blob: 4e0e86f987ae79bae74903114a776db79832f385 [file] [log] [blame]
/*
* 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 <memory>
#include "api/media_types.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_base.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/test/simulated_network.h"
#include "api/video_codecs/video_encoder.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "rtc_base/location.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/fake_encoder.h"
#include "test/gtest.h"
#include "test/video_encoder_proxy_factory.h"
namespace webrtc {
namespace {
constexpr int kSilenceTimeoutMs = 2000;
}
class NetworkStateEndToEndTest : public test::CallTest {
protected:
class UnusedTransport : public Transport {
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
ADD_FAILURE() << "Unexpected RTP sent.";
return false;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
ADD_FAILURE() << "Unexpected RTCP sent.";
return false;
}
};
class RequiredTransport : public Transport {
public:
RequiredTransport(bool rtp_required, bool rtcp_required)
: need_rtp_(rtp_required), need_rtcp_(rtcp_required) {}
~RequiredTransport() {
if (need_rtp_) {
ADD_FAILURE() << "Expected RTP packet not sent.";
}
if (need_rtcp_) {
ADD_FAILURE() << "Expected RTCP packet not sent.";
}
}
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
MutexLock lock(&mutex_);
need_rtp_ = false;
return true;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
need_rtcp_ = false;
return true;
}
bool need_rtp_;
bool need_rtcp_;
Mutex mutex_;
};
void VerifyNewVideoSendStreamsRespectNetworkState(
MediaType network_to_bring_up,
VideoEncoder* encoder,
Transport* transport);
void VerifyNewVideoReceiveStreamsRespectNetworkState(
MediaType network_to_bring_up,
Transport* transport);
};
void NetworkStateEndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
MediaType network_to_bring_up,
VideoEncoder* encoder,
Transport* transport) {
test::VideoEncoderProxyFactory encoder_factory(encoder);
SendTask(RTC_FROM_HERE, task_queue(),
[this, network_to_bring_up, &encoder_factory, transport]() {
CreateSenderCall(Call::Config(send_event_log_.get()));
sender_call_->SignalChannelNetworkState(network_to_bring_up,
kNetworkUp);
CreateSendConfig(1, 0, 0, transport);
GetVideoSendConfig()->encoder_settings.encoder_factory =
&encoder_factory;
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
SleepMs(kSilenceTimeoutMs);
SendTask(RTC_FROM_HERE, task_queue(), [this]() {
Stop();
DestroyStreams();
DestroyCalls();
});
}
void NetworkStateEndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
MediaType network_to_bring_up,
Transport* transport) {
std::unique_ptr<test::DirectTransport> sender_transport;
SendTask(
RTC_FROM_HERE, task_queue(),
[this, &sender_transport, network_to_bring_up, transport]() {
CreateCalls();
receiver_call_->SignalChannelNetworkState(network_to_bring_up,
kNetworkUp);
sender_transport = std::make_unique<test::DirectTransport>(
task_queue(),
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(), std::make_unique<SimulatedNetwork>(
BuiltInNetworkBehaviorConfig())),
sender_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(transport);
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
SleepMs(kSilenceTimeoutMs);
SendTask(RTC_FROM_HERE, task_queue(), [this, &sender_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
DestroyCalls();
});
}
TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
// TODO(pbos): Remove accepted downtime packets etc. when signaling network
// down blocks until no more packets will be sent.
