a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
101 lines
3.2 KiB
C++
101 lines
3.2 KiB
C++
/*
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* Copyright 2018 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 <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include <vector>
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#include "api/video/video_frame.h"
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#include "api/video/video_sink_interface.h"
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#include "api/video_codecs/video_encoder_config.h"
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#include "call/rtp_config.h"
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#include "call/video_receive_stream.h"
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#include "call/video_send_stream.h"
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#include "rtc_base/event.h"
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#include "test/call_test.h"
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#include "test/frame_generator_capturer.h"
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#include "test/gtest.h"
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#include "test/single_threaded_task_queue.h"
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#include "video/end_to_end_tests/multi_stream_tester.h"
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namespace webrtc {
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class MultiStreamEndToEndTest : public test::CallTest {
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public:
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MultiStreamEndToEndTest() = default;
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};
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// Each renderer verifies that it receives the expected resolution, and as soon
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// as every renderer has received a frame, the test finishes.
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TEST_F(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) {
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class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
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public:
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VideoOutputObserver(const MultiStreamTester::CodecSettings& settings,
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uint32_t ssrc,
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test::FrameGeneratorCapturer** frame_generator)
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: settings_(settings), ssrc_(ssrc), frame_generator_(frame_generator) {}
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void OnFrame(const VideoFrame& video_frame) override {
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EXPECT_EQ(settings_.width, video_frame.width());
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EXPECT_EQ(settings_.height, video_frame.height());
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(*frame_generator_)->Stop();
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done_.Set();
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}
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uint32_t Ssrc() { return ssrc_; }
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bool Wait() { return done_.Wait(kDefaultTimeoutMs); }
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private:
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const MultiStreamTester::CodecSettings& settings_;
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const uint32_t ssrc_;
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test::FrameGeneratorCapturer** const frame_generator_;
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rtc::Event done_;
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};
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class Tester : public MultiStreamTester {
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public:
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explicit Tester(test::SingleThreadedTaskQueueForTesting* task_queue)
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: MultiStreamTester(task_queue) {}
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virtual ~Tester() {}
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protected:
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void Wait() override {
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for (const auto& observer : observers_) {
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EXPECT_TRUE(observer->Wait())
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<< "Time out waiting for from on ssrc " << observer->Ssrc();
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}
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}
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void UpdateSendConfig(
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size_t stream_index,
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VideoSendStream::Config* send_config,
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VideoEncoderConfig* encoder_config,
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test::FrameGeneratorCapturer** frame_generator) override {
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observers_[stream_index].reset(new VideoOutputObserver(
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codec_settings[stream_index], send_config->rtp.ssrcs.front(),
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frame_generator));
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}
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void UpdateReceiveConfig(
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size_t stream_index,
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VideoReceiveStream::Config* receive_config) override {
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receive_config->renderer = observers_[stream_index].get();
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}
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private:
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std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
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} tester(&task_queue_);
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tester.RunTest();
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}
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} // namespace webrtc
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