Remove no-op and unused methods from AudioCodingModule

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 13 matching lines @@
 
 namespace webrtc {
 
 class CriticalSectionWrapper;
 class AudioCodingImpl;
 
 namespace acm2 {
 
 class ACMDTMFDetection;
 
-class AudioCodingModuleImpl : public AudioCodingModule {
+class AudioCodingModuleImpl final : public AudioCodingModule {
  public:
   friend webrtc::AudioCodingImpl;
 
   explicit AudioCodingModuleImpl(const AudioCodingModule::Config& config);
   ~AudioCodingModuleImpl() override;
 
   /////////////////////////////////////////
   //   Sender
   //
 
-  // Reset send codec.
-  int ResetEncoder() override;
-
   // Can be called multiple times for Codec, CNG, RED.
   int RegisterSendCodec(const CodecInst& send_codec) override;
 
   void RegisterExternalSendCodec(
       AudioEncoderMutable* external_speech_encoder) override;
 
   // Get current send codec.
   int SendCodec(CodecInst* current_codec) const override;
 
   // Get current send frequency.
   int SendFrequency() const override;
 
-  // Get encode bit-rate.
-  // Adaptive rate codecs return their current encode target rate, while other
-  // codecs return there long-term average or their fixed rate.
-  int SendBitrate() const override;
-
   // Sets the bitrate to the specified value in bits/sec. In case the codec does
   // not support the requested value it will choose an appropriate value
   // instead.
   void SetBitRate(int bitrate_bps) override;
 
-  // Set available bandwidth, inform the encoder about the
-  // estimated bandwidth received from the remote party.
-  int SetReceivedEstimatedBandwidth(int bw) override;
-
   // Register a transport callback which will be
   // called to deliver the encoded buffers.
   int RegisterTransportCallback(AudioPacketizationCallback* transport) override;
 
   // Add 10 ms of raw (PCM) audio data to the encoder.
   int Add10MsData(const AudioFrame& audio_frame) override;
 
   /////////////////////////////////////////
   // (RED) Redundant Coding
   //
@@ skipping 33 matching lines @@
 
   int RegisterVADCallback(ACMVADCallback* vad_callback) override;
 
   /////////////////////////////////////////
   //   Receiver
   //
 
   // Initialize receiver, resets codec database etc.
   int InitializeReceiver() override;
 
-  // Reset the decoder state.
-  int ResetDecoder() override;
-
   // Get current receive frequency.
   int ReceiveFrequency() const override;
 
   // Get current playout frequency.
   int PlayoutFrequency() const override;
 
   // Register possible receive codecs, can be called multiple times,
   // for codecs, CNG, DTMF, RED.
   int RegisterReceiveCodec(const CodecInst& receive_codec) override;
 
@@ skipping 33 matching lines @@
   // TODO(turajs): DTMF playout is always activated in NetEq these APIs should
   // be removed, as well as all VoE related APIs and methods.
   //
   // Configure Dtmf playout status i.e on/off playout the incoming outband Dtmf
   // tone.
   int SetDtmfPlayoutStatus(bool enable) override;
 
   // Get Dtmf playout status.
   bool DtmfPlayoutStatus() const override;
 
-  // Estimate the Bandwidth based on the incoming stream, needed
-  // for one way audio where the RTCP send the BW estimate.
-  // This is also done in the RTP module .
-  int DecoderEstimatedBandwidth() const override;
-
   // Set playout mode voice, fax.
   int SetPlayoutMode(AudioPlayoutMode mode) override;
 
   // Get playout mode voice, fax.
   AudioPlayoutMode PlayoutMode() const override;
 
   // Get playout timestamp.
   int PlayoutTimestamp(uint32_t* timestamp) override;
 
   // Get 10 milliseconds of raw audio data to play out, and
   // automatic resample to the requested frequency if > 0.
   int PlayoutData10Ms(int desired_freq_hz, AudioFrame* audio_frame) override;
 
   /////////////////////////////////////////
   //   Statistics
   //
 
   int GetNetworkStatistics(NetworkStatistics* statistics) override;
 
-  // GET RED payload for iSAC. The method id called when 'this' ACM is
-  // the default ACM.
-  // TODO(henrik.lundin) Not used. Remove?
-  int REDPayloadISAC(int isac_rate,
-                     int isac_bw_estimate,
-                     uint8_t* payload,
-                     int16_t* length_bytes);
-
-  int ReplaceInternalDTXWithWebRtc(bool use_webrtc_dtx) override;
-
-  int IsInternalDTXReplacedWithWebRtc(bool* uses_webrtc_dtx) override;
-
   int SetISACMaxRate(int max_bit_per_sec) override;
 
   int SetISACMaxPayloadSize(int max_size_bytes) override;
 
-  int ConfigISACBandwidthEstimator(int frame_size_ms,
-                                   int rate_bit_per_sec,
-                                   bool enforce_frame_size = false) override;
-
   int SetOpusApplication(OpusApplicationMode application) override;
 
   // If current send codec is Opus, informs it about the maximum playback rate
   // the receiver will render.
   int SetOpusMaxPlaybackRate(int frequency_hz) override;
 
   int EnableOpusDtx() override;
 
   int DisableOpusDtx() override;
 
@@ skipping 168 matching lines @@
   int playout_frequency_hz_;
   // TODO(henrik.lundin): All members below this line are temporary and should
   // be removed after refactoring is completed.
   rtc::scoped_ptr<acm2::AudioCodingModuleImpl> acm_old_;
   CodecInst current_send_codec_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_AUDIO_CODING_MODULE_IMPL_H_