diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.cc b/talk/app/webrtc/test/fakeaudiocapturemodule.cc
index 934af441dd..c6339d3c3f 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.cc
@@ -704,20 +704,12 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
     }
     ResetRecBuffer();
     uint32_t nSamplesOut = 0;
-#ifdef USE_WEBRTC_DEV_BRANCH
     int64_t elapsed_time_ms = 0;
-#else
-    uint32_t rtp_timestamp = 0;
-#endif
     int64_t ntp_time_ms = 0;
     if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
                                          kNumberOfChannels, kSamplesPerSecond,
                                          rec_buffer_, nSamplesOut,
-#ifdef USE_WEBRTC_DEV_BRANCH
                                          &elapsed_time_ms, &ntp_time_ms) != 0) {
-#else
-                                         &rtp_timestamp, &ntp_time_ms) != 0) {
-#endif
       ASSERT(false);
     }
     ASSERT(nSamplesOut == kNumberSamples);
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index f9c9ea7298..8fcbfd7012 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -85,11 +85,7 @@ class FakeAdmTest : public testing::Test,
                                    const uint32_t samplesPerSec,
                                    void* audioSamples,
                                    uint32_t& nSamplesOut,
-#ifdef USE_WEBRTC_DEV_BRANCH
                                    int64_t* elapsed_time_ms,
-#else
-                                   uint32_t* rtp_timestamp,
-#endif
                                    int64_t* ntp_time_ms) {
     ++pull_iterations_;
     const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
@@ -97,11 +93,7 @@ class FakeAdmTest : public testing::Test,
         CopyFromRecBuffer(audioSamples, audio_buffer_size):
         GenerateZeroBuffer(audioSamples, audio_buffer_size);
     nSamplesOut = bytes_out / nBytesPerSample;
-#ifdef USE_WEBRTC_DEV_BRANCH
     *elapsed_time_ms = 0;
-#else
-    *rtp_timestamp = 0;
-#endif
     *ntp_time_ms = 0;
     return 0;
   }
diff --git a/talk/build/common.gypi b/talk/build/common.gypi
index 296165a2fb..6cde99fa29 100644
--- a/talk/build/common.gypi
+++ b/talk/build/common.gypi
@@ -67,7 +67,6 @@
       'HAVE_SRTP',
       'HAVE_WEBRTC_VIDEO',
       'HAVE_WEBRTC_VOICE',
-      'USE_WEBRTC_DEV_BRANCH',
     ],
     'conditions': [
       # TODO(ronghuawu): Support dynamic library build.
diff --git a/talk/media/webrtc/fakewebrtccommon.h b/talk/media/webrtc/fakewebrtccommon.h
index d1c2320035..4e867074e6 100644
--- a/talk/media/webrtc/fakewebrtccommon.h
+++ b/talk/media/webrtc/fakewebrtccommon.h
@@ -41,10 +41,8 @@ namespace cricket {
 #define WEBRTC_BOOL_STUB(method, args) \
   virtual bool method args OVERRIDE { return true; }
 
-#ifdef USE_WEBRTC_DEV_BRANCH
 #define WEBRTC_BOOL_STUB_CONST(method, args) \
   virtual bool method args const OVERRIDE { return true; }
-#endif
 
 #define WEBRTC_VOID_STUB(method, args) \
   virtual void method args OVERRIDE {}
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 56dc19a32f..566da3c5ec 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -40,9 +40,7 @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/stringutils.h"
-#ifdef USE_WEBRTC_DEV_BRANCH
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
-#endif
 #include "webrtc/video_engine/include/vie_network.h"
 
 namespace cricket {
@@ -76,7 +74,6 @@ static const int kOpusBandwidthFb = 20000;
     } \
   } while (0);
 
-#ifdef USE_WEBRTC_DEV_BRANCH
 class FakeAudioProcessing : public webrtc::AudioProcessing {
  public:
   FakeAudioProcessing() : experimental_ns_enabled_(false) {}
@@ -156,7 +153,6 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
  private:
   bool experimental_ns_enabled_;
 };
-#endif
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
@@ -442,11 +438,7 @@ class FakeWebRtcVoiceEngine
     return 0;
   }
   virtual webrtc::AudioProcessing* audio_processing() OVERRIDE {
-#ifdef USE_WEBRTC_DEV_BRANCH
     return &audio_processing_;
-#else
-    return NULL;
-#endif
   }
   WEBRTC_FUNC(CreateChannel, ()) {
     return AddChannel();
@@ -628,7 +620,6 @@ class FakeWebRtcVoiceEngine
   WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
                              webrtc::VadModes& mode, bool& disabledDTX));
 