// Pacer will send from its packet list and then send required padding before
// checking paused_ again. This should be enough for one round of pacing,
// otherwise increase.
static const int kNumAcceptedDowntimeRtp = 5;
// A single RTCP may be in the pipeline.
static const int kNumAcceptedDowntimeRtcp = 1;
class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
public:
explicit NetworkStateTest(TaskQueueBase* task_queue)
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
e2e_test_task_queue_(task_queue),
task_queue_(CreateDefaultTaskQueueFactory()->CreateTaskQueue(
"NetworkStateTest",
TaskQueueFactory::Priority::NORMAL)),
sender_call_(nullptr),
receiver_call_(nullptr),
encoder_factory_(this),
sender_state_(kNetworkUp),
sender_rtp_(0),
sender_padding_(0),
sender_rtcp_(0),
receiver_rtcp_(0),
down_frames_(0) {}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
MutexLock lock(&test_mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet, length));
if (rtp_packet.payload_size() == 0)
++sender_padding_;
++sender_rtp_;
packet_event_.Set();
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&test_mutex_);
++sender_rtcp_;
packet_event_.Set();
return SEND_PACKET;
}
Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&test_mutex_);
++receiver_rtcp_;
packet_event_.Set();
return SEND_PACKET;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder_factory = &encoder_factory_;
}
void SignalChannelNetworkState(Call* call,
MediaType media_type,
NetworkState network_state) {
SendTask(RTC_FROM_HERE, e2e_test_task_queue_,
[call, media_type, network_state] {
call->SignalChannelNetworkState(media_type, network_state);
});
}
void PerformTest() override {
EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
<< "No frames received by the encoder.";
SendTask(RTC_FROM_HERE, task_queue_.get(), [this]() {
// Wait for packets from both sender/receiver.
WaitForPacketsOrSilence(false, false);
// Sender-side network down for audio; there should be no effect on
// video
SignalChannelNetworkState(sender_call_, MediaType::AUDIO, kNetworkDown);
WaitForPacketsOrSilence(false, false);
// Receiver-side network down for audio; no change expected
SignalChannelNetworkState(receiver_call_, MediaType::AUDIO,
kNetworkDown);
WaitForPacketsOrSilence(false, false);
// Sender-side network down.
SignalChannelNetworkState(sender_call_, MediaType::VIDEO, kNetworkDown);
{
MutexLock lock(&test_mutex_);
// After network goes down we shouldn't be encoding more frames.
sender_state_ = kNetworkDown;
}
// Wait for receiver-packets and no sender packets.
WaitForPacketsOrSilence(true, false);
// Receiver-side network down.
SignalChannelNetworkState(receiver_call_, MediaType::VIDEO,
kNetworkDown);
WaitForPacketsOrSilence(true, true);
// Network up for audio for both sides; video is still not expected to
// start
SignalChannelNetworkState(sender_call_, MediaType::AUDIO, kNetworkUp);
SignalChannelNetworkState(receiver_call_, MediaType::AUDIO, kNetworkUp);
WaitForPacketsOrSilence(true, true);
// Network back up again for both.
{
MutexLock lock(&test_mutex_);
// It's OK to encode frames again, as we're about to bring up the
// network.
sender_state_ = kNetworkUp;
}
SignalChannelNetworkState(sender_call_, MediaType::VIDEO, kNetworkUp);
SignalChannelNetworkState(receiver_call_, MediaType::VIDEO, kNetworkUp);
WaitForPacketsOrSilence(false, false);
// TODO(skvlad): add tests to verify that the audio streams are stopped
// when the network goes down for audio once the workaround in
// paced_sender.cc is removed.
