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

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

 /*
  *  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.
  */
 
 #include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
 
 #include <string>
 
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
+namespace webrtc {
+namespace voetest {
+
 namespace {
-  static const unsigned int kReflectorSsrc = 0x0000;
-  static const unsigned int kLocalSsrc = 0x0001;
-  static const unsigned int kFirstRemoteSsrc = 0x0002;
-  static const webrtc::CodecInst kCodecInst =
-      {120, "opus", 48000, 960, 2, 64000};
-  static const int kAudioLevelHeaderId = 1;
 
-  static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
-    const size_t ssrc_pos = 4;
-    unsigned int ssrc = 0;
-    if (len >= (ssrc_pos + sizeof(ssrc))) {
-      ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
-    }
-    return ssrc;
+static const unsigned int kReflectorSsrc = 0x0000;
+static const unsigned int kLocalSsrc = 0x0001;
+static const unsigned int kFirstRemoteSsrc = 0x0002;
+static const webrtc::CodecInst kCodecInst = {120, "opus", 48000, 960, 2, 64000};
+static const int kAudioLevelHeaderId = 1;
+
+static unsigned int ParseRtcpSsrc(const void* data, size_t len) {
+  const size_t ssrc_pos = 4;
+  unsigned int ssrc = 0;
+  if (len >= (ssrc_pos + sizeof(ssrc))) {
+    ssrc = rtc::GetBE32(static_cast<const char*>(data) + ssrc_pos);
   }
+  return ssrc;
+}
+
 }  // namespace
 
-namespace voetest {
-
 ConferenceTransport::ConferenceTransport()
     : packet_event_(webrtc::EventWrapper::Create()),
       thread_(Run, this, "ConferenceTransport"),
       rtt_ms_(0),
       stream_count_(0),
       rtp_header_parser_(webrtc::RtpHeaderParser::Create()) {
   rtp_header_parser_->
       RegisterRtpHeaderExtension(webrtc::kRtpExtensionAudioLevel,
                                  kAudioLevelHeaderId);
 
@@ skipping 239 matching lines @@
 
 bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
                                                 webrtc::CallStatistics* stats) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }
+
 }  // namespace voetest
+}  // namespace webrtc