diff --git a/webrtc/modules/audio_coding/neteq/interface/neteq.h b/webrtc/modules/audio_coding/neteq/interface/neteq.h
index 7196bc196f..925cb231c4 100644
--- a/webrtc/modules/audio_coding/neteq/interface/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/interface/neteq.h
@@ -74,13 +74,15 @@ class NetEq {
           max_packets_in_buffer(50),
           // |max_delay_ms| has the same effect as calling SetMaximumDelay().
           max_delay_ms(2000),
-          background_noise_mode(kBgnOff) {}
+          background_noise_mode(kBgnOff),
+          playout_mode(kPlayoutOn) {}
 
     int sample_rate_hz;  // Initial vale. Will change with input data.
     bool enable_audio_classifier;
     int max_packets_in_buffer;
     int max_delay_ms;
     BackgroundNoiseMode background_noise_mode;
+    NetEqPlayoutMode playout_mode;
   };
 
   enum ReturnCodes {
@@ -202,9 +204,13 @@ class NetEq {
   virtual int CurrentDelay() = 0;
 
   // Sets the playout mode to |mode|.
+  // Deprecated. Set the mode in the Config struct passed to the constructor.
+  // TODO(henrik.lundin) Delete.
   virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0;
 
   // Returns the current playout mode.
+  // Deprecated.
+  // TODO(henrik.lundin) Delete.
   virtual NetEqPlayoutMode PlayoutMode() const = 0;
 
   // Writes the current network statistics to |stats|. The statistics are reset
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index d714733534..edf618efc6 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -91,6 +91,7 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
       error_code_(0),
       decoder_error_code_(0),
       background_noise_mode_(config.background_noise_mode),
+      playout_mode_(config.playout_mode),
       decoded_packet_sequence_number_(-1),
       decoded_packet_timestamp_(0) {
   int fs = config.sample_rate_hz;
@@ -278,18 +279,21 @@ int NetEqImpl::LeastRequiredDelayMs() const {
   return delay_manager_->least_required_delay_ms();
 }
 
+// Deprecated.
+// TODO(henrik.lundin) Delete.
 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.
-    CreateDecisionLogic(mode);
+  if (mode != playout_mode_) {
+    playout_mode_ = mode;
+    CreateDecisionLogic();
   }
 }
 
+// Deprecated.
+// TODO(henrik.lundin) Delete.
 NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
   CriticalSectionScoped lock(crit_sect_.get());
-  assert(decision_logic_.get());
-  return decision_logic_->playout_mode();
+  return playout_mode_;
 }
 
 int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
@@ -1904,7 +1908,7 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
   // 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);
+    CreateDecisionLogic();
   }
   decision_logic_->SetSampleRate(fs_hz_, output_size_samples_);
 }
@@ -1926,9 +1930,9 @@ NetEqOutputType NetEqImpl::LastOutputType() {
   }
 }
 
-void NetEqImpl::CreateDecisionLogic(NetEqPlayoutMode mode) {
+void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
-                                              mode,
+                                              playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index cfe6a402fe..fc2284d981 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -138,9 +138,13 @@ class NetEqImpl : public webrtc::NetEq {
   virtual int CurrentDelay() OVERRIDE { return kNotImplemented; }
 
   // Sets the playout mode to |mode|.
+  // Deprecated.
+  // TODO(henrik.lundin) Delete.
   virtual void SetPlayoutMode(NetEqPlayoutMode mode) OVERRIDE;
 
   // Returns the current playout mode.
+  // Deprecated.
+  // TODO(henrik.lundin) Delete.
   virtual NetEqPlayoutMode PlayoutMode() const OVERRIDE;
 
   // Writes the current network statistics to |stats|. The statistics are reset
@@ -327,9 +331,8 @@ class NetEqImpl : public webrtc::NetEq {
   virtual void UpdatePlcComponents(int fs_hz, size_t channels)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  // Creates DecisionLogic object for the given mode.
-  virtual void CreateDecisionLogic(NetEqPlayoutMode mode)
-      EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+  // Creates DecisionLogic object with the mode given by |playout_mode_|.
+  virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   const scoped_ptr<CriticalSectionWrapper> crit_sect_;
   const scoped_ptr<BufferLevelFilter> buffer_level_filter_
@@ -383,6 +386,7 @@ class NetEqImpl : public webrtc::NetEq {
   int error_code_ GUARDED_BY(crit_sect_);  // Store last error code.
   int decoder_error_code_ GUARDED_BY(crit_sect_);
   const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
+  NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
 
   // These values are used by NACK module to estimate time-to-play of
   // a missing packet. Occasionally, NetEq might decide to decode more
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 256c557d60..4a7dbecfe7 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -442,12 +442,16 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) {
   }
 }
 
-// TODO(hlundin): Re-enable test once the statistics interface is up and again.
-TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
-  // Use fax mode to avoid time-scaling. This is to simplify the testing of
-  // packet waiting times in the packet buffer.
-  neteq_->SetPlayoutMode(kPlayoutFax);
-  ASSERT_EQ(kPlayoutFax, neteq_->PlayoutMode());
+// Use fax mode to avoid time-scaling. This is to simplify the testing of
+// packet waiting times in the packet buffer.
+class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
+ protected:
+  NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
+    config_.playout_mode = kPlayoutFax;
+  }
+};
+
+TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
   // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
   size_t num_frames = 30;
   const int kSamples = 10 * 16;