diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 42b74ff..890de51 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -58,6 +58,27 @@
   ]
 }
 if (rtc_include_tests) {
+  rtc_source_set("audio_end_to_end_test") {
+    testonly = true
+
+    sources = [
+      "test/audio_end_to_end_test.cc",
+      "test/audio_end_to_end_test.h",
+    ]
+    deps = [
+      ":audio",
+      "../system_wrappers:system_wrappers",
+      "../test:fake_audio_device",
+      "../test:test_common",
+      "../test:test_support",
+    ]
+
+    if (!build_with_chromium && is_clang) {
+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+    }
+  }
+
   rtc_source_set("audio_tests") {
     testonly = true
 
@@ -80,6 +101,7 @@
     ]
     deps = [
       ":audio",
+      ":audio_end_to_end_test",
       "../api:mock_audio_mixer",
       "../call:rtp_receiver",
       "../modules/audio_device:mock_audio_device",
@@ -96,6 +118,11 @@
       "//testing/gtest",
     ]
 
+    if (!rtc_use_memcheck) {
+      # This test is timing dependent, which rules out running on memcheck bots.
+      sources += [ "test/audio_stats_test.cc" ]
+    }
+
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
@@ -108,10 +135,10 @@
 
       sources = [
         "test/low_bandwidth_audio_test.cc",
-        "test/low_bandwidth_audio_test.h",
       ]
 
       deps = [
+        ":audio_end_to_end_test",
         "../common_audio",
         "../rtc_base:rtc_base_approved",
         "../system_wrappers",
diff --git a/audio/test/audio_end_to_end_test.cc b/audio/test/audio_end_to_end_test.cc
new file mode 100644
index 0000000..5d4cbf0
--- /dev/null
+++ b/audio/test/audio_end_to_end_test.cc
@@ -0,0 +1,105 @@
+/*
+ *  Copyright (c) 2017 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 <algorithm>
+
+#include "webrtc/audio/test/audio_end_to_end_test.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/fake_audio_device.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+// Wait half a second between stopping sending and stopping receiving audio.
+constexpr int kExtraRecordTimeMs = 500;
+
+constexpr int kSampleRate = 48000;
+}  // namespace
+
+AudioEndToEndTest::AudioEndToEndTest()
+    : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
+
+FakeNetworkPipe::Config AudioEndToEndTest::GetNetworkPipeConfig() const {
+  return FakeNetworkPipe::Config();
+}
+
+size_t AudioEndToEndTest::GetNumVideoStreams() const {
+  return 0;
+}
+
+size_t AudioEndToEndTest::GetNumAudioStreams() const {
+  return 1;
+}
+
+size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
+  return 0;
+}
+
+std::unique_ptr<test::FakeAudioDevice::Capturer>
+    AudioEndToEndTest::CreateCapturer() {
+  return test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, kSampleRate);
+}
+
+std::unique_ptr<test::FakeAudioDevice::Renderer>
+    AudioEndToEndTest::CreateRenderer() {
+  return test::FakeAudioDevice::CreateDiscardRenderer(kSampleRate);
+}
+
+void AudioEndToEndTest::OnFakeAudioDevicesCreated(
+    test::FakeAudioDevice* send_audio_device,
+    test::FakeAudioDevice* recv_audio_device) {
+  send_audio_device_ = send_audio_device;
+}
+
+test::PacketTransport* AudioEndToEndTest::CreateSendTransport(
+    SingleThreadedTaskQueueForTesting* task_queue,
+    Call* sender_call) {
+  return new test::PacketTransport(
+      task_queue, sender_call, this, test::PacketTransport::kSender,
+      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
+}
+
+test::PacketTransport* AudioEndToEndTest::CreateReceiveTransport(
+      SingleThreadedTaskQueueForTesting* task_queue) {
+  return new test::PacketTransport(
+      task_queue, nullptr, this, test::PacketTransport::kReceiver,
+      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
+}
+
+void AudioEndToEndTest::ModifyAudioConfigs(
+  AudioSendStream::Config* send_config,
+  std::vector<AudioReceiveStream::Config>* receive_configs) {
+  // Large bitrate by default.
+  const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
+                                              {{"stereo", "1"}});
+  send_config->send_codec_spec =
+      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
+          {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
+}
+
+void AudioEndToEndTest::OnAudioStreamsCreated(
+    AudioSendStream* send_stream,
+    const std::vector<AudioReceiveStream*>& receive_streams) {
+  ASSERT_NE(nullptr, send_stream);
+  ASSERT_EQ(1u, receive_streams.size());
+  ASSERT_NE(nullptr, receive_streams[0]);
+  send_stream_ = send_stream;
+  receive_stream_ = receive_streams[0];
+}
+
+void AudioEndToEndTest::PerformTest() {
+  // Wait until the input audio file is done...
+  send_audio_device_->WaitForRecordingEnd();
+  // and some extra time to account for network delay.
+  SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
+}
+}  // namespace test
+}  // namespace webrtc
diff --git a/audio/test/low_bandwidth_audio_test.h b/audio/test/audio_end_to_end_test.h
similarity index 64%
rename from audio/test/low_bandwidth_audio_test.h
rename to audio/test/audio_end_to_end_test.h
index ae75707..d14b7a1 100644
--- a/audio/test/low_bandwidth_audio_test.h
+++ b/audio/test/audio_end_to_end_test.h
@@ -7,28 +7,28 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
-#define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
+#ifndef WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
+#define WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/test/call_test.h"
-#include "webrtc/test/fake_audio_device.h"
 
 namespace webrtc {
 namespace test {
 
-class AudioQualityTest : public test::EndToEndTest {
+class AudioEndToEndTest : public test::EndToEndTest {
  public:
-  AudioQualityTest();
+  AudioEndToEndTest();
 
  protected:
-  virtual std::string AudioInputFile();
-  virtual std::string AudioOutputFile();
+  test::FakeAudioDevice* send_audio_device() { return send_audio_device_; }
+  const AudioSendStream* send_stream() const { return send_stream_; }
+  const AudioReceiveStream* receive_stream() const { return receive_stream_; }
 
