/* * Copyright 2018 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 TEST_SCENARIO_CALL_CLIENT_H_ #define TEST_SCENARIO_CALL_CLIENT_H_
namespace test { // Helper class to capture network controller state. class NetworkControleUpdateCache : public NetworkControllerInterface { public: explicit NetworkControleUpdateCache(
std::unique_ptr<NetworkControllerInterface> controller);
struct CallClientFakeAudio {
rtc::scoped_refptr<AudioProcessing> apm;
rtc::scoped_refptr<AudioDeviceModule> fake_audio_device;
rtc::scoped_refptr<AudioState> audio_state;
}; // CallClient represents a participant in a call scenario. It is created by the // Scenario class and is used as sender and receiver when setting up a media // stream session. class CallClient : public EmulatedNetworkReceiverInterface { public:
CallClient(TimeController* time_controller,
std::unique_ptr<LogWriterFactoryInterface> log_writer_factory,
CallClientConfig config);
// Exposed publicly so that tests can execute tasks such as querying stats // for media streams in the expected runtime environment (essentially what // CallClient does internally for GetStats()). void SendTask(std::function<void()> task);
int next_video_ssrc_index_ = 0; int next_video_local_ssrc_index_ = 0; int next_rtx_ssrc_index_ = 0; int next_audio_ssrc_index_ = 0; int next_audio_local_ssrc_index_ = 0;
std::map<uint32_t, MediaType> ssrc_media_types_; // Defined last so it's destroyed first.
TaskQueueForTest task_queue_;
};
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