});
}
int32_t Encode(const VideoFrame& input_image,
const std::vector<VideoFrameType>* frame_types) override {
{
MutexLock lock(&test_mutex_);
if (sender_state_ == kNetworkDown) {
++down_frames_;
EXPECT_LE(down_frames_, 1)
<< "Encoding more than one frame while network is down.";
if (down_frames_ > 1)
encoded_frames_.Set();
} else {
encoded_frames_.Set();
}
}
return test::FakeEncoder::Encode(input_image, frame_types);
}
private:
void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
int64_t initial_time_ms = clock_->TimeInMilliseconds();
int initial_sender_rtp;
int initial_sender_rtcp;
int initial_receiver_rtcp;
{
MutexLock lock(&test_mutex_);
initial_sender_rtp = sender_rtp_;
initial_sender_rtcp = sender_rtcp_;
initial_receiver_rtcp = receiver_rtcp_;
}
bool sender_done = false;
bool receiver_done = false;
while (!sender_done || !receiver_done) {
packet_event_.Wait(kSilenceTimeoutMs);
int64_t time_now_ms = clock_->TimeInMilliseconds();
MutexLock lock(&test_mutex_);
if (sender_down) {
ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
kNumAcceptedDowntimeRtp)
<< "RTP sent during sender-side downtime.";
ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
kNumAcceptedDowntimeRtcp)
<< "RTCP sent during sender-side downtime.";
if (time_now_ms - initial_time_ms >=
static_cast<int64_t>(kSilenceTimeoutMs)) {
sender_done = true;
}
} else {
if (sender_rtp_ > initial_sender_rtp + kNumAcceptedDowntimeRtp)
sender_done = true;
}
if (receiver_down) {
ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
kNumAcceptedDowntimeRtcp)
<< "RTCP sent during receiver-side downtime.";
if (time_now_ms - initial_time_ms >=
static_cast<int64_t>(kSilenceTimeoutMs)) {
receiver_done = true;
}
} else {
if (receiver_rtcp_ > initial_receiver_rtcp + kNumAcceptedDowntimeRtcp)
receiver_done = true;
}
}
}
TaskQueueBase* const e2e_test_task_queue_;
std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
Mutex test_mutex_;
rtc::Event encoded_frames_;
rtc::Event packet_event_;
Call* sender_call_;
Call* receiver_call_;
test::VideoEncoderProxyFactory encoder_factory_;
NetworkState sender_state_ RTC_GUARDED_BY(test_mutex_);
int sender_rtp_ RTC_GUARDED_BY(test_mutex_);
int sender_padding_ RTC_GUARDED_BY(test_mutex_);
int sender_rtcp_ RTC_GUARDED_BY(test_mutex_);
int receiver_rtcp_ RTC_GUARDED_BY(test_mutex_);
int down_frames_ RTC_GUARDED_BY(test_mutex_);
} test(task_queue());
RunBaseTest(&test);
}
TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
class UnusedEncoder : public test::FakeEncoder {
public:
UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
int32_t InitEncode(const VideoCodec* config,
const Settings& settings) override {
EXPECT_GT(config->startBitrate, 0u);
return 0;
}
int32_t Encode(const VideoFrame& input_image,
const std::vector<VideoFrameType>* frame_types) override {
ADD_FAILURE() << "Unexpected frame encode.";
return test::FakeEncoder::Encode(input_image, frame_types);
}
};
UnusedEncoder unused_encoder;
UnusedTransport unused_transport;
VerifyNewVideoSendStreamsRespectNetworkState(
MediaType::AUDIO, &unused_encoder, &unused_transport);
}
TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
class RequiredEncoder : public test::FakeEncoder {
public:
RequiredEncoder()
: FakeEncoder(Clock::GetRealTimeClock()), encoded_frame_(false) {}
~RequiredEncoder() {
if (!encoded_frame_) {
ADD_FAILURE() << "Didn't encode an expected frame";
}
}
int32_t Encode(const VideoFrame& input_image,
const std::vector<VideoFrameType>* frame_types) override {
encoded_frame_ = true;
return test::FakeEncoder::Encode(input_image, frame_types);
}
private:
bool encoded_frame_;
};
RequiredTransport required_transport(true /*rtp*/, false /*rtcp*/);
RequiredEncoder required_encoder;
VerifyNewVideoSendStreamsRespectNetworkState(
MediaType::VIDEO, &required_encoder, &required_transport);
}
TEST_F(NetworkStateEndToEndTest,
NewVideoReceiveStreamsRespectVideoNetworkDown) {
UnusedTransport transport;
VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
}
TEST_F(NetworkStateEndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
RequiredTransport transport(false /*rtp*/, true /*rtcp*/);
VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
}
} // namespace webrtc