Fix clang style warnings in webrtc/modules/audio_coding/neteq

Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.

BUG=163
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44109004

Cr-Commit-Position: refs/heads/master@{#8960}

webrtc/modules/audio_coding/BUILD.gn

@@ -757,12 +757,6 @@ source_set("neteq") {
     ":neteq_config",
   ]
 
-  if (is_clang) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-
   deps = [
     ":audio_decoder_interface",
     ":cng",

webrtc/modules/audio_coding/neteq/accelerate.h

@@ -34,8 +34,6 @@ class Accelerate : public TimeStretch {
       : TimeStretch(sample_rate_hz, num_channels, background_noise) {
   }
 
-  virtual ~Accelerate() {}
-
   // This method performs the actual Accelerate operation. The samples are
   // read from |input|, of length |input_length| elements, and are written to
   // |output|. The number of samples removed through time-stretching is

webrtc/modules/audio_coding/neteq/audio_classifier.cc

@@ -68,4 +68,8 @@ bool AudioClassifier::Analysis(const int16_t* input,
   return is_music_;
 }
 
+bool AudioClassifier::is_music() const {
+  return is_music_;
+}
+
 }  // namespace webrtc

webrtc/modules/audio_coding/neteq/audio_classifier.h

@@ -37,7 +37,7 @@ class AudioClassifier {
   bool Analysis(const int16_t* input, int input_length, int channels);
 
   // Gets the current classification : true = music, false = speech.
-  virtual bool is_music() const { return is_music_; }
+  virtual bool is_music() const;
 
   // Gets the current music probability.
   float music_probability() const { return music_probability_; }

webrtc/modules/audio_coding/neteq/audio_multi_vector.cc

@@ -183,6 +183,10 @@ void AudioMultiVector::CrossFade(const AudioMultiVector& append_this,
   }
 }
 
+size_t AudioMultiVector::Channels() const {
+  return num_channels_;
+}
+
 size_t AudioMultiVector::Size() const {
   assert(channels_[0]);
   return channels_[0]->Size();

webrtc/modules/audio_coding/neteq/audio_multi_vector.h

@@ -106,7 +106,7 @@ class AudioMultiVector {
                          size_t fade_length);
 
   // Returns the number of channels.
-  virtual size_t Channels() const { return num_channels_; }
+  virtual size_t Channels() const;
 
   // Returns the number of elements per channel in this AudioMultiVector.
   virtual size_t Size() const;

webrtc/modules/audio_coding/neteq/audio_vector.cc

@@ -18,6 +18,21 @@
 
 namespace webrtc {
 
+AudioVector::AudioVector()
+    : array_(new int16_t[kDefaultInitialSize]),
+      first_free_ix_(0),
+      capacity_(kDefaultInitialSize) {
+}
+
+AudioVector::AudioVector(size_t initial_size)
+    : array_(new int16_t[initial_size]),
+      first_free_ix_(initial_size),
+      capacity_(initial_size) {
+  memset(array_.get(), 0, initial_size * sizeof(int16_t));
+}
+
+AudioVector::~AudioVector() = default;
+
 void AudioVector::Clear() {
   first_free_ix_ = 0;
 }
@@ -145,6 +160,16 @@ void AudioVector::CrossFade(const AudioVector& append_this,
     PushBack(&append_this[fade_length], samples_to_push_back);
 }
 
+// Returns the number of elements in this AudioVector.
+size_t AudioVector::Size() const {
+  return first_free_ix_;
+}
+
+// Returns true if this AudioVector is empty.
+bool AudioVector::Empty() const {
+  return first_free_ix_ == 0;
+}
+
 const int16_t& AudioVector::operator[](size_t index) const {
   return array_[index];
 }

webrtc/modules/audio_coding/neteq/audio_vector.h

@@ -22,20 +22,12 @@ namespace webrtc {
 class AudioVector {
  public:
   // Creates an empty AudioVector.
-  AudioVector()
-      : array_(new int16_t[kDefaultInitialSize]),
-        first_free_ix_(0),
-        capacity_(kDefaultInitialSize) {}
+  AudioVector();
 