-  virtual FakeNetworkPipe::Config GetNetworkPipeConfig();
+  virtual FakeNetworkPipe::Config GetNetworkPipeConfig() const;
 
   size_t GetNumVideoStreams() const override;
   size_t GetNumAudioStreams() const override;
@@ -50,15 +50,19 @@
   void ModifyAudioConfigs(
       AudioSendStream::Config* send_config,
       std::vector<AudioReceiveStream::Config>* receive_configs) override;
+  void OnAudioStreamsCreated(
+      AudioSendStream* send_stream,
+      const std::vector<AudioReceiveStream*>& receive_streams) override;
 
   void PerformTest() override;
-  void OnTestFinished() override;
 
  private:
-  test::FakeAudioDevice* send_audio_device_;
+  test::FakeAudioDevice* send_audio_device_ = nullptr;
+  AudioSendStream* send_stream_ = nullptr;
+  AudioReceiveStream* receive_stream_ = nullptr;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
-#endif  // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_
+#endif  // WEBRTC_AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
diff --git a/audio/test/audio_stats_test.cc b/audio/test/audio_stats_test.cc
new file mode 100644
index 0000000..57dfbed
--- /dev/null
+++ b/audio/test/audio_stats_test.cc
@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2017 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/audio/test/audio_end_to_end_test.h"
+#include "webrtc/rtc_base/safe_compare.h"
+#include "webrtc/system_wrappers/include/sleep.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+
+bool IsNear(int reference, int v) {
+  // Margin is 10%.
+  const int error = reference / 10 + 1;
+  return std::abs(reference - v) <= error;
+}
+
+class NoLossTest : public AudioEndToEndTest {
+ public:
+  const int kTestDurationMs = 8000;
+  const int kBytesSent = 69351;
+  const int32_t kPacketsSent = 400;
+  const int64_t kRttMs = 100;
+
+  NoLossTest() = default;
+
+  FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
+    FakeNetworkPipe::Config pipe_config;
+    pipe_config.queue_delay_ms = kRttMs / 2;
+    return pipe_config;
+  }
+
+  void PerformTest() override {
+    SleepMs(kTestDurationMs);
+    send_audio_device()->StopRecording();
+    AudioEndToEndTest::PerformTest();
+  }
+
+  void OnStreamsStopped() override {
+    AudioSendStream::Stats send_stats = send_stream()->GetStats();
+    EXPECT_PRED2(IsNear, kBytesSent, send_stats.bytes_sent);
+    EXPECT_PRED2(IsNear, kPacketsSent, send_stats.packets_sent);
+    EXPECT_EQ(0, send_stats.packets_lost);
+    EXPECT_EQ(0.0f, send_stats.fraction_lost);
+    EXPECT_EQ("opus", send_stats.codec_name);
+    // send_stats.jitter_ms
+    EXPECT_PRED2(IsNear, kRttMs, send_stats.rtt_ms);
+    // Send level is 0 because it is cleared in TransmitMixer::StopSend().
+    EXPECT_EQ(0, send_stats.audio_level);
+    // send_stats.total_input_energy
+    // send_stats.total_input_duration
+    EXPECT_EQ(-1.0f, send_stats.aec_quality_min);
+    EXPECT_EQ(-1, send_stats.echo_delay_median_ms);
+    EXPECT_EQ(-1, send_stats.echo_delay_std_ms);
+    EXPECT_EQ(-100, send_stats.echo_return_loss);
+    EXPECT_EQ(-100, send_stats.echo_return_loss_enhancement);
+    EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood);
+    EXPECT_EQ(0.0f, send_stats.residual_echo_likelihood_recent_max);
+    EXPECT_EQ(false, send_stats.typing_noise_detected);
+
+    AudioReceiveStream::Stats recv_stats = receive_stream()->GetStats();
+    EXPECT_PRED2(IsNear, kBytesSent, recv_stats.bytes_rcvd);
+    EXPECT_PRED2(IsNear, kPacketsSent, recv_stats.packets_rcvd);
+    EXPECT_EQ(0u, recv_stats.packets_lost);
+    EXPECT_EQ(0.0f, recv_stats.fraction_lost);
+    EXPECT_EQ("opus", send_stats.codec_name);
+    // recv_stats.jitter_ms
+    // recv_stats.jitter_buffer_ms
+    EXPECT_EQ(20u, recv_stats.jitter_buffer_preferred_ms);
+    // recv_stats.delay_estimate_ms
+    // Receive level is 0 because it is cleared in Channel::StopPlayout().
+    EXPECT_EQ(0, recv_stats.audio_level);
+    // recv_stats.total_output_energy
+    // recv_stats.total_samples_received
+    // recv_stats.total_output_duration
+    // recv_stats.concealed_samples
+    // recv_stats.expand_rate
+    // recv_stats.speech_expand_rate
+    EXPECT_EQ(0.0, recv_stats.secondary_decoded_rate);
+    EXPECT_EQ(0.0, recv_stats.secondary_discarded_rate);
+    EXPECT_EQ(0.0, recv_stats.accelerate_rate);
+    EXPECT_EQ(0.0, recv_stats.preemptive_expand_rate);
+    EXPECT_EQ(0, recv_stats.decoding_calls_to_silence_generator);
+    // recv_stats.decoding_calls_to_neteq
+    // recv_stats.decoding_normal
+    // recv_stats.decoding_plc
+    EXPECT_EQ(0, recv_stats.decoding_cng);
+    // recv_stats.decoding_plc_cng
+    // recv_stats.decoding_muted_output
+    // Capture start time is -1 because we do not have an associated send stream
+    // on the receiver side.
+    EXPECT_EQ(-1, recv_stats.capture_start_ntp_time_ms);
+
+    // Match these stats between caller and receiver.
+    EXPECT_EQ(send_stats.local_ssrc, recv_stats.remote_ssrc);
+    EXPECT_EQ(*send_stats.codec_payload_type, *recv_stats.codec_payload_type);
+    EXPECT_TRUE(rtc::SafeEq(send_stats.ext_seqnum, recv_stats.ext_seqnum));
+  }
+};
+}  // namespace
+
+using AudioStatsTest = CallTest;
+
+TEST_F(AudioStatsTest, NoLoss) {
+  NoLossTest test;
+  RunBaseTest(&test);
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/audio/test/low_bandwidth_audio_test.cc b/audio/test/low_bandwidth_audio_test.cc
index ea0cdf0..8bbadfb 100644
--- a/audio/test/low_bandwidth_audio_test.cc
+++ b/audio/test/low_bandwidth_audio_test.cc
@@ -8,16 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <algorithm>
-
-#include "webrtc/audio/test/low_bandwidth_audio_test.h"
-#include "webrtc/common_audio/wav_file.h"
+#include "webrtc/audio/test/audio_end_to_end_test.h"
 #include "webrtc/rtc_base/flags.h"
 #include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
-
 DEFINE_int(sample_rate_hz, 16000,
            "Sample rate (Hz) of the produced audio files.");
 
