Change return type of AudioEncoder::SetMaxPlaybackRate to void

webrtc/modules/audio_coding/codecs/cng/include/audio_encoder_cng.h

 /*
  *  Copyright (c) 2014 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 46 matching lines @@
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
   void Reset() override;
   bool SetFec(bool enable) override;
   bool SetDtx(bool enable) override;
   bool SetApplication(Application application) override;
-  bool SetMaxPlaybackRate(int frequency_hz) override;
+  void SetMaxPlaybackRate(int frequency_hz) override;
   void SetProjectedPacketLossRate(double fraction) override;
   void SetTargetBitrate(int target_bps) override;
   void SetMaxBitrate(int max_bps) override;
   void SetMaxPayloadSize(int max_payload_size_bytes) override;
 
  private:
   EncodedInfo EncodePassive(size_t frames_to_encode,
                             size_t max_encoded_bytes,
                             uint8_t* encoded);
   EncodedInfo EncodeActive(size_t frames_to_encode,
                            size_t max_encoded_bytes,
                            uint8_t* encoded);
   size_t SamplesPer10msFrame() const;
 
   AudioEncoder* speech_encoder_;
   const int cng_payload_type_;
   const int num_cng_coefficients_;
   const int sid_frame_interval_ms_;
   std::vector<int16_t> speech_buffer_;
   std::vector<uint32_t> rtp_timestamps_;
   bool last_frame_active_;
   rtc::scoped_ptr<Vad> vad_;
   rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_INCLUDE_AUDIO_ENCODER_CNG_H_