   // Creates an AudioVector with an initial size.
-  explicit AudioVector(size_t initial_size)
-      : array_(new int16_t[initial_size]),
-        first_free_ix_(initial_size),
-        capacity_(initial_size) {
-    memset(array_.get(), 0, initial_size * sizeof(int16_t));
-  }
+  explicit AudioVector(size_t initial_size);
 
-  virtual ~AudioVector() {}
+  virtual ~AudioVector();
 
   // Deletes all values and make the vector empty.
   virtual void Clear();
@@ -94,10 +86,10 @@ class AudioVector {
   virtual void CrossFade(const AudioVector& append_this, size_t fade_length);
 
   // Returns the number of elements in this AudioVector.
-  virtual size_t Size() const { return first_free_ix_; }
+  virtual size_t Size() const;
 
   // Returns true if this AudioVector is empty.
-  virtual bool Empty() const { return (first_free_ix_ == 0); }
+  virtual bool Empty() const;
 
   // Accesses and modifies an element of AudioVector.
   const int16_t& operator[](size_t index) const;

webrtc/modules/audio_coding/neteq/buffer_level_filter.cc

@@ -57,4 +57,9 @@ void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level) {
     level_factor_ = 254;
   }
 }
+
+int BufferLevelFilter::filtered_current_level() const {
+  return filtered_current_level_;
+}
+
 }  // namespace webrtc

webrtc/modules/audio_coding/neteq/buffer_level_filter.h

@@ -34,7 +34,7 @@ class BufferLevelFilter {
   // filter coefficient.
   virtual void SetTargetBufferLevel(int target_buffer_level);
 
-  virtual int filtered_current_level() const { return filtered_current_level_; }
+  virtual int filtered_current_level() const;
 
  private:
   int level_factor_;  // Filter factor for the buffer level filter in Q8.

webrtc/modules/audio_coding/neteq/decision_logic_fax.h

@@ -34,9 +34,6 @@ class DecisionLogicFax : public DecisionLogic {
                       buffer_level_filter) {
   }
 
-  // Destructor.
-  virtual ~DecisionLogicFax() {}
-
  protected:
   // Returns the operation that should be done next. |sync_buffer| and |expand|
   // are provided for reference. |decoder_frame_length| is the number of samples

webrtc/modules/audio_coding/neteq/decision_logic_normal.h

@@ -34,9 +34,6 @@ class DecisionLogicNormal : public DecisionLogic {
                       buffer_level_filter) {
   }
 
-  // Destructor.
-  virtual ~DecisionLogicNormal() {}
-
  protected:
   static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
   static const int kReinitAfterExpands = 100;
@@ -51,12 +48,13 @@ class DecisionLogicNormal : public DecisionLogic {
   // should be set to true. The output variable |reset_decoder| will be set to
   // true if a reset is required; otherwise it is left unchanged (i.e., it can
   // remain true if it was true before the call).
-  virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
-                                            const Expand& expand,
-                                            int decoder_frame_length,
-                                            const RTPHeader* packet_header,
-                                            Modes prev_mode, bool play_dtmf,
-                                            bool* reset_decoder);
+  Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
+                                    const Expand& expand,
+                                    int decoder_frame_length,
+                                    const RTPHeader* packet_header,
+                                    Modes prev_mode,
+                                    bool play_dtmf,
+                                    bool* reset_decoder) override;
 
   // Returns the operation to do given that the expected packet is not
   // available, but a packet further into the future is at hand.

webrtc/modules/audio_coding/neteq/delay_peak_detector.cc

@@ -21,6 +21,8 @@ namespace webrtc {
 // peak-mode is engaged and the DelayManager asks the DelayPeakDetector for
 // the worst peak height.
 