@@ -25,122 +20,59 @@
             "Don't do the full audio recording. "
             "Used to quickly check that the test runs without crashing.");
 
+namespace webrtc {
+namespace test {
 namespace {
 
-// Wait half a second between stopping sending and stopping receiving audio.
-constexpr int kExtraRecordTimeMs = 500;
-
 std::string FileSampleRateSuffix() {
   return std::to_string(FLAG_sample_rate_hz / 1000);
 }
 
-}  // namespace
+class AudioQualityTest : public AudioEndToEndTest {
+ public:
+  AudioQualityTest() = default;
 
-namespace webrtc {
-namespace test {
-
-AudioQualityTest::AudioQualityTest()
-    : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
-
-size_t AudioQualityTest::GetNumVideoStreams() const {
-  return 0;
-}
-size_t AudioQualityTest::GetNumAudioStreams() const {
-  return 1;
-}
-size_t AudioQualityTest::GetNumFlexfecStreams() const {
-  return 0;
-}
-
-std::string AudioQualityTest::AudioInputFile() {
-  return test::ResourcePath("voice_engine/audio_tiny" + FileSampleRateSuffix(),
-                            "wav");
-}
-
-std::string AudioQualityTest::AudioOutputFile() {
-  const ::testing::TestInfo* const test_info =
-      ::testing::UnitTest::GetInstance()->current_test_info();
-  return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
-      "_" + FileSampleRateSuffix() + ".wav";
-}
-
-std::unique_ptr<test::FakeAudioDevice::Capturer>
-    AudioQualityTest::CreateCapturer() {
-  return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
-}
-
-std::unique_ptr<test::FakeAudioDevice::Renderer>
-    AudioQualityTest::CreateRenderer() {
-  return test::FakeAudioDevice::CreateBoundedWavFileWriter(
-      AudioOutputFile(), FLAG_sample_rate_hz);
-}
-
-void AudioQualityTest::OnFakeAudioDevicesCreated(
-    test::FakeAudioDevice* send_audio_device,
-    test::FakeAudioDevice* recv_audio_device) {
-  send_audio_device_ = send_audio_device;
-}
-
-FakeNetworkPipe::Config AudioQualityTest::GetNetworkPipeConfig() {
-  return FakeNetworkPipe::Config();
-}
-
-test::PacketTransport* AudioQualityTest::CreateSendTransport(
-    SingleThreadedTaskQueueForTesting* task_queue,
-    Call* sender_call) {
-  return new test::PacketTransport(
-      task_queue, sender_call, this, test::PacketTransport::kSender,
-      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
-}
-
-test::PacketTransport* AudioQualityTest::CreateReceiveTransport(
-    SingleThreadedTaskQueueForTesting* task_queue) {
-  return new test::PacketTransport(
-      task_queue, nullptr, this, test::PacketTransport::kReceiver,
-      test::CallTest::payload_type_map_, GetNetworkPipeConfig());
-}
-
-void AudioQualityTest::ModifyAudioConfigs(
-  AudioSendStream::Config* send_config,
-  std::vector<AudioReceiveStream::Config>* receive_configs) {
-  // Large bitrate by default.
-  const webrtc::SdpAudioFormat kDefaultFormat("OPUS", 48000, 2,
-                                              {{"stereo", "1"}});
-  send_config->send_codec_spec =
-      rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
-          {test::CallTest::kAudioSendPayloadType, kDefaultFormat});
-}
-
-void AudioQualityTest::PerformTest() {
-  if (FLAG_quick) {
-    // Let the recording run for a small amount of time to check if it works.
-    SleepMs(1000);
-  } else {
-    // Wait until the input audio file is done...
-    send_audio_device_->WaitForRecordingEnd();
-    // and some extra time to account for network delay.
-    SleepMs(GetNetworkPipeConfig().queue_delay_ms + kExtraRecordTimeMs);
+ private:
+  std::string AudioInputFile() const {
+    return test::ResourcePath(
+        "voice_engine/audio_tiny" + FileSampleRateSuffix(), "wav");
   }
-}
 
-void AudioQualityTest::OnTestFinished() {
-  const ::testing::TestInfo* const test_info =
-      ::testing::UnitTest::GetInstance()->current_test_info();
+  std::string AudioOutputFile() const {
+    const ::testing::TestInfo* const test_info =
+        ::testing::UnitTest::GetInstance()->current_test_info();
+    return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
+        "_" + FileSampleRateSuffix() + ".wav";
+  }
 
-  // Output information about the input and output audio files so that further
-  // processing can be done by an external process.
-  printf("TEST %s %s %s\n", test_info->name(),
-         AudioInputFile().c_str(), AudioOutputFile().c_str());
-}
+  std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override {
+    return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
+  }
 
+  std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override {
+    return test::FakeAudioDevice::CreateBoundedWavFileWriter(
+        AudioOutputFile(), FLAG_sample_rate_hz);
+  }
 
-using LowBandwidthAudioTest = CallTest;
+  void PerformTest() override {
+    if (FLAG_quick) {
+      // Let the recording run for a small amount of time to check if it works.
+      SleepMs(1000);
+    } else {
+      AudioEndToEndTest::PerformTest();
+    }
+  }
 
-TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
-  AudioQualityTest test;
-  RunBaseTest(&test);
-}
+  void OnStreamsStopped() override {
+    const ::testing::TestInfo* const test_info =
+        ::testing::UnitTest::GetInstance()->current_test_info();
 
