Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies.

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 <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/base/basictypes.h"
-#include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/base/buffer.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
 #include "webrtc/media/base/codec.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/base/videosinkinterface.h"
@@ skipping 307 matching lines @@
       return &(*it);
   }
   return NULL;
 }
 
 class MediaChannel : public sigslot::has_slots<> {
  public:
   class NetworkInterface {
    public:
     enum SocketType { ST_RTP, ST_RTCP };
-    virtual bool SendPacket(rtc::CopyOnWriteBuffer* packet,
+    virtual bool SendPacket(rtc::Buffer* packet,
                             const rtc::PacketOptions& options) = 0;
-    virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
+    virtual bool SendRtcp(rtc::Buffer* packet,
                           const rtc::PacketOptions& options) = 0;
     virtual int SetOption(SocketType type, rtc::Socket::Option opt,
                           int option) = 0;
     virtual ~NetworkInterface() {}
   };
 
   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);
   }
   virtual rtc::DiffServCodePoint PreferredDscp() const {
     return rtc::DSCP_DEFAULT;
   }
   // Called when a RTP packet is received.
-  virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
+  virtual void OnPacketReceived(rtc::Buffer* packet,
                                 const rtc::PacketTime& packet_time) = 0;
   // Called when a RTCP packet is received.
-  virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
+  virtual void OnRtcpReceived(rtc::Buffer* packet,
                               const rtc::PacketTime& packet_time) = 0;
   // Called when the socket's ability to send has changed.
   virtual void OnReadyToSend(bool ready) = 0;
   // Creates a new outgoing media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddSendStream(const StreamParams& sp) = 0;
   // Removes an outgoing media stream.
   // ssrc must be the first SSRC of the media stream if the stream uses
   // multiple SSRCs.
   virtual bool RemoveSendStream(uint32_t ssrc) = 0;
   // Creates a new incoming media stream with SSRCs and CNAME as described
   // by sp.
   virtual bool AddRecvStream(const StreamParams& sp) = 0;
   // Removes an incoming media stream.
   // ssrc must be the first SSRC of the media stream if the stream uses
   // multiple SSRCs.
   virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
 
   // Returns the absoulte sendtime extension id value from media channel.
   virtual int GetRtpSendTimeExtnId() const {
     return -1;
   }
 
   // Base method to send packet using NetworkInterface.
-  bool SendPacket(rtc::CopyOnWriteBuffer* packet,
-                  const rtc::PacketOptions& options) {
+  bool SendPacket(rtc::Buffer* packet, const rtc::PacketOptions& options) {
     return DoSendPacket(packet, false, options);
   }
 
-  bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
-                const rtc::PacketOptions& options) {
+  bool SendRtcp(rtc::Buffer* packet, const rtc::PacketOptions& options) {
     return DoSendPacket(packet, true, options);
   }
 
   int SetOption(NetworkInterface::SocketType type,
                 rtc::Socket::Option opt,
                 int option) {
     rtc::CritScope cs(&network_interface_crit_);
     if (!network_interface_)
       return -1;
 
     return network_interface_->SetOption(type, opt, option);
   }
 
  private:
   // This method sets DSCP |value| on both RTP and RTCP channels.
   int SetDscp(rtc::DiffServCodePoint value) {
     int ret;
     ret = SetOption(NetworkInterface::ST_RTP,
                     rtc::Socket::OPT_DSCP,
                     value);
     if (ret == 0) {
       ret = SetOption(NetworkInterface::ST_RTCP,
                       rtc::Socket::OPT_DSCP,
                       value);
     }
     return ret;
   }
 
-  bool DoSendPacket(rtc::CopyOnWriteBuffer* packet,
+  bool DoSendPacket(rtc::Buffer* packet,
                     bool rtcp,
                     const rtc::PacketOptions& options) {
     rtc::CritScope cs(&network_interface_crit_);
     if (!network_interface_)
       return false;
 
     return (!rtcp) ? network_interface_->SendPacket(packet, options)
                    : network_interface_->SendRtcp(packet, options);
   }
 
@@ skipping 646 matching lines @@
   virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
 
   // TODO(pthatcher): Implement this.
   virtual bool GetStats(DataMediaInfo* info) { return true; }
 
   virtual bool SetSend(bool send) = 0;
   virtual bool SetReceive(bool receive) = 0;
 
   virtual bool SendData(
       const SendDataParams& params,
-      const rtc::CopyOnWriteBuffer& payload,
+      const rtc::Buffer& payload,
       SendDataResult* result = NULL) = 0;
   // Signals when data is received (params, data, len)
   sigslot::signal3<const ReceiveDataParams&,
                    const char*,
                    size_t> SignalDataReceived;
   // 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_