413ee9a010
DirectTransport has so far used its own thread, which led to a different threading-model for in the unit-tests than is used in actual WebRTC. Because of that, some critical-sections that weren't truly necessary in WebRTC could not be replaced with thread-checks, because those checks failed in unit-tests. This CL introduces SingleThreadedTaskQueue - a TaskQueue which guarantees to run all of its tasks on the same thread (rtc::TaskQueue doesn't guarantee that on Mac) - and uses that for DirectTransport. CLs based on top of this will uncomment thread-checks which had to be commented out before, and remove unnecessary critical-sections. Future work would probably replace the thread-checkers by more sophisticated serialized-access checks, allowing us to move from the SingleThreadedTaskQueue to a normal TaskQueue. Related implementation notes: * This CL has made DirectTransport::StopSending() superfluous, and so it was deleted. BUG=webrtc:8113, webrtc:7405, webrtc:8056, webrtc:8116 Review-Url: https://codereview.webrtc.org/2998923002 Cr-Commit-Position: refs/heads/master@{#19445}
58 lines
1.7 KiB
C++
58 lines
1.7 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 WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
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#define WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
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#include <memory>
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#include <string>
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#include "webrtc/test/call_test.h"
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#include "webrtc/test/fake_audio_device.h"
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#include "webrtc/test/single_threaded_task_queue.h"
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namespace webrtc {
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namespace test {
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class AudioBweTest : public test::EndToEndTest {
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public:
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AudioBweTest();
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protected:
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virtual std::string AudioInputFile() = 0;
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virtual FakeNetworkPipe::Config GetNetworkPipeConfig() = 0;
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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<test::FakeAudioDevice::Capturer> CreateCapturer() override;
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void OnFakeAudioDevicesCreated(
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test::FakeAudioDevice* send_audio_device,
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test::FakeAudioDevice* recv_audio_device) override;
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test::PacketTransport* CreateSendTransport(
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SingleThreadedTaskQueueForTesting* task_queue,
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Call* sender_call) override;
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test::PacketTransport* CreateReceiveTransport(
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SingleThreadedTaskQueueForTesting* task_queue) override;
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void PerformTest() override;
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private:
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test::FakeAudioDevice* send_audio_device_;
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};
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} // namespace test
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} // namespace webrtc
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#endif // WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
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