Remove VoECodec

voice_engine/channel_proxy.h

 /*
  *  Copyright (c) 2015 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 75 matching lines @@
   virtual int GetSpeechOutputLevelFullRange() const;
   // See description of "totalAudioEnergy" in the WebRTC stats spec:
   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
   virtual double GetTotalOutputEnergy() const;
   virtual double GetTotalOutputDuration() const;
   virtual uint32_t GetDelayEstimate() const;
   virtual bool SetSendTelephoneEventPayloadType(int payload_type,
                                                 int payload_frequency);
   virtual bool SendTelephoneEventOutband(int event, int duration_ms);
   virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
-  virtual void SetRecPayloadType(int payload_type,
-                                 const SdpAudioFormat& format);
   virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
   virtual void RegisterExternalTransport(Transport* transport);
   virtual void DeRegisterExternalTransport();
 
   // Implements RtpPacketSinkInterface
   void OnRtpPacket(const RtpPacketReceived& packet) override;
   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
   virtual const rtc::scoped_refptr<AudioDecoderFactory>&
       GetAudioDecoderFactory() const;
   virtual void SetChannelOutputVolumeScaling(float scaling);
   virtual void SetRtcEventLog(RtcEventLog* event_log);
   virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
       int sample_rate_hz,
       AudioFrame* audio_frame);
   virtual int NeededFrequency() const;
   virtual void SetTransportOverhead(int transport_overhead_per_packet);
   virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
   virtual void DisassociateSendChannel();
   virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
                           RtpReceiver** rtp_receiver) const;
   virtual uint32_t GetPlayoutTimestamp() const;
   virtual void SetMinimumPlayoutDelay(int delay_ms);
   virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
   virtual bool GetRecCodec(CodecInst* codec_inst) const;
   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
   virtual void OnRecoverableUplinkPacketLossRate(
       float recoverable_packet_loss_rate);
-  virtual void RegisterLegacyReceiveCodecs();
   virtual std::vector<webrtc::RtpSource> GetSources() const;
 
  private:
   Channel* channel() const;
 
   // Thread checkers document and lock usage of some methods on voe::Channel to
   // specific threads we know about. The goal is to eventually split up
   // voe::Channel into parts with single-threaded semantics, and thereby reduce
   // the need for locks.
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker module_process_thread_checker_;
   // Methods accessed from audio and video threads are checked for sequential-
   // only access. We don't necessarily own and control these threads, so thread
   // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
   // audio thread to another, but access is still sequential.
   rtc::RaceChecker audio_thread_race_checker_;
   rtc::RaceChecker video_capture_thread_race_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // VOICE_ENGINE_CHANNEL_PROXY_H_