1 files changed, 109 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/test/low_bandwidth_audio_test.cc b/third_party/libwebrtc/audio/test/low_bandwidth_audio_test.cc
new file mode 100644
index 0000000000..f385eb9dcc
--- /dev/null
+++ b/third_party/libwebrtc/audio/test/low_bandwidth_audio_test.cc
@@ -0,0 +1,109 @@
+/*
+ *  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 "absl/flags/declare.h"
+#include "absl/flags/flag.h"
+#include "api/test/simulated_network.h"
+#include "audio/test/audio_end_to_end_test.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_DECLARE_FLAG(int, sample_rate_hz);
+ABSL_DECLARE_FLAG(bool, quick);
+
+namespace webrtc {
+namespace test {
+namespace {
+
+std::string FileSampleRateSuffix() {
+  return std::to_string(absl::GetFlag(FLAGS_sample_rate_hz) / 1000);
+}
+
+class AudioQualityTest : public AudioEndToEndTest {
+ public:
+  AudioQualityTest() = default;
+
+ private:
+  std::string AudioInputFile() const {
+    return test::ResourcePath(
+        "voice_engine/audio_tiny" + FileSampleRateSuffix(), "wav");
+  }
+
+  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";
+  }
+
+  std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override {
+    return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
+  }
+
+  std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override {
+    return TestAudioDeviceModule::CreateBoundedWavFileWriter(
+        AudioOutputFile(), absl::GetFlag(FLAGS_sample_rate_hz));
+  }
+
+  void PerformTest() override {
+    if (absl::GetFlag(FLAGS_quick)) {
+      // Let the recording run for a small amount of time to check if it works.
+      SleepMs(1000);
+    } else {
+      AudioEndToEndTest::PerformTest();
+    }
+  }
+
+  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,
+                          std::vector<AudioReceiveStreamInterface::Config>*
+                              receive_configs) override {
+    send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+        test::CallTest::kAudioSendPayloadType,
+        {"OPUS",
+         48000,
+         2,
+         {{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
+  }
+
+  BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
+    BuiltInNetworkBehaviorConfig pipe_config;
+    pipe_config.link_capacity_kbps = 12;
+    pipe_config.queue_length_packets = 1500;
+    pipe_config.queue_delay_ms = 400;
+    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