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webrtc_m130/video/end_to_end_tests/call_operation_tests.cc
Sergey Silkin 1ad210876f Revert "Storing frame if encoder is paused." 
This reverts commit dcc7e88cc79ab4f7aeb87c13f402e007e1320fd8.

Reason for revert: breaks downstream projects

Original change's description:
> Storing frame if encoder is paused.
> 
> Adds a pending frame to VideoStreamEncoder that is used to store frames
> that are not sent because the encoder is paused. If the encoder is
> resumed within 200 ms, the pending frame will be encoded and sent. This
> ensures that resuming a stream instantly starts sending frames if it is
> possible.
> 
> This also protects against a race between submitting the first frame
> and enabling the encoder that caused flakiness in end to end tests
> when using the task queue based congestion controller.
> 
> Bug: webrtc:8415
> Change-Id: If4bd897187fbfdc4926855f39503230bdad4a93a
> Reviewed-on: https://webrtc-review.googlesource.com/67141
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22781}

TBR=sprang@webrtc.org,srte@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8415
Change-Id: I4449eca65a64e2bc2fb25d866e5775e9a085cee9
Reviewed-on: https://webrtc-review.googlesource.com/68280
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22788}
2018-04-09 10:01:13 +00:00

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/*
* 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 "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/frame_generator.h"
#include "test/gtest.h"
#include "test/null_transport.h"
namespace webrtc {
class CallOperationEndToEndTest
: public test::CallTest,
public testing::WithParamInterface<std::string> {
public:
CallOperationEndToEndTest() : field_trial_(GetParam()) {}
virtual ~CallOperationEndToEndTest() {
EXPECT_EQ(nullptr, video_send_stream_);
EXPECT_TRUE(video_receive_streams_.empty());
}
private:
test::ScopedFieldTrials field_trial_;
};
INSTANTIATE_TEST_CASE_P(
FieldTrials,
CallOperationEndToEndTest,
::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
"WebRTC-RoundRobinPacing/Enabled/",
"WebRTC-TaskQueueCongestionControl/Enabled/"));
TEST_P(CallOperationEndToEndTest, ReceiverCanBeStartedTwice) {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateVideoStreams();
video_receive_streams_[0]->Start();
video_receive_streams_[0]->Start();
DestroyStreams();
}
TEST_P(CallOperationEndToEndTest, ReceiverCanBeStoppedTwice) {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateVideoStreams();
video_receive_streams_[0]->Stop();
video_receive_streams_[0]->Stop();
DestroyStreams();
}
TEST_P(CallOperationEndToEndTest, ReceiverCanBeStoppedAndRestarted) {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateVideoStreams();
video_receive_streams_[0]->Stop();
video_receive_streams_[0]->Start();
video_receive_streams_[0]->Stop();
DestroyStreams();
}
TEST_P(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
static const int kWidth = 320;
static const int kHeight = 240;
// This constant is chosen to be higher than the timeout in the video_render
// module. This makes sure that frames aren't dropped if there are no other
// frames in the queue.
static const int kRenderDelayMs = 1000;
class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
public:
Renderer() : event_(false, false) {}
void OnFrame(const VideoFrame& video_frame) override {
SleepMs(kRenderDelayMs);
event_.Set();
}
bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
rtc::Event event_;
} renderer;
test::FrameForwarder frame_forwarder;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &renderer, &frame_forwarder, &sender_transport,
&receiver_transport]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_receive_configs_[0].renderer = &renderer;
CreateVideoStreams();
Start();
// Create frames that are smaller than the send width/height, this is done
// to check that the callbacks are done after processing video.
std::unique_ptr<test::FrameGenerator> frame_generator(
test::FrameGenerator::CreateSquareGenerator(
kWidth, kHeight, rtc::nullopt, rtc::nullopt));
video_send_stream_->SetSource(
&frame_forwarder,
VideoSendStream::DegradationPreference::kMaintainFramerate);
frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(renderer.Wait())
<< "Timed out while waiting for the frame to render.";
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
// TODO(bugs.webrtc.org/9060): Re-enable this test with the TaskQueue when it
// is no longer flaky.
class CallOperationEndToEndTestNoTaskQueueCongestionControl
: public CallOperationEndToEndTest {};
INSTANTIATE_TEST_CASE_P(FieldTrials,
CallOperationEndToEndTestNoTaskQueueCongestionControl,
::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
"WebRTC-RoundRobinPacing/Enabled/"));
TEST_P(CallOperationEndToEndTestNoTaskQueueCongestionControl,
TransmitsFirstFrame) {
class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
public:
Renderer() : event_(false, false) {}
void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
rtc::Event event_;
} renderer;
std::unique_ptr<test::FrameGenerator> frame_generator;
test::FrameForwarder frame_forwarder;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &renderer, &frame_generator, &frame_forwarder,
&sender_transport, &receiver_transport]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_receive_configs_[0].renderer = &renderer;
CreateVideoStreams();
Start();
frame_generator = test::FrameGenerator::CreateSquareGenerator(
kDefaultWidth, kDefaultHeight, rtc::nullopt, rtc::nullopt);
video_send_stream_->SetSource(
&frame_forwarder,
VideoSendStream::DegradationPreference::kMaintainFramerate);
frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(renderer.Wait())
<< "Timed out while waiting for the frame to render.";
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
// TODO(bugs.webrtc.org/9060): Re-enable this test with the TaskQueue when it
// is no longer flaky.
TEST_P(CallOperationEndToEndTestNoTaskQueueCongestionControl,
ObserversEncodedFrames) {
class EncodedFrameTestObserver : public EncodedFrameObserver {
public:
EncodedFrameTestObserver()
: length_(0), frame_type_(kEmptyFrame), called_(false, false) {}
virtual ~EncodedFrameTestObserver() {}
virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
frame_type_ = encoded_frame.frame_type_;
length_ = encoded_frame.length_;
buffer_.reset(new uint8_t[length_]);
memcpy(buffer_.get(), encoded_frame.data_, length_);
called_.Set();
}
bool Wait() { return called_.Wait(kDefaultTimeoutMs); }
void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
ASSERT_EQ(length_, observer.length_)
<< "Observed frames are of different lengths.";
EXPECT_EQ(frame_type_, observer.frame_type_)
<< "Observed frames have different frame types.";
EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_))
<< "Observed encoded frames have different content.";
}
private:
std::unique_ptr<uint8_t[]> buffer_;
size_t length_;
FrameType frame_type_;
rtc::Event called_;
};
EncodedFrameTestObserver post_encode_observer;
EncodedFrameTestObserver pre_decode_observer;
test::FrameForwarder forwarder;
std::unique_ptr<test::FrameGenerator> frame_generator;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([&]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_send_config_.post_encode_callback = &post_encode_observer;
video_receive_configs_[0].pre_decode_callback = &pre_decode_observer;
CreateVideoStreams();
Start();
frame_generator = test::FrameGenerator::CreateSquareGenerator(
kDefaultWidth, kDefaultHeight, rtc::nullopt, rtc::nullopt);
video_send_stream_->SetSource(
&forwarder, VideoSendStream::DegradationPreference::kMaintainFramerate);
forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(post_encode_observer.Wait())
<< "Timed out while waiting for send-side encoded-frame callback.";
EXPECT_TRUE(pre_decode_observer.Wait())
<< "Timed out while waiting for pre-decode encoded-frame callback.";
post_encode_observer.ExpectEqualFrames(pre_decode_observer);
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
} // namespace webrtc
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