b/webrtc/modules/audio_coding/neteq4/sync_buffer.h
@@ -78,7 +78,8 @@ class SyncBuffer : public AudioMultiVector {
   // created.
   void Flush();
 
-  const AudioVector& Channel(size_t n) { return *channels_[n]; }
+  const AudioVector& Channel(size_t n) const { return *channels_[n]; }
+  AudioVector& Channel(size_t n) { return *channels_[n]; }
 
   // Accessors and mutators.
   size_t next_index() const { return next_index_; }
diff --git a/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc b/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc
index 188c18b71..529038ac3 100644
--- a/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq4/time_stretch_unittest.cc
@@ -21,14 +21,17 @@ namespace webrtc {
 TEST(TimeStretch, CreateAndDestroy) {
   const int kSampleRate = 8000;
   const size_t kNumChannels = 1;
+  const int kOverlapSamples = 5 * kSampleRate / 8000;
   BackgroundNoise bgn(kNumChannels);
   Accelerate accelerate(kSampleRate, kNumChannels, bgn);
-  PreemptiveExpand preemptive_expand(kSampleRate, kNumChannels, bgn);
+  PreemptiveExpand preemptive_expand(
+      kSampleRate, kNumChannels, bgn, kOverlapSamples);
 }
 
 TEST(TimeStretch, CreateUsingFactory) {
   const int kSampleRate = 8000;
   const size_t kNumChannels = 1;
+  const int kOverlapSamples = 5 * kSampleRate / 8000;
   BackgroundNoise bgn(kNumChannels);
 
   AccelerateFactory accelerate_factory;
@@ -38,8 +41,8 @@ TEST(TimeStretch, CreateUsingFactory) {
   delete accelerate;
 
   PreemptiveExpandFactory preemptive_expand_factory;
-  PreemptiveExpand* preemptive_expand =
-      preemptive_expand_factory.Create(kSampleRate, kNumChannels, bgn);
+  PreemptiveExpand* preemptive_expand = preemptive_expand_factory.Create(
+      kSampleRate, kNumChannels, bgn, kOverlapSamples);
   EXPECT_TRUE(preemptive_expand != NULL);
   delete preemptive_expand;
 }