Minor lint-fixes in MediaChannel and VideoEngine2.

webrtc/media/base/mediachannel.h

 /*
  *  Copyright (c) 2004 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.
  */
 
 #ifndef WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 #define WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_
 
+#include <algorithm>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/copyonwritebuffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
@@ skipping 321 matching lines @@
     enum SocketType { ST_RTP, ST_RTCP };
     virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
                             const rtc::PacketOptions& options) = 0;
     virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
                           const rtc::PacketOptions& options) = 0;
     virtual int SetOption(SocketType type, rtc::Socket::Option opt,
                           int option) = 0;
     virtual ~NetworkInterface() {}
   };
 
-  MediaChannel(const MediaConfig& config)
+  explicit MediaChannel(const MediaConfig& config)
       : enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
   MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
   virtual ~MediaChannel() {}
 
   // Sets the abstract interface class for sending RTP/RTCP data.
   virtual void SetInterface(NetworkInterface *iface) {
     rtc::CritScope cs(&network_interface_crit_);
     network_interface_ = iface;
     SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
   }
@@ skipping 513 matching lines @@
     ERROR_PLAY_DEVICE_OPEN_FAILED = 200,  // Could not open playout.
     ERROR_PLAY_DEVICE_MUTED,              // Playout muted by OS.
     ERROR_PLAY_DEVICE_REMOVED,            // Playout removed while active.
     ERROR_PLAY_RUNTIME_ERROR,             // Errors in voice processing.
     ERROR_PLAY_SRTP_ERROR,                // Generic SRTP failure.
     ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
     ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
   };
 
   VoiceMediaChannel() {}
-  VoiceMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
+  explicit VoiceMediaChannel(const MediaConfig& config)
+      : MediaChannel(config) {}
   virtual ~VoiceMediaChannel() {}
   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetRtpSendParameters(
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
   virtual webrtc::RtpParameters GetRtpReceiveParameters(
       uint32_t ssrc) const = 0;
   virtual bool SetRtpReceiveParameters(
@@ skipping 63 matching lines @@
     ERROR_REC_DEVICE_REMOVED,             // Device is removed.
     ERROR_REC_SRTP_ERROR,                 // Generic sender SRTP failure.
     ERROR_REC_SRTP_AUTH_FAILED,           // Failed to authenticate packets.
     ERROR_REC_CPU_MAX_CANT_DOWNGRADE,     // Can't downgrade capture anymore.
     ERROR_PLAY_SRTP_ERROR = 200,          // Generic receiver SRTP failure.
     ERROR_PLAY_SRTP_AUTH_FAILED,          // Failed to authenticate packets.
     ERROR_PLAY_SRTP_REPLAY,               // Packet replay detected.
   };
 
   VideoMediaChannel() {}
-  VideoMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
+  explicit VideoMediaChannel(const MediaConfig& config)
+      : MediaChannel(config) {}
   virtual ~VideoMediaChannel() {}
 
   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
   virtual bool SetRtpSendParameters(
       uint32_t ssrc,
       const webrtc::RtpParameters& parameters) = 0;
   virtual webrtc::RtpParameters GetRtpReceiveParameters(
       uint32_t ssrc) const = 0;
@@ skipping 138 matching lines @@
   // Signal when the media channel is ready to send the stream. Arguments are:
   //     writable(bool)
   sigslot::signal1<bool> SignalReadyToSend;
   // Signal for notifying that the remote side has closed the DataChannel.
   sigslot::signal1<uint32_t> SignalStreamClosedRemotely;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_BASE_MEDIACHANNEL_H_