+DelayPeakDetector::~DelayPeakDetector() = default;
+
 DelayPeakDetector::DelayPeakDetector()
   : peak_found_(false),
     peak_detection_threshold_(0),
@@ -40,6 +42,10 @@ void DelayPeakDetector::SetPacketAudioLength(int length_ms) {
   }
 }
 
+bool DelayPeakDetector::peak_found() {
+  return peak_found_;
+}
+
 int DelayPeakDetector::MaxPeakHeight() const {
   int max_height = -1;  // Returns -1 for an empty history.
   std::list<Peak>::const_iterator it;

webrtc/modules/audio_coding/neteq/delay_peak_detector.h

@@ -22,7 +22,7 @@ namespace webrtc {
 class DelayPeakDetector {
  public:
   DelayPeakDetector();
-  virtual ~DelayPeakDetector() {}
+  virtual ~DelayPeakDetector();
   virtual void Reset();
 
   // Notifies the DelayPeakDetector of how much audio data is carried in each
@@ -31,7 +31,7 @@ class DelayPeakDetector {
 
   // Returns true if peak-mode is active. That is, delay peaks were observed
   // recently.
-  virtual bool peak_found() { return peak_found_; }
+  virtual bool peak_found();
 
   // Calculates and returns the maximum delay peak height. Returns -1 if no
   // delay peaks have been observed recently. The unit is number of packets.

webrtc/modules/audio_coding/neteq/dtmf_buffer.cc

@@ -20,6 +20,16 @@
 
 namespace webrtc {
 
+DtmfBuffer::DtmfBuffer(int fs_hz) {
+  SetSampleRate(fs_hz);
+}
+
+DtmfBuffer::~DtmfBuffer() = default;
+
+void DtmfBuffer::Flush() {
+  buffer_.clear();
+}
+
 // The ParseEvent method parses 4 bytes from |payload| according to this format
 // from RFC 4733:
 //
@@ -173,6 +183,14 @@ bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
   return false;
 }
 
+size_t DtmfBuffer::Length() const {
+  return buffer_.size();
+}
+
+bool DtmfBuffer::Empty() const {
+  return buffer_.empty();
+}
+
 int DtmfBuffer::SetSampleRate(int fs_hz) {
   if (fs_hz != 8000 &&
       fs_hz != 16000 &&

webrtc/modules/audio_coding/neteq/dtmf_buffer.h

@@ -55,14 +55,12 @@ class DtmfBuffer {
   };
 
   // Set up the buffer for use at sample rate |fs_hz|.
-  explicit DtmfBuffer(int fs_hz) {
-    SetSampleRate(fs_hz);
-  }
+  explicit DtmfBuffer(int fs_hz);
 
-  virtual ~DtmfBuffer() {}
+  virtual ~DtmfBuffer();
 
   // Flushes the buffer.
-  virtual void Flush() { buffer_.clear(); }
+  virtual void Flush();
 
   // Static method to parse 4 bytes from |payload| as a DTMF event (RFC 4733)
   // and write the parsed information into the struct |event|. Input variable
@@ -82,9 +80,9 @@ class DtmfBuffer {
   virtual bool GetEvent(uint32_t current_timestamp, DtmfEvent* event);
 
   // Number of events in the buffer.
-  virtual size_t Length() const { return buffer_.size(); }
+  virtual size_t Length() const;
 
-  virtual bool Empty() const { return buffer_.empty(); }
+  virtual bool Empty() const;
 
   // Set a new sample rate.
   virtual int SetSampleRate(int fs_hz);

webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc

@@ -189,4 +189,8 @@ int DtmfToneGenerator::Generate(int num_samples,
   return num_samples;
 }
 
+bool DtmfToneGenerator::initialized() const {
+  return initialized_;
+}
+
 }  // namespace webrtc

webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h

@@ -31,7 +31,7 @@ class DtmfToneGenerator {
   virtual int Init(int fs, int event, int attenuation);
   virtual void Reset();
   virtual int Generate(int num_samples, AudioMultiVector* output);
-  virtual bool initialized() const { return initialized_; }
+  virtual bool initialized() const;
 
  private:
   static const int kCoeff1[4][16];  // 1st oscillator model coefficient table.

webrtc/modules/audio_coding/neteq/expand.cc

@@ -24,6 +24,32 @@
 
 namespace webrtc {
 
+Expand::Expand(BackgroundNoise* background_noise,
+               SyncBuffer* sync_buffer,
+               RandomVector* random_vector,
+               int fs,
+               size_t num_channels)
+    : random_vector_(random_vector),
+      sync_buffer_(sync_buffer),
+      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),
+      stop_muting_(false),
+      channel_parameters_(new ChannelParameters[num_channels_]) {
+  assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
+  assert(fs <= kMaxSampleRate);  // Should not be possible.
+  assert(num_channels_ > 0);
+  memset(expand_lags_, 0, sizeof(expand_lags_));
+  Reset();
+}
+
+Expand::~Expand() = default;
+
 void Expand::Reset() {
   first_expand_ = true;
   consecutive_expands_ = 0;
@@ -289,6 +315,10 @@ void Expand::SetParametersForMergeAfterExpand() {
   stop_muting_ = true;
 }
 
