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9a99905301
This emulates behaviour from frame buffer 2, but does not handle stats. In contrast to frame buffer 2, all work happens on the same task queue. FrameBuffer3Proxy encapsulates FrameBuffer3 and scheduler behind a field trial WebRTC-FrameBuffer3. This separates frame scheduling behaviour into a few components, VideoReceiveStreamTimeoutTracker * Handles the stream timeouts. FrameDecodeScheduler * Manages the scheduling and cancelling of frames being sent to the decoder. FrameDecodeTiming * Handles the timing and ordering of frames to be decoded. Other changes * Adds CurrentSize() method to FrameBuffer3 * Move timing to a separate library * Does a thread check for Receive statistics as this is now on the worker thread. * Adds `FlushImmediate` method to RunLoop so that video_receive_stream2_unittest can pass when scheduling is happening on the worker thread. Change-Id: Ia8d2e5650d1708cdc1be3631a5214134583a0721 Bug: webrtc:13343 Tested: Ran webrtc_perf_tests, video_engine_tests, rtc_unittests forcing frame buffer3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241603 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Markus Handell <handellm@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Commit-Queue: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35847}
74 lines
2 KiB
C++
74 lines
2 KiB
C++
/*
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* Copyright (c) 2013 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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#include "test/run_loop.h"
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#include "rtc_base/task_utils/to_queued_task.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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namespace test {
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RunLoop::RunLoop() {
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worker_thread_.WrapCurrent();
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}
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RunLoop::~RunLoop() {
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worker_thread_.UnwrapCurrent();
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}
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TaskQueueBase* RunLoop::task_queue() {
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return &worker_thread_;
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}
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void RunLoop::Run() {
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worker_thread_.ProcessMessages(WorkerThread::kForever);
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}
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void RunLoop::Quit() {
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socket_server_.FailNextWait();
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}
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void RunLoop::Flush() {
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worker_thread_.PostTask(
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ToQueuedTask([this]() { socket_server_.FailNextWait(); }));
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// If a test clock is used, like with GlobalSimulatedTimeController then the
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// thread will loop forever since time never increases. Since the clock is
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// simulated, 0ms can be used as the loop delay, which will process all
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// messages ready for execution.
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int cms = rtc::GetClockForTesting() ? 0 : 1000;
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worker_thread_.ProcessMessages(cms);
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}
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RunLoop::FakeSocketServer::FakeSocketServer() = default;
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RunLoop::FakeSocketServer::~FakeSocketServer() = default;
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void RunLoop::FakeSocketServer::FailNextWait() {
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fail_next_wait_ = true;
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}
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bool RunLoop::FakeSocketServer::Wait(int cms, bool process_io) {
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if (fail_next_wait_) {
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fail_next_wait_ = false;
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return false;
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}
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return true;
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}
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void RunLoop::FakeSocketServer::WakeUp() {}
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rtc::Socket* RunLoop::FakeSocketServer::CreateSocket(int family, int type) {
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return nullptr;
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}
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RunLoop::WorkerThread::WorkerThread(rtc::SocketServer* ss)
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: rtc::Thread(ss), tq_setter_(this) {}
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} // namespace test
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} // namespace webrtc
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