Use C++11 loops in WebRtcVoiceMediaEngine/Channel.

talk/media/webrtc/webrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 142 matching lines @@
   // Starts AEC dump using existing file.
   bool StartAecDump(rtc::PlatformFile file);
 
   // Check whether the supplied trace should be ignored.
   bool ShouldIgnoreTrace(const std::string& trace);
 
   // Create a VoiceEngine Channel.
   int CreateMediaVoiceChannel();
 
  private:
-  typedef std::vector<WebRtcVoiceMediaChannel *> ChannelList;
+  typedef std::vector<WebRtcVoiceMediaChannel*> ChannelList;
   typedef sigslot::
       signal3<uint32, MediaProcessorDirection, AudioFrame*> FrameSignal;
 
   void Construct();
   void ConstructCodecs();
   bool GetVoeCodec(int index, webrtc::CodecInst* codec);
   bool InitInternal();
   void SetTraceFilter(int filter);
   void SetTraceOptions(const std::string& options);
   // Applies either options or overrides.  Every option that is "set"
@@ skipping 176 matching lines @@
   int SendRTCPPacket(int channel, const void* data, size_t len) override {
     rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
                        kMaxRtpPacketLen);
     return VoiceMediaChannel::SendRtcp(&packet) ? static_cast<int>(len) : -1;
   }
 
   bool FindSsrc(int channel_num, uint32* ssrc);
   void OnError(uint32 ssrc, int error);
 
   bool sending() const { return send_ != SEND_NOTHING; }
-  int GetReceiveChannelNum(uint32 ssrc);
-  int GetSendChannelNum(uint32 ssrc);
+  int GetReceiveChannelNum(uint32 ssrc) const;
+  int GetSendChannelNum(uint32 ssrc) const;
 
   void SetCall(webrtc::Call* call);
 
  private:
   WebRtcVoiceEngine* engine() { return engine_; }
   int GetLastEngineError() { return engine()->GetLastEngineError(); }
   int GetOutputLevel(int channel);
   bool GetRedSendCodec(const AudioCodec& red_codec,
                        const std::vector<AudioCodec>& all_codecs,
                        webrtc::CodecInst* send_codec);
@@ skipping 27 matching lines @@
   bool IsDefaultChannel(int channel_id) const {
     return channel_id == voe_channel();
   }
   bool SetSendCodecs(int channel, const std::vector<AudioCodec>& codecs);
   bool SetSendBitrateInternal(int bps);
 
   bool SetHeaderExtension(ExtensionSetterFunction setter, int channel_id,
                           const RtpHeaderExtension* extension);
   void TryAddAudioRecvStream(uint32 ssrc);
   void TryRemoveAudioRecvStream(uint32 ssrc);
+  bool SetRecvCodecsInternal(const std::vector<AudioCodec>& new_codecs);
 
   bool SetChannelRecvRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
   bool SetChannelSendRtpHeaderExtensions(
     int channel_id,
     const std::vector<RtpHeaderExtension>& extensions);
 
   rtc::ThreadChecker thread_checker_;
 
@@ skipping 35 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   // Do not lock this on the VoE media processor thread; potential for deadlock
   // exists.
   mutable rtc::CriticalSection receive_channels_cs_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_MEDIA_WEBRTCVOICEENGINE_H_