+size_t Expand::overlap_length() const {
+  return overlap_length_;
+}
+
 void Expand::InitializeForAnExpandPeriod() {
   lag_index_direction_ = 1;
   current_lag_index_ = -1;
@@ -712,6 +742,18 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
   }
 }
 
+Expand::ChannelParameters::ChannelParameters()
+    : mute_factor(16384),
+      ar_gain(0),
+      ar_gain_scale(0),
+      voice_mix_factor(0),
+      current_voice_mix_factor(0),
+      onset(false),
+      mute_slope(0) {
+  memset(ar_filter, 0, sizeof(ar_filter));
+  memset(ar_filter_state, 0, sizeof(ar_filter_state));
+}
+
 int16_t Expand::Correlation(const int16_t* input, size_t input_length,
                             int16_t* output, int16_t* output_scale) const {
   // Set parameters depending on sample rate.

webrtc/modules/audio_coding/neteq/expand.h

@@ -35,27 +35,9 @@ class Expand {
          SyncBuffer* sync_buffer,
          RandomVector* random_vector,
          int fs,
-         size_t num_channels)
-      : random_vector_(random_vector),
-        sync_buffer_(sync_buffer),
-        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),
-        stop_muting_(false),
-        channel_parameters_(new ChannelParameters[num_channels_]) {
-    assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
-    assert(fs <= kMaxSampleRate);  // Should not be possible.
-    assert(num_channels_ > 0);
-    memset(expand_lags_, 0, sizeof(expand_lags_));
-    Reset();
-  }
+         size_t num_channels);
 
-  virtual ~Expand() {}
+  virtual ~Expand();
 
   // Resets the object.
   virtual void Reset();
@@ -85,7 +67,7 @@ class Expand {
   }
 
   // Accessors and mutators.
-  virtual size_t overlap_length() const { return overlap_length_; }
+  virtual size_t overlap_length() const;
   int16_t max_lag() const { return max_lag_; }
 
  protected:
@@ -126,18 +108,7 @@ class Expand {
   static const int kNumLags = 3;
 
   struct ChannelParameters {
-    // Constructor.
-    ChannelParameters()
-        : mute_factor(16384),
-          ar_gain(0),
-          ar_gain_scale(0),
-          voice_mix_factor(0),
-          current_voice_mix_factor(0),
-          onset(false),
-          mute_slope(0) {
-      memset(ar_filter, 0, sizeof(ar_filter));
-      memset(ar_filter_state, 0, sizeof(ar_filter_state));
-    }
+    ChannelParameters();
     int16_t mute_factor;
     int16_t ar_filter[kUnvoicedLpcOrder + 1];
     int16_t ar_filter_state[kUnvoicedLpcOrder];

webrtc/modules/audio_coding/neteq/merge.cc

@@ -24,6 +24,20 @@
 
 namespace webrtc {
 
+Merge::Merge(int fs_hz,
+             size_t num_channels,
+             Expand* expand,
+             SyncBuffer* sync_buffer)
+    : fs_hz_(fs_hz),
+      num_channels_(num_channels),
+      fs_mult_(fs_hz_ / 8000),
+      timestamps_per_call_(fs_hz_ / 100),
+      expand_(expand),
+      sync_buffer_(sync_buffer),
+      expanded_(num_channels_) {
+  assert(num_channels_ > 0);
+}
+
 int Merge::Process(int16_t* input, size_t input_length,
                    int16_t* external_mute_factor_array,
                    AudioMultiVector* output) {

webrtc/modules/audio_coding/neteq/merge.h

@@ -33,17 +33,10 @@ class SyncBuffer;
 // what the Merge class does.
 class Merge {
  public:
-  Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
-      : fs_hz_(fs_hz),
-        num_channels_(num_channels),
-        fs_mult_(fs_hz_ / 8000),
-        timestamps_per_call_(fs_hz_ / 100),
-        expand_(expand),
-        sync_buffer_(sync_buffer),
-        expanded_(num_channels_) {
-    assert(num_channels_ > 0);
-  }
-
+  Merge(int fs_hz,
+        size_t num_channels,
+        Expand* expand,
+        SyncBuffer* sync_buffer);
   virtual ~Merge() {}
 