+    // Output information about the input and output audio files so that further
+    // processing can be done by an external process.
+    printf("TEST %s %s %s\n", test_info->name(),
+           AudioInputFile().c_str(), AudioOutputFile().c_str());
+  }
+};
 
 class Mobile2GNetworkTest : public AudioQualityTest {
   void ModifyAudioConfigs(AudioSendStream::Config* send_config,
@@ -156,7 +88,7 @@
                {"stereo", "1"}}}});
   }
 
-  FakeNetworkPipe::Config GetNetworkPipeConfig() override {
+  FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
     FakeNetworkPipe::Config pipe_config;
     pipe_config.link_capacity_kbps = 12;
     pipe_config.queue_length_packets = 1500;
@@ -164,11 +96,18 @@
     return pipe_config;
   }
 };
+}  // namespace
+
+using LowBandwidthAudioTest = CallTest;
+
+TEST_F(LowBandwidthAudioTest, GoodNetworkHighBitrate) {
+  AudioQualityTest test;
+  RunBaseTest(&test);
+}
 
 TEST_F(LowBandwidthAudioTest, Mobile2GNetwork) {
   Mobile2GNetworkTest test;
   RunBaseTest(&test);
 }
-
 }  // namespace test
 }  // namespace webrtc
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 641be6f..6760b14 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -1914,6 +1914,33 @@
   EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
 }
 
+// Test that we can get capture start ntp time.
+TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioOnlyMediaStream();
+
+  auto audio_track = callee()->CreateLocalAudioTrack();
+  callee()->AddMediaStreamFromTracks(audio_track, nullptr);
+
+  // Do offer/answer, wait for the callee to receive some frames.
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+  // Get the remote audio track created on the receiver, so they can be used as
+  // GetStats filters.
+  StreamCollectionInterface* remote_streams = callee()->remote_streams();
+  ASSERT_EQ(1u, remote_streams->count());
+  ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
+  MediaStreamTrackInterface* remote_audio_track =
+      remote_streams->at(0)->GetAudioTracks()[0];
+
+  // Get the audio output level stats. Note that the level is not available
+  // until an RTCP packet has been received.
+  EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track)->
+                   CaptureStartNtpTime() > 0, 2 * kMaxWaitForFramesMs);
+}
+
 // Test that we can get stats (using the new stats implemnetation) for
 // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
 // SDP.
diff --git a/pc/test/mockpeerconnectionobservers.h b/pc/test/mockpeerconnectionobservers.h
index 5367eeb..84c80f8 100644
--- a/pc/test/mockpeerconnectionobservers.h
+++ b/pc/test/mockpeerconnectionobservers.h
@@ -125,6 +125,8 @@
             &stats_.bytes_received);
         GetIntValue(r, StatsReport::kStatsValueNameBytesSent,
             &stats_.bytes_sent);
+        GetInt64Value(r, StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
+            &stats_.capture_start_ntp_time);
       } else if (r->type() == StatsReport::kStatsReportTypeBwe) {
         stats_.timestamp = r->timestamp();
         GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
@@ -163,6 +165,11 @@
     return stats_.bytes_sent;
   }
 
+  int64_t CaptureStartNtpTime() const {
+    RTC_CHECK(called_);
+    return stats_.capture_start_ntp_time;
+  }
+
   int AvailableReceiveBandwidth() const {
     RTC_CHECK(called_);
     return stats_.available_receive_bandwidth;
@@ -190,6 +197,17 @@
     return v != nullptr;
   }
 
+  bool GetInt64Value(const StatsReport* report,
+                   StatsReport::StatsValueName name,
+                   int64_t* value) {
+    const StatsReport::Value* v = report->FindValue(name);
+    if (v) {
+      // TODO(tommi): We should really just be using an int here :-/
+      *value = rtc::FromString<int64_t>(v->ToString());
+    }
+    return v != nullptr;
+  }
+
   bool GetStringValue(const StatsReport* report,
                       StatsReport::StatsValueName name,
                       std::string* value) {
@@ -208,6 +226,7 @@
       audio_input_level = 0;
       bytes_received = 0;
       bytes_sent = 0;
+      capture_start_ntp_time = 0;
       available_receive_bandwidth = 0;
       dtls_cipher.clear();
       srtp_cipher.clear();
@@ -219,6 +238,7 @@
     int audio_input_level;
     int bytes_received;
     int bytes_sent;
+    int64_t capture_start_ntp_time;
     int available_receive_bandwidth;
     std::string dtls_cipher;
     std::string srtp_cipher;
diff --git a/test/call_test.cc b/test/call_test.cc
index b5d7236..d4084d5 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -153,8 +153,9 @@
 
   test->PerformTest();
 
-  task_queue_.SendTask([this]() {
+  task_queue_.SendTask([this, test]() {
     Stop();
+    test->OnStreamsStopped();
     DestroyStreams();
     send_transport_.reset();
     receive_transport_.reset();
@@ -162,8 +163,6 @@
     if (num_audio_streams_ > 0)
       DestroyVoiceEngines();
   });
-
-  test->OnTestFinished();
 }
 
 void CallTest::CreateCalls(const Call::Config& sender_config,
@@ -223,7 +222,7 @@
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
     audio_send_config_.send_codec_spec =
         rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
-            {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}});
+            {kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}}});
     audio_send_config_.encoder_factory = encoder_factory_;
   }
 
@@ -590,7 +589,7 @@
     FrameGeneratorCapturer* frame_generator_capturer) {
 }
 
-void BaseTest::OnTestFinished() {
+void BaseTest::OnStreamsStopped() {
 }
 
 SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
diff --git a/test/call_test.h b/test/call_test.h
index b0ae3f6..3372015 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -223,7 +223,7 @@
   virtual void OnFrameGeneratorCapturerCreated(
       FrameGeneratorCapturer* frame_generator_capturer);
 
