1 files changed, 91 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc
new file mode 100644
index 0000000000..b7dfdb89f6
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+++ b/third_party/libwebrtc/audio/test/audio_end_to_end_test.cc
@@ -0,0 +1,91 @@
+/*
+ *  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 "audio/test/audio_end_to_end_test.h"
+
+#include <algorithm>
+#include <memory>
+
+#include "api/task_queue/task_queue_base.h"
+#include "call/fake_network_pipe.h"
+#include "call/simulated_network.h"
+#include "system_wrappers/include/sleep.h"
+#include "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::kDefaultTimeout) {}
+
+size_t AudioEndToEndTest::GetNumVideoStreams() const {
+  return 0;
+}
+
+size_t AudioEndToEndTest::GetNumAudioStreams() const {
+  return 1;
+}
+
+size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
+  return 0;
+}
+
+std::unique_ptr<TestAudioDeviceModule::Capturer>
+AudioEndToEndTest::CreateCapturer() {
+  return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate);
+}
+
+std::unique_ptr<TestAudioDeviceModule::Renderer>
+AudioEndToEndTest::CreateRenderer() {
+  return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate);
+}
+
+void AudioEndToEndTest::OnFakeAudioDevicesCreated(
+    TestAudioDeviceModule* send_audio_device,
+    TestAudioDeviceModule* recv_audio_device) {
+  send_audio_device_ = send_audio_device;
+}
+
+void AudioEndToEndTest::ModifyAudioConfigs(
+    AudioSendStream::Config* send_config,
+    std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {
+  // Large bitrate by default.
+  const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
+                                              {{"stereo", "1"}});
+  send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
+      test::CallTest::kAudioSendPayloadType, kDefaultFormat);
+  send_config->min_bitrate_bps = 32000;
+  send_config->max_bitrate_bps = 32000;
+}
+
+void AudioEndToEndTest::OnAudioStreamsCreated(
+    AudioSendStream* send_stream,
+    const std::vector<AudioReceiveStreamInterface*>& 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(GetSendTransportConfig().queue_delay_ms + kExtraRecordTimeMs);
+}
+}  // namespace test
+}  // namespace webrtc