Test and fix for huge bwe drop after alr state.

webrtc/audio/test/audio_bwe_integration_test.h

+/*
+ *  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.
+ */
+#ifndef WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
+#define WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/test/call_test.h"
+#include "webrtc/test/fake_audio_device.h"
+
+namespace webrtc {
+namespace test {
+
+class AudioBweTest : public test::EndToEndTest {
+ public:
+  AudioBweTest();
+
+ protected:
+  virtual std::string AudioInputFile() = 0;
+
+  virtual FakeNetworkPipe::Config GetNetworkPipeConfig() = 0;
+
+  size_t GetNumVideoStreams() const override;
+  size_t GetNumAudioStreams() const override;
+  size_t GetNumFlexfecStreams() const override;
+
+  std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override;
+
+  void OnFakeAudioDevicesCreated(
+      test::FakeAudioDevice* send_audio_device,
+      test::FakeAudioDevice* recv_audio_device) override;
+
+  test::PacketTransport* CreateSendTransport(Call* sender_call) override;
+  test::PacketTransport* CreateReceiveTransport() override;
+
+  void PerformTest() override;
+
+ private:
+  test::FakeAudioDevice* send_audio_device_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_