webrtc_m130/video/end_to_end_tests/multi_stream_tests.cc
Sebastian Jansson bed801e005 Removes unnecessary destructor checks in tests.
Removes checks that are not relevant to the particular tests. The checks
create dependencies on the CallTest base class. This prepares for
further refactoring in CallTest.

Bug: webrtc:9510
Change-Id: Icd2bb4fe168aabd377d349b1de3de833fb7ae075
Reviewed-on: https://webrtc-review.googlesource.com/87823
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23902}
2018-07-10 11:29:45 +00:00

103 lines
3.4 KiB
C++

/*
* 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 "logging/rtc_event_log/rtc_event_log.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "test/call_test.h"
#include "test/encoder_settings.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "video/end_to_end_tests/multi_stream_tester.h"
namespace webrtc {
class MultiStreamEndToEndTest
: public test::CallTest,
public testing::WithParamInterface<std::string> {
public:
MultiStreamEndToEndTest() : field_trial_(GetParam()) {}
private:
test::ScopedFieldTrials field_trial_;
};
INSTANTIATE_TEST_CASE_P(RoundRobin,
MultiStreamEndToEndTest,
::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
"WebRTC-RoundRobinPacing/Enabled/"));
// Each renderer verifies that it receives the expected resolution, and as soon
// as every renderer has received a frame, the test finishes.
TEST_P(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) {
class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
public:
VideoOutputObserver(const MultiStreamTester::CodecSettings& settings,
uint32_t ssrc,
test::FrameGeneratorCapturer** frame_generator)
: settings_(settings),
ssrc_(ssrc),
frame_generator_(frame_generator),
done_(false, false) {}
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(kDefaultTimeoutMs); }
private:
const MultiStreamTester::CodecSettings& settings_;
const uint32_t ssrc_;
test::FrameGeneratorCapturer** const frame_generator_;
rtc::Event done_;
};
class Tester : public MultiStreamTester {
public:
explicit Tester(test::SingleThreadedTaskQueueForTesting* task_queue)
: MultiStreamTester(task_queue) {}
virtual ~Tester() {}
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].reset(new VideoOutputObserver(
codec_settings[stream_index], send_config->rtp.ssrcs.front(),
frame_generator));
}
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) override {
receive_config->renderer = observers_[stream_index].get();
}
private:
std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
} tester(&task_queue_);
tester.RunTest();
}
} // namespace webrtc