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44db436e87
actual task queue implementation for these tests is intentionally unchanged for now. while at it, change return type of created transports to unique_ptr to note passing ownership. Bug: webrtc:10933 Change-Id: I324597b503e647c471f43511340eb9c07ba03ee8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154743 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29335}
70 lines
2.3 KiB
C++
70 lines
2.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
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#define AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/task_queue/task_queue_base.h"
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#include "api/test/simulated_network.h"
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#include "test/call_test.h"
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namespace webrtc {
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namespace test {
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class AudioEndToEndTest : public test::EndToEndTest {
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public:
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AudioEndToEndTest();
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protected:
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TestAudioDeviceModule* send_audio_device() { return send_audio_device_; }
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const AudioSendStream* send_stream() const { return send_stream_; }
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const AudioReceiveStream* receive_stream() const { return receive_stream_; }
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virtual BuiltInNetworkBehaviorConfig GetNetworkPipeConfig() const;
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size_t GetNumVideoStreams() const override;
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size_t GetNumAudioStreams() const override;
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size_t GetNumFlexfecStreams() const override;
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std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
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std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override;
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void OnFakeAudioDevicesCreated(
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TestAudioDeviceModule* send_audio_device,
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TestAudioDeviceModule* recv_audio_device) override;
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std::unique_ptr<test::PacketTransport> CreateSendTransport(
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TaskQueueBase* task_queue,
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Call* sender_call) override;
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std::unique_ptr<test::PacketTransport> CreateReceiveTransport(
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TaskQueueBase* task_queue) override;
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void ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs) override;
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void OnAudioStreamsCreated(
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AudioSendStream* send_stream,
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const std::vector<AudioReceiveStream*>& receive_streams) override;
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void PerformTest() override;
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private:
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TestAudioDeviceModule* send_audio_device_ = nullptr;
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AudioSendStream* send_stream_ = nullptr;
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AudioReceiveStream* receive_stream_ = nullptr;
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};
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} // namespace test
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} // namespace webrtc
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#endif // AUDIO_TEST_AUDIO_END_TO_END_TEST_H_
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