-#ifdef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
     WEBRTC_CHECK_CHANNEL(channel);
     if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
@@ -663,7 +654,6 @@ class FakeWebRtcVoiceEngine
       channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
     return 0;
   }
-#endif  // USE_WEBRTC_DEV_BRANCH
 
   // webrtc::VoEDtmf
   WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code,
@@ -970,11 +960,9 @@ class FakeWebRtcVoiceEngine
     stats.packetsReceived = kIntStatValue;
     return 0;
   }
-#ifdef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
     return SetFECStatus(channel, enable, redPayloadtype);
   }
-#endif
   // TODO(minyue): remove the below function when transition to SetREDStatus
   //               is finished.
   WEBRTC_FUNC(SetFECStatus, (int channel, bool enable, int redPayloadtype)) {
@@ -983,11 +971,9 @@ class FakeWebRtcVoiceEngine
     channels_[channel]->red_type = redPayloadtype;
     return 0;
   }
-#ifdef USE_WEBRTC_DEV_BRANCH
   WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
     return GetFECStatus(channel, enable, redPayloadtype);
   }
-#endif
   // TODO(minyue): remove the below function when transition to GetREDStatus
   //               is finished.
   WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) {
@@ -1310,9 +1296,7 @@ class FakeWebRtcVoiceEngine
   int playout_sample_rate_;
   DtmfInfo dtmf_info_;
   webrtc::VoEMediaProcess* media_processor_;
-#ifdef USE_WEBRTC_DEV_BRANCH
   FakeAudioProcessing audio_processing_;
-#endif
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
diff --git a/talk/media/webrtc/webrtcvideoengine_unittest.cc b/talk/media/webrtc/webrtcvideoengine_unittest.cc
index a3ade11a89..1e9f2ce8d9 100644
--- a/talk/media/webrtc/webrtcvideoengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvideoengine_unittest.cc
@@ -1213,11 +1213,9 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeUsageMethod) {
   EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
   EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
   EXPECT_TRUE(cpu_option.enable_encode_usage_method);
-#ifdef USE_WEBRTC_DEV_BRANCH
   // Verify that optional encode rsd thresholds are not set.
   EXPECT_EQ(-1, cpu_option.low_encode_time_rsd_threshold);
   EXPECT_EQ(-1, cpu_option.high_encode_time_rsd_threshold);
-#endif
 
   // Add a new send stream and verify that cpu options are set from start.
   EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(3)));
@@ -1228,11 +1226,9 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeUsageMethod) {
   EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
   EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
   EXPECT_TRUE(cpu_option.enable_encode_usage_method);
-#ifdef USE_WEBRTC_DEV_BRANCH
   // Verify that optional encode rsd thresholds are not set.
   EXPECT_EQ(-1, cpu_option.low_encode_time_rsd_threshold);
   EXPECT_EQ(-1, cpu_option.high_encode_time_rsd_threshold);
-#endif
 }
 
 TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeRsdThresholds) {
@@ -1255,10 +1251,8 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeRsdThresholds) {
   EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
   EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
   EXPECT_TRUE(cpu_option.enable_encode_usage_method);
-#ifdef USE_WEBRTC_DEV_BRANCH
   EXPECT_EQ(30, cpu_option.low_encode_time_rsd_threshold);
   EXPECT_EQ(40, cpu_option.high_encode_time_rsd_threshold);
-#endif
 
   // Add a new send stream and verify that cpu options are set from start.
   EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(3)));
@@ -1269,10 +1263,8 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeRsdThresholds) {
   EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
   EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
   EXPECT_TRUE(cpu_option.enable_encode_usage_method);
-#ifdef USE_WEBRTC_DEV_BRANCH
   EXPECT_EQ(30, cpu_option.low_encode_time_rsd_threshold);
   EXPECT_EQ(40, cpu_option.high_encode_time_rsd_threshold);
-#endif
 }
 
 // Test that AddRecvStream doesn't create new channel for 1:1 call.
@@ -2105,7 +2097,6 @@ TEST_F(WebRtcVideoEngineTestFake, ExternalCodecIgnored) {
   EXPECT_NE("VP8", codecs[codecs.size() - 1].name);
 }
 