   // The main method to produce the audio data. The decoded data is supplied in

webrtc/modules/audio_coding/neteq/neteq_impl.cc

@@ -281,6 +281,18 @@ int NetEqImpl::LeastRequiredDelayMs() const {
   return delay_manager_->least_required_delay_ms();
 }
 
+int NetEqImpl::SetTargetDelay() {
+  return kNotImplemented;
+}
+
+int NetEqImpl::TargetDelay() {
+  return kNotImplemented;
+}
+
+int NetEqImpl::CurrentDelay() {
+  return kNotImplemented;
+}
+
 // Deprecated.
 // TODO(henrik.lundin) Delete.
 void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
@@ -354,6 +366,14 @@ bool NetEqImpl::GetPlayoutTimestamp(uint32_t* timestamp) {
   return true;
 }
 
+int NetEqImpl::SetTargetNumberOfChannels() {
+  return kNotImplemented;
+}
+
+int NetEqImpl::SetTargetSampleRate() {
+  return kNotImplemented;
+}
+
 int NetEqImpl::LastError() const {
   CriticalSectionScoped lock(crit_sect_.get());
   return error_code_;

webrtc/modules/audio_coding/neteq/neteq_impl.h

@@ -73,7 +73,7 @@ class NetEqImpl : public webrtc::NetEq {
             PreemptiveExpandFactory* preemptive_expand_factory,
             bool create_components = true);
 
-  virtual ~NetEqImpl();
+  ~NetEqImpl() override;
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
@@ -133,11 +133,11 @@ class NetEqImpl : public webrtc::NetEq {
 
   int LeastRequiredDelayMs() const override;
 
-  int SetTargetDelay() override { return kNotImplemented; }
+  int SetTargetDelay() override;
 
-  int TargetDelay() override { return kNotImplemented; }
+  int TargetDelay() override;
 
-  int CurrentDelay() override { return kNotImplemented; }
+  int CurrentDelay() override;
 
   // Sets the playout mode to |mode|.
   // Deprecated.
@@ -174,9 +174,9 @@ class NetEqImpl : public webrtc::NetEq {
 
   bool GetPlayoutTimestamp(uint32_t* timestamp) override;
 
-  int SetTargetNumberOfChannels() override { return kNotImplemented; }
+  int SetTargetNumberOfChannels() override;
 
-  int SetTargetSampleRate() override { return kNotImplemented; }
+  int SetTargetSampleRate() override;
 
   // Returns the error code for the last occurred error. If no error has
   // occurred, 0 is returned.

webrtc/modules/audio_coding/neteq/packet_buffer.cc

@@ -49,6 +49,10 @@ void PacketBuffer::Flush() {
   DeleteAllPackets(&buffer_);
 }
 
+bool PacketBuffer::Empty() const {
+  return buffer_.empty();
+}
+
 int PacketBuffer::InsertPacket(Packet* packet) {
   if (!packet || !packet->payload) {
     if (packet) {
@@ -229,6 +233,14 @@ int PacketBuffer::DiscardOldPackets(uint32_t timestamp_limit,
   return 0;
 }
 
+int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) {
+  return DiscardOldPackets(timestamp_limit, 0);
+}
+
+int PacketBuffer::NumPacketsInBuffer() const {
+  return static_cast<int>(buffer_.size());
+}
+
 int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
                                      int last_decoded_length) const {
   PacketList::const_iterator it;

webrtc/modules/audio_coding/neteq/packet_buffer.h

@@ -43,7 +43,7 @@ class PacketBuffer {
   virtual void Flush();
 
   // Returns true for an empty buffer.
-  virtual bool Empty() const { return buffer_.empty(); }
+  virtual bool Empty() const;
 