-  virtual void OnTestFinished();
+  virtual void OnStreamsStopped();
 
   std::unique_ptr<webrtc::RtcEventLog> event_log_;
 };
diff --git a/voice_engine/BUILD.gn b/voice_engine/BUILD.gn
index 809333b..627f412 100644
--- a/voice_engine/BUILD.gn
+++ b/voice_engine/BUILD.gn
@@ -266,10 +266,6 @@
 
       sources = [
         "test/auto_test/automated_mode.cc",
-        "test/auto_test/fakes/conference_transport.cc",
-        "test/auto_test/fakes/conference_transport.h",
-        "test/auto_test/fakes/loudest_filter.cc",
-        "test/auto_test/fakes/loudest_filter.h",
         "test/auto_test/fixtures/after_initialization_fixture.cc",
         "test/auto_test/fixtures/after_initialization_fixture.h",
         "test/auto_test/fixtures/after_streaming_fixture.cc",
@@ -284,7 +280,6 @@
         "test/auto_test/standard/rtp_rtcp_before_streaming_test.cc",
         "test/auto_test/standard/rtp_rtcp_extensions.cc",
         "test/auto_test/standard/rtp_rtcp_test.cc",
-        "test/auto_test/voe_conference_test.cc",
         "test/auto_test/voe_standard_test.cc",
         "test/auto_test/voe_standard_test.h",
         "test/auto_test/voe_test_defines.h",
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.cc b/voice_engine/test/auto_test/fakes/conference_transport.cc
deleted file mode 100644
index 305b236..0000000
--- a/voice_engine/test/auto_test/fakes/conference_transport.cc
+++ /dev/null
@@ -1,307 +0,0 @@
-/*
- *  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/rtc_base/byteorder.h"
-#include "webrtc/rtc_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;
-}
-
-}  // namespace
-
-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);
-
-  local_voe_ = webrtc::VoiceEngine::Create();
-  local_base_ = webrtc::VoEBase::GetInterface(local_voe_);
-  local_network_ = webrtc::VoENetwork::GetInterface(local_voe_);
-  local_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(local_voe_);
-
-  local_apm_ = webrtc::AudioProcessing::Create();
-  local_base_->Init(nullptr, local_apm_.get(), nullptr);
-
-  // In principle, we can use one VoiceEngine to achieve the same goal. Well, in
-  // here, we use two engines to make it more like reality.
-  remote_voe_ = webrtc::VoiceEngine::Create();
-  remote_base_ = webrtc::VoEBase::GetInterface(remote_voe_);
-  remote_codec_ = webrtc::VoECodec::GetInterface(remote_voe_);
-  remote_network_ = webrtc::VoENetwork::GetInterface(remote_voe_);
-  remote_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(remote_voe_);
-  remote_file_ = webrtc::VoEFile::GetInterface(remote_voe_);
-
-  remote_apm_ = webrtc::AudioProcessing::Create();
-  remote_base_->Init(nullptr, remote_apm_.get(), nullptr);
-
-  local_sender_ = local_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
-      ->GetChannelProxy(local_sender_)
-      ->RegisterLegacyReceiveCodecs();
-  EXPECT_EQ(0, local_network_->RegisterExternalTransport(local_sender_, *this));
-  EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(local_sender_, kLocalSsrc));
-  EXPECT_EQ(0, local_rtp_rtcp_->
-      SetSendAudioLevelIndicationStatus(local_sender_, true,
-                                        kAudioLevelHeaderId));
-  EXPECT_EQ(0, local_base_->StartSend(local_sender_));
-
-  reflector_ = remote_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
-      ->GetChannelProxy(reflector_)
-      ->RegisterLegacyReceiveCodecs();
-  EXPECT_EQ(0, remote_network_->RegisterExternalTransport(reflector_, *this));
-  EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(reflector_, kReflectorSsrc));
-
-  thread_.Start();
-  thread_.SetPriority(rtc::kHighPriority);
-}
-
-ConferenceTransport::~ConferenceTransport() {
-  // Must stop sending, otherwise DispatchPackets() cannot quit.
-  EXPECT_EQ(0, remote_network_->DeRegisterExternalTransport(reflector_));
-  EXPECT_EQ(0, local_network_->DeRegisterExternalTransport(local_sender_));
-
-  while (!streams_.empty()) {
-    auto stream = streams_.begin();
-    RemoveStream(stream->first);
-  }
-
-  thread_.Stop();
-
-  remote_file_->Release();
-  remote_rtp_rtcp_->Release();
-  remote_network_->Release();
-  remote_base_->Release();
-
-  local_rtp_rtcp_->Release();
-  local_network_->Release();
-  local_base_->Release();
-
-  EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
-  EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
-}
-
-bool ConferenceTransport::SendRtp(const uint8_t* data,
-                                  size_t len,
-                                  const webrtc::PacketOptions& options) {
-  StorePacket(Packet::Rtp, data, len);
-  return true;
-}
-
-bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) {
-  StorePacket(Packet::Rtcp, data, len);
-  return true;
-}
-
-int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
-    const {
-  rtc::CritScope lock(&stream_crit_);
-  auto it = streams_.find(sender_ssrc);
-  if (it != streams_.end()) {
-    return it->second.second;
-  }
-  return -1;
-}
-
-void ConferenceTransport::StorePacket(Packet::Type type,
-                                      const void* data,
-                                      size_t len) {
-  {
-    rtc::CritScope lock(&pq_crit_);
-    packet_queue_.push_back(Packet(type, data, len, rtc::TimeMillis()));
-  }
-  packet_event_->Set();
-}
-
-// This simulates the flow of RTP and RTCP packets. Complications like that
-// a packet is first sent to the reflector, and then forwarded to the receiver
-// are simplified, in this particular case, to a direct link between the sender
-// and the receiver.
-void ConferenceTransport::SendPacket(const Packet& packet) {
-  int destination = -1;
-
-  switch (packet.type_) {
-    case Packet::Rtp: {
-      webrtc::RTPHeader rtp_header;
-      rtp_header_parser_->Parse(packet.data_, packet.len_, &rtp_header);
-      if (rtp_header.ssrc == kLocalSsrc) {
-        remote_network_->ReceivedRTPPacket(reflector_, packet.data_,
-                                           packet.len_, webrtc::PacketTime());
-      } else {
-        if (loudest_filter_.ForwardThisPacket(rtp_header)) {
-          destination = GetReceiverChannelForSsrc(rtp_header.ssrc);