-#ifdef USE_WEBRTC_DEV_BRANCH
 TEST_F(WebRtcVideoEngineTestFake, SetSendCodecsWithExternalH264) {
   encoder_factory_.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264");
   engine_.SetExternalEncoderFactory(&encoder_factory_);
@@ -2241,7 +2232,6 @@ TEST_F(WebRtcVideoEngineTestFake, SetRecvCodecsWithVP8AndExternalH264) {
   // The RTX payload type should have been set.
   EXPECT_EQ(rtx_codec.id, vie_.GetRtxRecvPayloadType(channel_num));
 }
-#endif
 
 // Tests that OnReadyToSend will be propagated into ViE.
 TEST_F(WebRtcVideoEngineTestFake, OnReadyToSend) {
@@ -2445,19 +2435,11 @@ TEST_F(WebRtcVideoMediaChannelTest, DISABLED_SendVp8HdAndReceiveAdaptedVp8Vga) {
   EXPECT_FRAME_WAIT(1, codec.width, codec.height, kTimeout);
 }
 
-#ifdef USE_WEBRTC_DEV_BRANCH
 TEST_F(WebRtcVideoMediaChannelTest, GetStats) {
-#else
-TEST_F(WebRtcVideoMediaChannelTest, DISABLED_GetStats) {
-#endif
   Base::GetStats();
 }
 
-#ifdef USE_WEBRTC_DEV_BRANCH
 TEST_F(WebRtcVideoMediaChannelTest, GetStatsMultipleRecvStreams) {
-#else
-TEST_F(WebRtcVideoMediaChannelTest, DISABLED_GetStatsMultipleRecvStreams) {
-#endif
   Base::GetStatsMultipleRecvStreams();
 }
 
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 533e5b81e1..1de22bc20a 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -956,7 +956,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
         new webrtc::DelayCorrection(experimental_aec));
   }
 
-#ifdef USE_WEBRTC_DEV_BRANCH
   experimental_ns_.SetFrom(options.experimental_ns);
   bool experimental_ns;
   if (experimental_ns_.Get(&experimental_ns)) {
@@ -964,7 +963,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     config.Set<webrtc::ExperimentalNs>(
         new webrtc::ExperimentalNs(experimental_ns));
   }
-#endif
 
   // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
   // returns NULL on audio_processing().
@@ -973,24 +971,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
     audioproc->SetExtraOptions(config);
   }
 
-#ifndef USE_WEBRTC_DEV_BRANCH
-  bool experimental_ns;
-  if (options.experimental_ns.Get(&experimental_ns)) {
-    LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
-    // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
-    // returns NULL on audio_processing().
-    if (audioproc) {
-      if (audioproc->EnableExperimentalNs(experimental_ns) == -1) {
-        LOG_RTCERR1(EnableExperimentalNs, experimental_ns);
-        return false;
-      }
-    } else {
-      LOG(LS_VERBOSE) << "Experimental noise suppression set to "
-                      << experimental_ns;
-    }
-  }
-#endif
-
   uint32 recording_sample_rate;
   if (options.recording_sample_rate.Get(&recording_sample_rate)) {
     LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
@@ -2073,13 +2053,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
   // Disable VAD, FEC, and RED unless we know the other side wants them.
   engine()->voe()->codec()->SetVADStatus(channel, false);
   engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
-#ifdef USE_WEBRTC_DEV_BRANCH
   engine()->voe()->rtp()->SetREDStatus(channel, false);
   engine()->voe()->codec()->SetFECStatus(channel, false);
-#else
-  // TODO(minyue): Remove code under #else case after new WebRTC roll.
-  engine()->voe()->rtp()->SetFECStatus(channel, false);
-#endif  // USE_WEBRTC_DEV_BRANCH
 
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD and DTMF.
@@ -2121,16 +2096,9 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
 
       // Enable redundant encoding of the specified codec. Treat any
       // failure as a fatal internal error.
-#ifdef USE_WEBRTC_DEV_BRANCH
       LOG(LS_INFO) << "Enabling RED on channel " << channel;
       if (engine()->voe()->rtp()->SetREDStatus(channel, true, it->id) == -1) {
         LOG_RTCERR3(SetREDStatus, channel, true, it->id);
-#else
-      // TODO(minyue): Remove code under #else case after new WebRTC roll.
-      LOG(LS_INFO) << "Enabling FEC";
-      if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) {
-        LOG_RTCERR3(SetFECStatus, channel, true, it->id);
-#endif  // USE_WEBRTC_DEV_BRANCH
         return false;
       }
     } else {
@@ -2166,13 +2134,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
   if (enable_codec_fec) {
     LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
                  << channel;
-#ifdef USE_WEBRTC_DEV_BRANCH
     if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
       // Enable codec internal FEC. Treat any failure as fatal internal error.
       LOG_RTCERR2(SetFECStatus, channel, true);
       return false;
     }
-#endif  // USE_WEBRTC_DEV_BRANCH
   }
 
