/* * 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. */ #include #include #include #include #include "api/video/video_frame.h" #include "api/video/video_sink_interface.h" #include "call/rtp_config.h" #include "call/video_receive_stream.h" #include "call/video_send_stream.h" #include "rtc_base/event.h" #include "test/frame_generator_capturer.h" #include "test/gtest.h" #include "video/config/video_encoder_config.h" #include "video/end_to_end_tests/multi_stream_tester.h" namespace webrtc { // Each renderer verifies that it receives the expected resolution, and as soon // as every renderer has received a frame, the test finishes. TEST(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) { class VideoOutputObserver : public rtc::VideoSinkInterface { public: VideoOutputObserver(const MultiStreamTester::CodecSettings& settings, uint32_t ssrc, test::FrameGeneratorCapturer** frame_generator) : settings_(settings), ssrc_(ssrc), frame_generator_(frame_generator) {} void OnFrame(const VideoFrame& video_frame) override { EXPECT_EQ(settings_.width, video_frame.width()); EXPECT_EQ(settings_.height, video_frame.height()); (*frame_generator_)->Stop(); done_.Set(); } uint32_t Ssrc() { return ssrc_; } bool Wait() { return done_.Wait(TimeDelta::Seconds(30)); } private: const MultiStreamTester::CodecSettings& settings_; const uint32_t ssrc_; test::FrameGeneratorCapturer** const frame_generator_; rtc::Event done_; }; class Tester : public MultiStreamTester { public: Tester() = default; ~Tester() override = default; protected: void Wait() override { for (const auto& observer : observers_) { EXPECT_TRUE(observer->Wait()) << "Time out waiting for from on ssrc " << observer->Ssrc(); } } void UpdateSendConfig( size_t stream_index, VideoSendStream::Config* send_config, VideoEncoderConfig* encoder_config, test::FrameGeneratorCapturer** frame_generator) override { observers_[stream_index] = std::make_unique( codec_settings[stream_index], send_config->rtp.ssrcs.front(), frame_generator); } void UpdateReceiveConfig( size_t stream_index, VideoReceiveStreamInterface::Config* receive_config) override { receive_config->renderer = observers_[stream_index].get(); } private: std::unique_ptr observers_[kNumStreams]; } tester; tester.RunTest(); } } // namespace webrtc