-          if (destination != -1) {
-            local_network_->ReceivedRTPPacket(destination, packet.data_,
-                                              packet.len_,
-                                              webrtc::PacketTime());
-          }
-        }
-      }
-      break;
-    }
-    case Packet::Rtcp: {
-      unsigned int sender_ssrc = ParseRtcpSsrc(packet.data_, packet.len_);
-      if (sender_ssrc == kLocalSsrc) {
-        remote_network_->ReceivedRTCPPacket(reflector_, packet.data_,
-                                            packet.len_);
-      } else if (sender_ssrc == kReflectorSsrc) {
-        local_network_->ReceivedRTCPPacket(local_sender_, packet.data_,
-                                           packet.len_);
-      } else {
-        destination = GetReceiverChannelForSsrc(sender_ssrc);
-        if (destination != -1) {
-          local_network_->ReceivedRTCPPacket(destination, packet.data_,
-                                             packet.len_);
-        }
-      }
-      break;
-    }
-  }
-}
-
-bool ConferenceTransport::DispatchPackets() {
-  switch (packet_event_->Wait(1000)) {
-    case webrtc::kEventSignaled:
-      break;
-    case webrtc::kEventTimeout:
-      return true;
-    case webrtc::kEventError:
-      ADD_FAILURE() << "kEventError encountered.";
-      return true;
-  }
-
-  while (true) {
-    Packet packet;
-    {
-      rtc::CritScope lock(&pq_crit_);
-      if (packet_queue_.empty())
-        break;
-      packet = packet_queue_.front();
-      packet_queue_.pop_front();
-    }
-
-    int32_t elapsed_time_ms = rtc::TimeSince(packet.send_time_ms_);
-    int32_t sleep_ms = rtt_ms_ / 2 - elapsed_time_ms;
-    if (sleep_ms > 0) {
-      // Every packet should be delayed by half of RTT.
-      webrtc::SleepMs(sleep_ms);
-    }
-
-    SendPacket(packet);
-  }
-  return true;
-}
-
-void ConferenceTransport::SetRtt(unsigned int rtt_ms) {
-  rtt_ms_ = rtt_ms;
-}
-
-unsigned int ConferenceTransport::AddStream(std::string file_name,
-                                            webrtc::FileFormats format) {
-  const int new_sender = remote_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(remote_voe_)
-      ->GetChannelProxy(new_sender)
-      ->RegisterLegacyReceiveCodecs();
-  EXPECT_EQ(0, remote_network_->RegisterExternalTransport(new_sender, *this));
-
-  const unsigned int remote_ssrc = kFirstRemoteSsrc + stream_count_++;
-  EXPECT_EQ(0, remote_rtp_rtcp_->SetLocalSSRC(new_sender, remote_ssrc));
-  EXPECT_EQ(0, remote_rtp_rtcp_->
-      SetSendAudioLevelIndicationStatus(new_sender, true, kAudioLevelHeaderId));
-
-  EXPECT_EQ(0, remote_codec_->SetSendCodec(new_sender, kCodecInst));
-  EXPECT_EQ(0, remote_base_->StartSend(new_sender));
-  EXPECT_EQ(0, remote_file_->StartPlayingFileAsMicrophone(
-      new_sender, file_name.c_str(), true, false, format, 1.0));
-
-  const int new_receiver = local_base_->CreateChannel();
-  static_cast<webrtc::VoiceEngineImpl*>(local_voe_)
-      ->GetChannelProxy(new_receiver)
-      ->RegisterLegacyReceiveCodecs();
-  EXPECT_EQ(0, local_base_->AssociateSendChannel(new_receiver, local_sender_));
-
-  EXPECT_EQ(0, local_network_->RegisterExternalTransport(new_receiver, *this));
-  // Receive channels have to have the same SSRC in order to send receiver
-  // reports with this SSRC.
-  EXPECT_EQ(0, local_rtp_rtcp_->SetLocalSSRC(new_receiver, kLocalSsrc));
-
-  {
-    rtc::CritScope lock(&stream_crit_);
-    streams_[remote_ssrc] = std::make_pair(new_sender, new_receiver);
-  }
-  return remote_ssrc;  // remote ssrc used as stream id.
-}
-
-bool ConferenceTransport::RemoveStream(unsigned int id) {
-  rtc::CritScope lock(&stream_crit_);
-  auto it = streams_.find(id);
-  if (it == streams_.end()) {
-    return false;
-  }
-  EXPECT_EQ(0, remote_network_->
-      DeRegisterExternalTransport(it->second.second));
-  EXPECT_EQ(0, local_network_->
-      DeRegisterExternalTransport(it->second.first));
-  EXPECT_EQ(0, remote_base_->DeleteChannel(it->second.second));
-  EXPECT_EQ(0, local_base_->DeleteChannel(it->second.first));
-  streams_.erase(it);
-  return true;
-}
-
-bool ConferenceTransport::StartPlayout(unsigned int id) {
-  int dst = GetReceiverChannelForSsrc(id);
-  if (dst == -1) {
-    return false;
-  }
-  EXPECT_EQ(0, local_base_->StartPlayout(dst));
-  return true;
-}
-
-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
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.h b/voice_engine/test/auto_test/fakes/conference_transport.h
deleted file mode 100644
index a0acd9e..0000000
--- a/voice_engine/test/auto_test/fakes/conference_transport.h
+++ /dev/null
@@ -1,168 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
-#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
-
-#include <deque>
-#include <map>
-#include <memory>
-#include <utility>
-
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/platform_thread.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;
-
-// 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:
-  ConferenceTransport();
-  virtual ~ConferenceTransport();
-
-  /* SetRtt()
-   * Set RTT between local channels and reflector.
-   *
-   * Input:
-   *   rtt_ms : RTT in milliseconds.
-   */
-  void SetRtt(unsigned int rtt_ms);
-
-  /* AddStream()
-   * Adds a stream in the conference.
-   *
-   * Input:
-   *   file_name : name of the file to be added as microphone input.
-   *   format    : format of the input file.
-   *
-   * Returns stream id.
-   */
-  unsigned int AddStream(std::string file_name, webrtc::FileFormats format);
-
-  /* RemoveStream()
-   * Removes a stream with specified ID from the conference.
-   *
-   * Input:
-   *   id : stream id.
-   *
-   * Returns false if the specified stream does not exist, true if succeeds.
-   */
-  bool RemoveStream(unsigned int id);
-
-  /* StartPlayout()
-   * Starts playing out the stream with specified ID, using the default device.
-   *
-   * Input:
-   *   id : stream id.
-   *
-   * Returns false if the specified stream does not exist, true if succeeds.
-   */
-  bool StartPlayout(unsigned int id);
-