   // maxplaybackrate should be set after SetSendCodec.
@@ -2183,12 +2149,10 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
                  << opus_max_playback_rate
                  << " Hz on channel "
                  << channel;
-#ifdef USE_WEBRTC_DEV_BRANCH
     if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
         channel, opus_max_playback_rate) == -1) {
       LOG(LS_WARNING) << "Could not set maximum playback rate.";
     }
-#endif
   }
 
   // Always update the |send_codec_| to the currently set send codec.
@@ -3432,9 +3396,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
       rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
       rinfo.packets_lost = cs.cumulativeLost;
       rinfo.ext_seqnum = cs.extendedMax;
-#ifdef USE_WEBRTC_DEV_BRANCH
       rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
-#endif
       if (codec.pltype != -1) {
         rinfo.codec_name = codec.plname;
       }
diff --git a/talk/media/webrtc/webrtcvoiceengine_unittest.cc b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
index 434548a918..b013e023d5 100644
--- a/talk/media/webrtc/webrtcvoiceengine_unittest.cc
+++ b/talk/media/webrtc/webrtcvoiceengine_unittest.cc
@@ -1159,7 +1159,6 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
   EXPECT_TRUE(voe_.GetNACK(channel_num));
 }
 
-#ifdef USE_WEBRTC_DEV_BRANCH
 // Test that without useinbandfec, Opus FEC is off.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFec) {
   EXPECT_TRUE(SetupEngine());
@@ -1410,7 +1409,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
   EXPECT_EQ(cricket::kOpusBandwidthNb,
             voe_.GetMaxEncodingBandwidth(channel_num));
 }
-#endif  // USE_WEBRTC_DEV_BRANCH
 
 // Test that we can apply CELT with stereo mode but fail with mono mode.
 TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCelt) {
diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn
index e758d868dc..ee9f42c149 100644
--- a/webrtc/base/BUILD.gn
+++ b/webrtc/base/BUILD.gn
@@ -19,7 +19,6 @@ config("rtc_base_config") {
   defines = [
     "FEATURE_ENABLE_SSL",
     "LOGGING=1",
-    "USE_WEBRTC_DEV_BRANCH",
   ]
 
   # TODO(henrike): issue 3307, make rtc_base build without disabling
@@ -149,7 +148,6 @@ static_library("rtc_base") {
 
   defines = [
     "LOGGING=1",
-    "USE_WEBRTC_DEV_BRANCH",
   ]
 
   sources = [
diff --git a/webrtc/base/base.gyp b/webrtc/base/base.gyp
index 74ba76ae32..5187496fd9 100644
--- a/webrtc/base/base.gyp
+++ b/webrtc/base/base.gyp
@@ -62,7 +62,6 @@
       'defines': [
         'FEATURE_ENABLE_SSL',
         'LOGGING=1',
-        'USE_WEBRTC_DEV_BRANCH',
       ],
       'sources': [
         'arraysize.h',
diff --git a/webrtc/p2p/client/basicportallocator.cc b/webrtc/p2p/client/basicportallocator.cc
index 14cd4a21fd..64fce889ad 100644
--- a/webrtc/p2p/client/basicportallocator.cc
+++ b/webrtc/p2p/client/basicportallocator.cc
@@ -414,11 +414,7 @@ void BasicPortAllocatorSession::DoAllocate() {
       }
 
       if (!(sequence_flags & PORTALLOCATOR_ENABLE_IPV6) &&
-#ifdef USE_WEBRTC_DEV_BRANCH
           networks[i]->GetBestIP().family() == AF_INET6) {
-#else  // USE_WEBRTC_DEV_BRANCH
-          networks[i]->ip().family() == AF_INET6) {
-#endif  // USE_WEBRTC_DEV_BRANCH
         // Skip IPv6 networks unless the flag's been set.
         continue;
       }
@@ -718,12 +714,7 @@ AllocationSequence::AllocationSequence(BasicPortAllocatorSession* session,
                                        uint32 flags)
     : session_(session),
       network_(network),
-
-#ifdef USE_WEBRTC_DEV_BRANCH
       ip_(network->GetBestIP()),
-#else  // USE_WEBRTC_DEV_BRANCH
-      ip_(network->ip()),
-#endif  // USE_WEBRTC_DEV_BRANCH
       config_(config),
       state_(kInit),
       flags_(flags),
@@ -766,11 +757,7 @@ AllocationSequence::~AllocationSequence() {
 