   // Inserts |packet| into the buffer. The buffer will take over ownership of
   // the packet object.
@@ -105,15 +105,11 @@ class PacketBuffer {
                                 uint32_t horizon_samples);
 
   // Discards all packets that are (strictly) older than timestamp_limit.
-  virtual int DiscardAllOldPackets(uint32_t timestamp_limit) {
-    return DiscardOldPackets(timestamp_limit, 0);
-  }
+  virtual int DiscardAllOldPackets(uint32_t timestamp_limit);
 
   // Returns the number of packets in the buffer, including duplicates and
   // redundant packets.
-  virtual int NumPacketsInBuffer() const {
-    return static_cast<int>(buffer_.size());
-  }
+  virtual int NumPacketsInBuffer() const;
 
   // Returns the number of samples in the buffer, including samples carried in
   // duplicate and redundant packets.

webrtc/modules/audio_coding/neteq/preemptive_expand.h

@@ -38,8 +38,6 @@ class PreemptiveExpand : public TimeStretch {
         overlap_samples_(overlap_samples) {
   }
 
-  virtual ~PreemptiveExpand() {}
-
   // This method performs the actual PreemptiveExpand operation. The samples are
   // read from |input|, of length |input_length| elements, and are written to
   // |output|. The number of samples added through time-stretching is
@@ -54,16 +52,18 @@ class PreemptiveExpand : public TimeStretch {
  protected:
   // Sets the parameters |best_correlation| and |peak_index| to suitable
   // values when the signal contains no active speech.
-  virtual void SetParametersForPassiveSpeech(size_t len,
-                                             int16_t* w16_bestCorr,
-                                             int* w16_bestIndex) const;
+  void SetParametersForPassiveSpeech(size_t len,
+                                     int16_t* w16_bestCorr,
+                                     int* w16_bestIndex) const override;
 
   // Checks the criteria for performing the time-stretching operation and,
   // if possible, performs the time-stretching.
-  virtual ReturnCodes CheckCriteriaAndStretch(
-      const int16_t *pw16_decoded, size_t len, size_t w16_bestIndex,
-      int16_t w16_bestCorr, bool w16_VAD,
-      AudioMultiVector* output) const;
+  ReturnCodes CheckCriteriaAndStretch(const int16_t* pw16_decoded,
+                                      size_t len,
+                                      size_t w16_bestIndex,
+                                      int16_t w16_bestCorr,
+                                      bool w16_VAD,
+                                      AudioMultiVector* output) const override;
 
  private:
   int old_data_length_per_channel_;

webrtc/modules/audio_coding/neteq/sync_buffer.h

@@ -25,8 +25,6 @@ class SyncBuffer : public AudioMultiVector {
         end_timestamp_(0),
         dtmf_index_(0) {}
 
-  virtual ~SyncBuffer() {}
-
   // Returns the number of samples yet to play out form the buffer.
   size_t FutureLength() const;
 
@@ -34,7 +32,7 @@ class SyncBuffer : public AudioMultiVector {
   // the same number of samples from the beginning of the SyncBuffer, to
   // maintain a constant buffer size. The |next_index_| is updated to reflect
   // the move of the beginning of "future" data.
-  void PushBack(const AudioMultiVector& append_this);
+  void PushBack(const AudioMultiVector& append_this) override;
 
   // Adds |length| zeros to the beginning of each channel. Removes
   // the same number of samples from the end of the SyncBuffer, to

webrtc/modules/audio_coding/neteq/timestamp_scaler.cc

@@ -16,6 +16,10 @@
 
 namespace webrtc {
 
+void TimestampScaler::Reset() {
+  first_packet_received_ = false;
+}
+
 void TimestampScaler::ToInternal(Packet* packet) {
   if (!packet) {
     return;

webrtc/modules/audio_coding/neteq/timestamp_scaler.h

@@ -36,7 +36,7 @@ class TimestampScaler {
   virtual ~TimestampScaler() {}
 
   // Start over.
-  virtual void Reset() { first_packet_received_ = false; }
+  virtual void Reset();
 
   // Scale the timestamp in |packet| from external to internal.
   virtual void ToInternal(Packet* packet);