-  /* GetReceiverStatistics()
-   * Gets RTCP statistics of the stream with specified ID.
-   *
-   * Input:
-   *   id : stream id;
-   *   stats : pointer to a CallStatistics to store the result.
-   *
-   * Returns false if the specified stream does not exist, true if succeeds.
-   */
-  bool GetReceiverStatistics(unsigned int id, webrtc::CallStatistics* stats);
-
-  // Inherit from class webrtc::Transport.
-  bool SendRtp(const uint8_t* data,
-               size_t len,
-               const webrtc::PacketOptions& options) override;
-  bool SendRtcp(const uint8_t *data, size_t len) override;
-
- private:
-  struct Packet {
-    enum Type { Rtp, Rtcp, } type_;
-
-    Packet() : len_(0) {}
-    Packet(Type type, const void* data, size_t len, int64_t time_ms)
-        : type_(type), len_(len), send_time_ms_(time_ms) {
-      EXPECT_LE(len_, kMaxPacketSizeByte);
-      memcpy(data_, data, len_);
-    }
-
-    uint8_t data_[kMaxPacketSizeByte];
-    size_t len_;
-    int64_t send_time_ms_;
-  };
-
-  static bool Run(void* transport) {
-    return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
-  }
-
-  int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
-  void StorePacket(Packet::Type type, const void* data, size_t len);
-  void SendPacket(const Packet& packet);
-  bool DispatchPackets();
-
-  rtc::CriticalSection pq_crit_;
-  rtc::CriticalSection stream_crit_;
-  const std::unique_ptr<webrtc::EventWrapper> packet_event_;
-  rtc::PlatformThread thread_;
-
-  unsigned int rtt_ms_;
-  unsigned int stream_count_;
-
-  std::map<unsigned int, std::pair<int, int>> streams_
-      RTC_GUARDED_BY(stream_crit_);
-  std::deque<Packet> packet_queue_ RTC_GUARDED_BY(pq_crit_);
-
-  int local_sender_;  // Channel Id of local sender
-  int reflector_;
-
-  webrtc::VoiceEngine* local_voe_;
-  webrtc::VoEBase* local_base_;
-  webrtc::VoERTP_RTCP* local_rtp_rtcp_;
-  webrtc::VoENetwork* local_network_;
-  rtc::scoped_refptr<webrtc::AudioProcessing> local_apm_;
-
-  webrtc::VoiceEngine* remote_voe_;
-  webrtc::VoEBase* remote_base_;
-  webrtc::VoECodec* remote_codec_;
-  webrtc::VoERTP_RTCP* remote_rtp_rtcp_;
-  webrtc::VoENetwork* remote_network_;
-  webrtc::VoEFile* remote_file_;
-  rtc::scoped_refptr<webrtc::AudioProcessing> remote_apm_;
-  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_
diff --git a/voice_engine/test/auto_test/fakes/loudest_filter.cc b/voice_engine/test/auto_test/fakes/loudest_filter.cc
deleted file mode 100644
index ec1d667..0000000
--- a/voice_engine/test/auto_test/fakes/loudest_filter.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- *  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/loudest_filter.h"
-
-#include "webrtc/rtc_base/checks.h"
-
-namespace webrtc {
-namespace voetest {
-
-void LoudestFilter::RemoveTimeoutStreams(int64_t time_ms) {
-  auto it = stream_levels_.begin();
-  while (it != stream_levels_.end()) {
-    if (rtc::TimeDiff(time_ms, it->second.last_time_ms) > kStreamTimeOutMs) {
-      stream_levels_.erase(it++);
-    } else {
-      ++it;
-    }
-  }
-}
-
-unsigned int LoudestFilter::FindQuietestStream() {
-  int quietest_level = kInvalidAudioLevel;
-  unsigned int quietest_ssrc = 0;
-  for (auto stream : stream_levels_) {
-    // A smaller value if audio level corresponds to a louder sound.
-    if (quietest_level == kInvalidAudioLevel ||
-        stream.second.audio_level > quietest_level) {
-      quietest_level = stream.second.audio_level;
-      quietest_ssrc = stream.first;
-    }
-  }
-  return quietest_ssrc;
-}
-
-bool LoudestFilter::ForwardThisPacket(const webrtc::RTPHeader& rtp_header) {
-  int64_t time_now_ms = rtc::TimeMillis();
-  RemoveTimeoutStreams(time_now_ms);
-
-  int source_ssrc = rtp_header.ssrc;
-  int audio_level = rtp_header.extension.hasAudioLevel ?
-      rtp_header.extension.audioLevel : kInvalidAudioLevel;
-
-  if (audio_level == kInvalidAudioLevel) {
-    // Always forward streams with unknown audio level, and don't keep their
-    // states.
-    return true;
-  }
-
-  auto it = stream_levels_.find(source_ssrc);
-  if (it != stream_levels_.end()) {
-    // Stream has been forwarded. Update and continue to forward.
-    it->second.audio_level = audio_level;
-    it->second.last_time_ms = time_now_ms;
-    return true;
-  }
-
-  if (stream_levels_.size() < kMaxMixSize) {
-    stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
-    return true;
-  }
-
-  unsigned int quietest_ssrc = FindQuietestStream();
-  RTC_CHECK_NE(0, quietest_ssrc);
-  // A smaller value if audio level corresponds to a louder sound.
-  if (audio_level < stream_levels_[quietest_ssrc].audio_level) {
-    stream_levels_.erase(quietest_ssrc);
-    stream_levels_[source_ssrc].Set(audio_level, time_now_ms);
-    return true;
-  }
-  return false;
-}
-
-}  // namespace voetest
-}  // namespace webrtc
diff --git a/voice_engine/test/auto_test/fakes/loudest_filter.h b/voice_engine/test/auto_test/fakes/loudest_filter.h
deleted file mode 100644
index dd5a426..0000000
--- a/voice_engine/test/auto_test/fakes/loudest_filter.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
-#define WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
-
-#include <map>
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/timeutils.h"
-
-namespace webrtc {
-namespace voetest {
-
-class LoudestFilter {
- public:
-  /* ForwardThisPacket()
-   * Decide whether to forward a RTP packet, given its header.
-   *
-   * Input:
-   *   rtp_header : Header of the RTP packet of interest.
-   */
-  bool ForwardThisPacket(const webrtc::RTPHeader& rtp_header);
-
- private:
-  struct Status {
-    void Set(int audio_level, int64_t last_time_ms) {
-      this->audio_level = audio_level;
-      this->last_time_ms = last_time_ms;
-    }
-    int audio_level;
-    int64_t last_time_ms;
-  };
-
-  void RemoveTimeoutStreams(int64_t time_ms);
-  unsigned int FindQuietestStream();
-
-  // Keeps the streams being forwarded in pair<SSRC, Status>.
-  std::map<unsigned int, Status> stream_levels_;
-
-  const int32_t kStreamTimeOutMs = 5000;
-  const size_t kMaxMixSize = 3;