 void AllocationSequence::DisableEquivalentPhases(rtc::Network* network,
     PortConfiguration* config, uint32* flags) {
-#ifdef USE_WEBRTC_DEV_BRANCH
   if (!((network == network_) && (ip_ == network->GetBestIP()))) {
-#else  // USE_WEBRTC_DEV_BRANCH
-  if (!((network == network_) && (ip_ == network->ip()))) {
-#endif  // USE_WEBRTC_DEV_BRANCH
     // Different network setup; nothing is equivalent.
     return;
   }
diff --git a/webrtc/p2p/client/connectivitychecker.cc b/webrtc/p2p/client/connectivitychecker.cc
index 1c3fc0edaf..1fb9165670 100644
--- a/webrtc/p2p/client/connectivitychecker.cc
+++ b/webrtc/p2p/client/connectivitychecker.cc
@@ -222,11 +222,7 @@ void ConnectivityChecker::OnRequestDone(rtc::AsyncHttpRequest* request) {
   }
   rtc::ProxyInfo proxy_info = request->proxy();
   NicMap::iterator i =
-#ifdef USE_WEBRTC_DEV_BRANCH
       nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
-#else  // USE_WEBRTC_DEV_BRANCH
-      nics_.find(NicId(networks[0]->ip(), proxy_info.address));
-#endif  // USE_WEBRTC_DEV_BRANCH
   if (i != nics_.end()) {
     int port = request->port();
     uint32 now = rtc::Time();
@@ -257,11 +253,7 @@ void ConnectivityChecker::OnRelayPortComplete(Port* port) {
   ASSERT(worker_ == rtc::Thread::Current());
   RelayPort* relay_port = reinterpret_cast<RelayPort*>(port);
   const ProtocolAddress* address = relay_port->ServerAddress(0);
-#ifdef USE_WEBRTC_DEV_BRANCH
   rtc::IPAddress ip = port->Network()->GetBestIP();
-#else  // USE_WEBRTC_DEV_BRANCH
-  rtc::IPAddress ip = port->Network()->ip();
-#endif  // USE_WEBRTC_DEV_BRANCH
   NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
   if (i != nics_.end()) {
     // We have it already, add the new information.
@@ -295,11 +287,7 @@ void ConnectivityChecker::OnStunPortComplete(Port* port) {
   ASSERT(worker_ == rtc::Thread::Current());
   const std::vector<Candidate> candidates = port->Candidates();
   Candidate c = candidates[0];
-#ifdef USE_WEBRTC_DEV_BRANCH
   rtc::IPAddress ip = port->Network()->GetBestIP();
-#else  // USE_WEBRTC_DEV_BRANCH
-  rtc::IPAddress ip = port->Network()->ip();
-#endif  // USE_WEBRTC_DEV_BRANCH
   NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
   if (i != nics_.end()) {
     // We have it already, add the new information.
@@ -318,11 +306,7 @@ void ConnectivityChecker::OnStunPortComplete(Port* port) {
 void ConnectivityChecker::OnStunPortError(Port* port) {
   ASSERT(worker_ == rtc::Thread::Current());
   LOG(LS_ERROR) << "Stun address error.";
-#ifdef USE_WEBRTC_DEV_BRANCH
   rtc::IPAddress ip = port->Network()->GetBestIP();
-#else  // USE_WEBRTC_DEV_BRANCH
-  rtc::IPAddress ip = port->Network()->ip();
-#endif  // USE_WEBRTC_DEV_BRANCH
   NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
   if (i != nics_.end()) {
     // We have it already, add the new information.
@@ -362,11 +346,7 @@ StunPort* ConnectivityChecker::CreateStunPort(
   return StunPort::Create(worker_,
                           socket_factory_.get(),
                           network,
-#ifdef USE_WEBRTC_DEV_BRANCH
                           network->GetBestIP(),
-#else  // USE_WEBRTC_DEV_BRANCH
-                          network->ip(),
-#endif  // USE_WEBRTC_DEV_BRANCH
                           0,
                           0,
                           username,
@@ -381,11 +361,7 @@ RelayPort* ConnectivityChecker::CreateRelayPort(
   return RelayPort::Create(worker_,
                            socket_factory_.get(),
                            network,
-#ifdef USE_WEBRTC_DEV_BRANCH
                            network->GetBestIP(),
-#else  // USE_WEBRTC_DEV_BRANCH
-                           network->ip(),
-#endif  // USE_WEBRTC_DEV_BRANCH
                            port_allocator_->min_port(),
                            port_allocator_->max_port(),
                            username,
@@ -406,11 +382,7 @@ void ConnectivityChecker::CreateRelayPorts(
        relay != config->relays.end(); ++relay) {
     for (uint32 i = 0; i < networks.size(); ++i) {
       NicMap::iterator iter =
-#ifdef USE_WEBRTC_DEV_BRANCH
           nics_.find(NicId(networks[i]->GetBestIP(), proxy_info.address));
-#else  // USE_WEBRTC_DEV_BRANCH
-          nics_.find(NicId(networks[i]->ip(), proxy_info.address));
-#endif  // USE_WEBRTC_DEV_BRANCH
       if (iter != nics_.end()) {
         // TODO: Now setting the same start time for all protocols.
         // This might affect accuracy, but since we are mainly looking for
@@ -467,11 +439,7 @@ void ConnectivityChecker::AllocatePorts() {
   rtc::ProxyInfo proxy_info = GetProxyInfo();
   bool allocate_relay_ports = false;
   for (uint32 i = 0; i < networks.size(); ++i) {
-#ifdef USE_WEBRTC_DEV_BRANCH
     if (AddNic(networks[i]->GetBestIP(), proxy_info.address)) {
-#else  // USE_WEBRTC_DEV_BRANCH
-    if (AddNic(networks[i]->ip(), proxy_info.address)) {
-#endif  // USE_WEBRTC_DEV_BRANCH
       Port* port = CreateStunPort(username, password, &config, networks[i]);
       if (port) {
 
