Remove use of RtpHeaderExtension and clean up

webrtc/pc/channel.cc

 /*
  *  Copyright 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.
  */
 
@@ skipping 25 matching lines @@
                        std::unique_ptr<webrtc::AudioSinkInterface>* sink) {
   channel->SetRawAudioSink(ssrc, std::move(*sink));
   return true;
 }
 
 struct SendPacketMessageData : public rtc::MessageData {
   rtc::CopyOnWriteBuffer packet;
   rtc::PacketOptions options;
 };
 
+#if defined(ENABLE_EXTERNAL_AUTH)
+// Returns the named header extension if found among all extensions,
+// empty extension otherwise.
+inline const webrtc::RtpExtension FindHeaderExtension(

[danilchap, 2016/05/18 20:00:09]

why changing function signature and returning copy of the extension instead of
const pointer?

[danilchap, 2016/05/18 20:00:09]

no need for inline inside .cc

[Irfan, 2016/05/19 00:56:22]

Done.

+    const std::vector<webrtc::RtpExtension>& extensions,
+    const std::string& uri) {
+  for (auto extension : extensions) {

[danilchap, 2016/05/18 20:00:09]

const auto& to avoid unnecessary copy (and beeing able to return const pointer)

[Irfan, 2016/05/19 00:56:22]

Done.

+    if (extension.uri == uri)
+      return extension;
+  }
+  return webrtc::RtpExtension();
+}
+#endif
+
 }  // namespace
 
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
   MSG_SEND_RTP_PACKET,
   MSG_SEND_RTCP_PACKET,
   MSG_CHANNEL_ERROR,
   MSG_READYTOSENDDATA,
   MSG_DATARECEIVED,
   MSG_FIRSTPACKETRECEIVED,
@@ skipping 1324 matching lines @@
         SafeSetError(desc.str(), error_desc);
         ret = false;
       }
     }
   }
   remote_streams_ = streams;
   return ret;
 }
 
 void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
-    const std::vector<RtpHeaderExtension>& extensions) {
+    const std::vector<webrtc::RtpExtension>& extensions) {
 // Absolute Send Time extension id is used only with external auth,
 // so do not bother searching for it and making asyncronious call to set
 // something that is not used.
 #if defined(ENABLE_EXTERNAL_AUTH)
-  const RtpHeaderExtension* send_time_extension =
-      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
+  const webrtc::RtpExtension send_time_extension =
+      FindHeaderExtension(extensions, webrtc::RtpExtension::kAbsSendTimeUri);
   int rtp_abs_sendtime_extn_id =
-      send_time_extension ? send_time_extension->id : -1;
+      send_time_extension.id ? send_time_extension.id : -1;
   invoker_.AsyncInvoke<void>(
       network_thread_, Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n,
                             this, rtp_abs_sendtime_extn_id));
 #endif
 }
 
 void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n(
     int rtp_abs_sendtime_extn_id) {
   rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
 }
@@ skipping 936 matching lines @@
   return data_channel_type_ == DCT_RTP && BaseChannel::ShouldSetupDtlsSrtp_n();
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
 }
 
 }  // namespace cricket