diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index ff38da0d80..4a3d8fd032 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -177,18 +177,27 @@ if (rtc_include_tests) {
 
       sources = [
         "test/low_bandwidth_audio_test.cc",
+        "test/pc_low_bandwidth_audio_test.cc",
       ]
 
       deps = [
         ":audio_end_to_end_test",
+        "../api:create_network_emulation_manager",
+        "../api:create_peerconnection_quality_test_fixture",
+        "../api:network_emulation_manager_api",
+        "../api:peer_connection_quality_test_fixture_api",
         "../api:simulated_network_api",
+        "../call:simulated_network",
         "../common_audio",
         "../rtc_base:rtc_base_approved",
         "../system_wrappers",
         "../test:fileutils",
         "../test:test_common",
         "../test:test_main",
+        "../test:test_support",
+        "../test/pc/e2e:network_quality_metrics_reporter",
         "//testing/gtest",
+        "//third_party/abseil-cpp/absl/memory:memory",
       ]
       if (is_android) {
         deps += [ "//testing/android/native_test:native_test_native_code" ]
diff --git a/audio/test/pc_low_bandwidth_audio_test.cc b/audio/test/pc_low_bandwidth_audio_test.cc
new file mode 100644
index 0000000000..d520f10f09
--- /dev/null
+++ b/audio/test/pc_low_bandwidth_audio_test.cc
@@ -0,0 +1,148 @@
+/*
+ *  Copyright (c) 2019 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 "absl/memory/memory.h"
+#include "api/test/create_network_emulation_manager.h"
+#include "api/test/create_peerconnection_quality_test_fixture.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/test/peerconnection_quality_test_fixture.h"
+#include "api/test/simulated_network.h"
+#include "call/simulated_network.h"
+#include "rtc_base/flags.h"
+#include "test/gtest.h"
+#include "test/pc/e2e/network_quality_metrics_reporter.h"
+#include "test/testsupport/file_utils.h"
+
+namespace webrtc {
+namespace test {
+
+using PeerConfigurer =
+    webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::PeerConfigurer;
+using RunParams = webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::RunParams;
+using AudioConfig =
+    webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture::AudioConfig;
+
+namespace {
+
+constexpr int kTestDurationSec = 45;
+
+EmulatedNetworkNode* CreateEmulatedNodeWithConfig(
+    NetworkEmulationManager* emulation,
+    const BuiltInNetworkBehaviorConfig& config) {
+  return emulation->CreateEmulatedNode(
+      absl::make_unique<SimulatedNetwork>(config));
+}
+
+std::pair<EmulatedNetworkManagerInterface*, EmulatedNetworkManagerInterface*>
+CreateTwoNetworkLinks(NetworkEmulationManager* emulation,
+                      const BuiltInNetworkBehaviorConfig& config) {
+  auto* alice_node = CreateEmulatedNodeWithConfig(emulation, config);
+  auto* bob_node = CreateEmulatedNodeWithConfig(emulation, config);
+
+  auto* alice_endpoint = emulation->CreateEndpoint(EmulatedEndpointConfig());
+  auto* bob_endpoint = emulation->CreateEndpoint(EmulatedEndpointConfig());
+
+  emulation->CreateRoute(alice_endpoint, {alice_node}, bob_endpoint);
+  emulation->CreateRoute(bob_endpoint, {bob_node}, alice_endpoint);
+
+  return {
+      emulation->CreateEmulatedNetworkManagerInterface({alice_endpoint}),
+      emulation->CreateEmulatedNetworkManagerInterface({bob_endpoint}),
+  };
+}
+
+std::unique_ptr<webrtc_pc_e2e::PeerConnectionE2EQualityTestFixture>
+CreateTestFixture(const std::string& test_case_name,
+                  std::pair<EmulatedNetworkManagerInterface*,
+                            EmulatedNetworkManagerInterface*> network_links,
+                  rtc::FunctionView<void(PeerConfigurer*)> alice_configurer,
+                  rtc::FunctionView<void(PeerConfigurer*)> bob_configurer) {
+  auto fixture = webrtc_pc_e2e::CreatePeerConnectionE2EQualityTestFixture(
+      test_case_name, /*audio_quality_analyzer=*/nullptr,
+      /*video_quality_analyzer=*/nullptr);
+  fixture->AddPeer(network_links.first->network_thread(),
+                   network_links.first->network_manager(), alice_configurer);
+  fixture->AddPeer(network_links.second->network_thread(),
+                   network_links.second->network_manager(), bob_configurer);
+  fixture->AddQualityMetricsReporter(
+      absl::make_unique<webrtc_pc_e2e::NetworkQualityMetricsReporter>(
+          network_links.first, network_links.second));
+  return fixture;
+}
+
+std::string AudioInputFile() {
+  return test::ResourcePath("voice_engine/audio_tiny48", "wav");
+}
+
+std::string AudioOutputFile() {
+  const ::testing::TestInfo* const test_info =
+      ::testing::UnitTest::GetInstance()->current_test_info();
+  return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
+         "_48.wav";
+}
+
+void PrintTestInfo() {
+  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());
+}
+
+}  // namespace
+
+TEST(PCLowBandwidthAudioTest, GoodNetworkHighBitrate) {
+  std::unique_ptr<NetworkEmulationManager> network_emulation_manager =
+      CreateNetworkEmulationManager();
+  auto fixture = CreateTestFixture(
+      "pc_good_network",
+      CreateTwoNetworkLinks(network_emulation_manager.get(),
+                            BuiltInNetworkBehaviorConfig()),
+      [](PeerConfigurer* alice) {
+        AudioConfig audio;
+        audio.stream_label = "alice-audio";
+        audio.mode = AudioConfig::Mode::kFile;
+        audio.input_file_name = AudioInputFile();
+        audio.output_dump_file_name = AudioOutputFile();
+        alice->SetAudioConfig(std::move(audio));
+      },
+      [](PeerConfigurer* bob) {});
+  fixture->Run(RunParams(TimeDelta::seconds(kTestDurationSec)));
+  PrintTestInfo();
+}
+
+TEST(PCLowBandwidthAudioTest, Mobile2GNetwork) {
+  std::unique_ptr<NetworkEmulationManager> network_emulation_manager =
+      CreateNetworkEmulationManager();
+  BuiltInNetworkBehaviorConfig config;
+  config.link_capacity_kbps = 12;
+  config.queue_length_packets = 1500;
+  config.queue_delay_ms = 400;
+  auto fixture = CreateTestFixture(
+      "pc_mobile_2g_network",
+      CreateTwoNetworkLinks(network_emulation_manager.get(), config),
+      [](PeerConfigurer* alice) {
+        AudioConfig audio;
+        audio.stream_label = "alice-audio";
+        audio.mode = AudioConfig::Mode::kFile;
+        audio.input_file_name = AudioInputFile();
+        audio.output_dump_file_name = AudioOutputFile();
+        alice->SetAudioConfig(std::move(audio));
+      },
+      [](PeerConfigurer* bob) {});
+  RunParams run_params(TimeDelta::seconds(kTestDurationSec));
+  fixture->Run(RunParams(TimeDelta::seconds(kTestDurationSec)));
+  PrintTestInfo();
+}
+
+}  // namespace test
+}  // namespace webrtc