Remove use of RtpHeaderExtension and clean up

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/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"
 #include "webrtc/base/networkroute.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/sigslot.h"
 #include "webrtc/base/socket.h"
 #include "webrtc/base/window.h"
+#include "webrtc/config.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"
 #include "webrtc/media/base/videosourceinterface.h"
 // TODO(juberti): re-evaluate this include
 #include "webrtc/pc/audiomonitor.h"
 
 namespace rtc {
 class RateLimiter;
 class Timing;
 }
 
 namespace webrtc {
 class AudioSinkInterface;
 }
 
 namespace cricket {
 
 class AudioSource;
 class ScreencastId;
 class VideoCapturer;
 class VideoFrame;
 struct RtpHeader;
 struct VideoFormat;
 
-const int kMinRtpHeaderExtensionId = 1;
-const int kMaxRtpHeaderExtensionId = 255;
 const int kScreencastDefaultFps = 5;
 
 template <class T>
 static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
   std::string str;
   if (val) {
     str = key;
     str += ": ";
     str += val ? rtc::ToString(*val) : "";
     str += ", ";
@@ skipping 243 matching lines @@
 
  private:
   template <typename T>
   static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
     if (o) {
       *s = o;
     }
   }
 };
 
-struct RtpHeaderExtension {
-  RtpHeaderExtension() : id(0) {}
-  RtpHeaderExtension(const std::string& u, int i) : uri(u), id(i) {}
-
-  bool operator==(const RtpHeaderExtension& ext) const {
-    // id is a reserved word in objective-c. Therefore the id attribute has to
-    // be a fully qualified name in order to compile on IOS.
-    return this->id == ext.id &&
-        uri == ext.uri;
-  }
-
-  std::string ToString() const {
-    std::ostringstream ost;
-    ost << "{";
-    ost << "uri: " << uri;
-    ost << ", id: " << id;
-    ost << "}";
-    return ost.str();
-  }
-
-  std::string uri;
-  int id;
-  // TODO(juberti): SendRecv direction;
-};
-
-// Returns the named header extension if found among all extensions, NULL
-// otherwise.
-inline const RtpHeaderExtension* FindHeaderExtension(
-    const std::vector<RtpHeaderExtension>& extensions,
-    const std::string& name) {
-  for (std::vector<RtpHeaderExtension>::const_iterator it = extensions.begin();
-       it != extensions.end(); ++it) {
-    if (it->uri == name)
-      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,
                             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,
@@ skipping 463 matching lines @@
   virtual std::string ToString() const {
     std::ostringstream ost;
     ost << "{";
     ost << "codecs: " << VectorToString(codecs) << ", ";
     ost << "extensions: " << VectorToString(extensions);
     ost << "}";
     return ost.str();
   }
 
   std::vector<Codec> codecs;
-  std::vector<RtpHeaderExtension> extensions;
+  std::vector<webrtc::RtpExtension> extensions;
   // TODO(pthatcher): Add streams.
   RtcpParameters rtcp;
   virtual ~RtpParameters() = default;
 };
 
 // TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
 // encapsulate all the parameters needed for an RtpSender.
 template <class Codec>
 struct RtpSendParameters : RtpParameters<Codec> {
   std::string ToString() const override {
@@ skipping 275 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_