-  const int kInvalidAudioLevel = 128;
-};
-
-}  // namespace voetest
-}  // namespace webrtc
-
-#endif  // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_LOUDEST_FILTER_H_
diff --git a/voice_engine/test/auto_test/voe_conference_test.cc b/voice_engine/test/auto_test/voe_conference_test.cc
deleted file mode 100644
index 9e466af..0000000
--- a/voice_engine/test/auto_test/voe_conference_test.cc
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- *  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 <queue>
-
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
-
-namespace webrtc {
-namespace {
-
-const int kRttMs = 25;
-
-bool IsNear(int ref, int comp, int error) {
-  return (ref - comp <= error) && (comp - ref >= -error);
-}
-
-void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
-  FILE* fid = fopen(silence_file.c_str(), "wb");
-  int16_t zero = 0;
-  for (int i = 0; i < sample_rate_hz; ++i) {
-    // Write 1 second, but it does not matter since the file will be looped.
-    fwrite(&zero, sizeof(int16_t), 1, fid);
-  }
-  fclose(fid);
-}
-
-}  // namespace
-
-namespace voetest {
-
-TEST(VoeConferenceTest, RttAndStartNtpTime) {
-  struct Stats {
-    Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
-        : rtt_receiver_1_(rtt_receiver_1),
-          rtt_receiver_2_(rtt_receiver_2),
-          ntp_delay_(ntp_delay) {
-    }
-    int64_t rtt_receiver_1_;
-    int64_t rtt_receiver_2_;
-    int64_t ntp_delay_;
-  };
-
-  const std::string input_file =
-      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
-
-  const int kDelayMs = 987;
-  ConferenceTransport trans;
-  trans.SetRtt(kRttMs);
-
-  unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
-  unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
-
-  EXPECT_TRUE(trans.StartPlayout(id_1));
-  // Start NTP time is the time when a stream is played out, rather than
-  // when it is added.
-  webrtc::SleepMs(kDelayMs);
-  EXPECT_TRUE(trans.StartPlayout(id_2));
-
-  const int kMaxRunTimeMs = 25000;
-  const int kNeedSuccessivePass = 3;
-  const int kStatsRequestIntervalMs = 1000;
-  const int kStatsBufferSize = 3;
-
-  int64_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
-  // Run the following up to |kMaxRunTimeMs| milliseconds.
-  int successive_pass = 0;
-  webrtc::CallStatistics stats_1;
-  webrtc::CallStatistics stats_2;
-  std::queue<Stats> stats_buffer;
-
-  while (rtc::TimeMillis() < deadline &&
-         successive_pass < kNeedSuccessivePass) {
-    webrtc::SleepMs(kStatsRequestIntervalMs);
-
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
-
-    // It is not easy to verify the NTP time directly. We verify it by testing
-    // the difference of two start NTP times.
-    int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
-        stats_1.capture_start_ntp_time_ms_;
-
-    // For the checks of RTT and start NTP time, We allow 10% accuracy.
-    if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
-        IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
-        IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
-      successive_pass++;
-    } else {
-      successive_pass = 0;
-    }
-    if (stats_buffer.size() >= kStatsBufferSize) {
-      stats_buffer.pop();
-    }
-    stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
-                            captured_start_ntp_delay));
-  }
-
-  EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
-      " start NTP time estimate within 10% of the correct value over "
-      << kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
-      << " seconds.";
-  if (successive_pass < kNeedSuccessivePass) {
-    printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
-        "NTP delay between receiver 1 and 2) are (from oldest):\n");
-    while (!stats_buffer.empty()) {
-      Stats stats = stats_buffer.front();
-      printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
-             stats.rtt_receiver_2_, stats.ntp_delay_);
-      stats_buffer.pop();
-    }
-  }
-}
-
-
-TEST(VoeConferenceTest, ReceivedPackets) {
-  const int kPackets = 50;
-  const int kPacketDurationMs = 20;  // Correspond to Opus.
-
-  const std::string input_file =
-      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
-  const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
-
-  const std::string silence_file =
-      webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
-  CreateSilenceFile(silence_file, 32000);
-
-  {
-    ConferenceTransport trans;
-    // Add silence to stream 0, so that it will be filtered out.
-    unsigned int id_0 = trans.AddStream(silence_file, kInputFormat);
-    unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
-    unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
-    unsigned int id_3 = trans.AddStream(input_file, kInputFormat);
-
-    EXPECT_TRUE(trans.StartPlayout(id_0));
-    EXPECT_TRUE(trans.StartPlayout(id_1));
-    EXPECT_TRUE(trans.StartPlayout(id_2));
-    EXPECT_TRUE(trans.StartPlayout(id_3));
-
-    webrtc::SleepMs(kPacketDurationMs * kPackets);
-
-    webrtc::CallStatistics stats_0;
-    webrtc::CallStatistics stats_1;
-    webrtc::CallStatistics stats_2;
-    webrtc::CallStatistics stats_3;
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0));
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
-    EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3));
-
-    // We expect stream 0 to be filtered out totally, but since it may join the
-    // call earlier than other streams and the beginning packets might have got
-    // through. So we only expect |packetsReceived| to be close to zero.
-    EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
-    // We expect |packetsReceived| to match |kPackets|, but the actual value
-    // depends on the sleep timer. So we allow a small off from |kPackets|.
-    EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
-    EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
-    EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
-  }
-
-  remove(silence_file.c_str());
-}
-
-}  // namespace voetest
-}  // namespace webrtc