Replace voe_conference_test.

webrtc/audio/test/audio_end_to_end_test.cc

Index: webrtc/audio/test/audio_end_to_end_test.cc
diff --git a/webrtc/audio/test/audio_end_to_end_test.cc b/webrtc/audio/test/audio_end_to_end_test.cc
new file mode 100644
index 0000000000000000000000000000000000000000..5d4cbf024a4f5e0cf25146cd8ebaa198225211d9
--- /dev/null
+++ b/webrtc/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