@@ -547,11 +515,7 @@ void ConnectivityChecker::RegisterHttpStart(int port) {
   }
   rtc::ProxyInfo proxy_info = GetProxyInfo();
   NicMap::iterator i =
-#ifdef USE_WEBRTC_DEV_BRANCH
       nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
-#else  // USE_WEBRTC_DEV_BRANCH
-      nics_.find(NicId(networks[0]->ip(), proxy_info.address));
-#endif  // USE_WEBRTC_DEV_BRANCH
   if (i != nics_.end()) {
     uint32 now = rtc::Time();
     NicInfo* nic_info = &i->second;
diff --git a/webrtc/p2p/client/connectivitychecker_unittest.cc b/webrtc/p2p/client/connectivitychecker_unittest.cc
index 4b0566f584..71c848397d 100644
--- a/webrtc/p2p/client/connectivitychecker_unittest.cc
+++ b/webrtc/p2p/client/connectivitychecker_unittest.cc
@@ -221,11 +221,7 @@ class ConnectivityCheckerForTest : public ConnectivityChecker {
     return new FakeStunPort(worker(),
                             socket_factory_,
                             network,
-#ifdef USE_WEBRTC_DEV_BRANCH
                             network->GetBestIP(),
-#else  // USE_WEBRTC_DEV_BRANCH
-                            network->ip(),
-#endif  // USE_WEBRTC_DEV_BRANCH
                             kMinPort,
                             kMaxPort,
                             username,
@@ -238,11 +234,7 @@ class ConnectivityCheckerForTest : public ConnectivityChecker {
     return new FakeRelayPort(worker(),
                              socket_factory_,
                              network,
-#ifdef USE_WEBRTC_DEV_BRANCH
                              network->GetBestIP(),
-#else  // USE_WEBRTC_DEV_BRANCH
-                             network->ip(),
-#endif  // USE_WEBRTC_DEV_BRANCH
                              kMinPort,
                              kMaxPort,
                              username,
diff --git a/webrtc/p2p/client/fakeportallocator.h b/webrtc/p2p/client/fakeportallocator.h
index e265834542..89093cf4d5 100644
--- a/webrtc/p2p/client/fakeportallocator.h
+++ b/webrtc/p2p/client/fakeportallocator.h
@@ -48,11 +48,7 @@ class FakePortAllocatorSession : public PortAllocatorSession {
       port_.reset(cricket::UDPPort::Create(worker_thread_,
                                            factory_,
                                            &network_,
-#ifdef USE_WEBRTC_DEV_BRANCH
                                            network_.GetBestIP(),
-#else  // USE_WEBRTC_DEV_BRANCH
-                                           network_.ip(),
-#endif  // USE_WEBRTC_DEV_BRANCH
                                            0,
                                            0,
                                            username(),