Don't make a top-level namespace called "voetest"

webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

 /*
  *  Copyright (c) 2015 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 12 matching lines @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_file.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/test/auto_test/fakes/loudest_filter.h"
 
+namespace webrtc {
+namespace voetest {
+
 static const size_t kMaxPacketSizeByte = 1500;
 
-namespace voetest {
-
 // This class is to simulate a conference call. There are two Voice Engines, one
 // for local channels and the other for remote channels. There is a simulated
 // reflector, which exchanges RTCP with local channels. For simplicity, it
 // also uses the Voice Engine for remote channels. One can add streams by
 // calling AddStream(), which creates a remote sender channel and a local
 // receive channel. The remote sender channel plays a file as microphone in a
 // looped fashion. Received streams are mixed and played.
 
 class ConferenceTransport: public webrtc::Transport {
  public:
@@ skipping 104 matching lines @@
   webrtc::VoEBase* remote_base_;
   webrtc::VoECodec* remote_codec_;
   webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
   webrtc::VoENetwork* remote_network_;
   webrtc::VoEFile* remote_file_;
 
   LoudestFilter loudest_filter_;
 
   const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
 };
+
 }  // namespace voetest
+}  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_