9668467d87
Running with a network thread provides a more realistic simulation. Like a real network, packets are handed off to a socket, or buffer, and then the call returns. This prevents weird scenarios when both the sending side and receiving side are on the call stack simultaneously, which can cause deadlocks as locks could otherwise be taken simultaneously in both the sender and receiver order by the same thread. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2000005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4522 4adac7df-926f-26a2-2b94-8c16560cd09d
105 lines
3.7 KiB
C++
105 lines
3.7 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 <stdio.h>
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#include <map>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/system_wrappers/interface/clock.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/typedefs.h"
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#include "webrtc/video_engine/new_include/video_engine.h"
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#include "webrtc/video_engine/test/common/direct_transport.h"
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#include "webrtc/video_engine/test/common/flags.h"
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#include "webrtc/video_engine/test/common/generate_ssrcs.h"
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#include "webrtc/video_engine/test/common/run_loop.h"
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#include "webrtc/video_engine/test/common/run_tests.h"
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#include "webrtc/video_engine/test/common/video_capturer.h"
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#include "webrtc/video_engine/test/common/video_renderer.h"
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namespace webrtc {
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class LoopbackTest : public ::testing::Test {
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protected:
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std::map<uint32_t, bool> reserved_ssrcs_;
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};
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TEST_F(LoopbackTest, Test) {
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scoped_ptr<test::VideoRenderer> local_preview(test::VideoRenderer::Create(
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"Local Preview", test::flags::Width(), test::flags::Height()));
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scoped_ptr<test::VideoRenderer> loopback_video(test::VideoRenderer::Create(
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"Loopback Video", test::flags::Width(), test::flags::Height()));
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scoped_ptr<newapi::VideoEngine> video_engine(
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newapi::VideoEngine::Create(webrtc::newapi::VideoEngineConfig()));
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test::DirectTransport transport;
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newapi::VideoCall::Config call_config;
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call_config.send_transport = &transport;
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call_config.overuse_detection = true;
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scoped_ptr<newapi::VideoCall> call(video_engine->CreateCall(call_config));
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// Loopback, call sends to itself.
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transport.SetReceiver(call->Receiver());
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newapi::VideoSendStream::Config send_config = call->GetDefaultSendConfig();
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test::GenerateRandomSsrcs(&send_config, &reserved_ssrcs_);
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send_config.local_renderer = local_preview.get();
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// TODO(pbos): static_cast shouldn't be required after mflodman refactors the
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// VideoCodec struct.
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send_config.codec.width = static_cast<uint16_t>(test::flags::Width());
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send_config.codec.height = static_cast<uint16_t>(test::flags::Height());
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send_config.codec.minBitrate =
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static_cast<unsigned int>(test::flags::MinBitrate());
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send_config.codec.startBitrate =
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static_cast<unsigned int>(test::flags::StartBitrate());
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send_config.codec.maxBitrate =
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static_cast<unsigned int>(test::flags::MaxBitrate());
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newapi::VideoSendStream* send_stream = call->CreateSendStream(send_config);
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Clock* test_clock = Clock::GetRealTimeClock();
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scoped_ptr<test::VideoCapturer> camera(
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test::VideoCapturer::Create(send_stream->Input(),
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test::flags::Width(),
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test::flags::Height(),
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test::flags::Fps(),
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test_clock));
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newapi::VideoReceiveStream::Config receive_config =
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call->GetDefaultReceiveConfig();
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receive_config.rtp.ssrc = send_config.rtp.ssrcs[0];
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receive_config.renderer = loopback_video.get();
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newapi::VideoReceiveStream* receive_stream =
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call->CreateReceiveStream(receive_config);
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receive_stream->StartReceive();
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send_stream->StartSend();
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camera->Start();
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test::PressEnterToContinue();
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camera->Stop();
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send_stream->StopSend();
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receive_stream->StopReceive();
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call->DestroyReceiveStream(receive_stream);
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call->DestroySendStream(send_stream);
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transport.StopSending();
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}
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} // webrtc
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