c5017136c7
end_to_end_tests.cc was over 5000 lines and covered many different areas in it's testing. In this change it is separated into multiple smaller test sets separated by the functionality they are testing. The reasoning behind this is that the fact that a test is working end to end should be secondary to what functionality the test is actually testing. A slight functional change is that for some of the tests the parametrization over round robin pacing being controlled with a field trial is removed since they are simple enough that they should not be affected by the pacing method. Bug: None Change-Id: I4b7eba80fc142ecfc8fa642dab9b6f587d914048 Reviewed-on: https://webrtc-review.googlesource.com/46143 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21876}
359 lines
12 KiB
C++
359 lines
12 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 "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "rtc_base/rate_limiter.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/call_test.h"
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#include "test/field_trial.h"
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#include "test/gtest.h"
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#include "test/rtcp_packet_parser.h"
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#include "test/rtp_rtcp_observer.h"
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namespace webrtc {
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class BandwidthEndToEndTest : public test::CallTest,
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public testing::WithParamInterface<std::string> {
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public:
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BandwidthEndToEndTest() : field_trial_(GetParam()) {}
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virtual ~BandwidthEndToEndTest() {
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EXPECT_EQ(nullptr, video_send_stream_);
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EXPECT_TRUE(video_receive_streams_.empty());
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}
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private:
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test::ScopedFieldTrials field_trial_;
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};
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INSTANTIATE_TEST_CASE_P(RoundRobin,
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BandwidthEndToEndTest,
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::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
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"WebRTC-RoundRobinPacing/Enabled/"));
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TEST_P(BandwidthEndToEndTest, ReceiveStreamSendsRemb) {
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class RembObserver : public test::EndToEndTest {
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public:
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RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
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void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) override {
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send_config->rtp.extensions.clear();
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send_config->rtp.extensions.push_back(RtpExtension(
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RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
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(*receive_configs)[0].rtp.remb = true;
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(*receive_configs)[0].rtp.transport_cc = false;
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}
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Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
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test::RtcpPacketParser parser;
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EXPECT_TRUE(parser.Parse(packet, length));
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if (parser.remb()->num_packets() > 0) {
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EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
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EXPECT_LT(0U, parser.remb()->bitrate_bps());
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EXPECT_EQ(1U, parser.remb()->ssrcs().size());
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EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
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observation_complete_.Set();
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}
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return SEND_PACKET;
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}
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void PerformTest() override {
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EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
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"receiver RTCP REMB packet to be "
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"sent.";
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}
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} test;
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RunBaseTest(&test);
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}
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class BandwidthStatsTest : public test::EndToEndTest {
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public:
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explicit BandwidthStatsTest(bool send_side_bwe)
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: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
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sender_call_(nullptr),
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receiver_call_(nullptr),
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has_seen_pacer_delay_(false),
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send_side_bwe_(send_side_bwe) {}
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void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) override {
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if (!send_side_bwe_) {
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send_config->rtp.extensions.clear();
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send_config->rtp.extensions.push_back(RtpExtension(
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RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
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(*receive_configs)[0].rtp.remb = true;
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(*receive_configs)[0].rtp.transport_cc = false;
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}
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}
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Action OnSendRtp(const uint8_t* packet, size_t length) override {
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Call::Stats sender_stats = sender_call_->GetStats();
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Call::Stats receiver_stats = receiver_call_->GetStats();
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if (!has_seen_pacer_delay_)
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has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
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if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) {
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if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0)
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observation_complete_.Set();
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}
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return SEND_PACKET;
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}
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void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
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sender_call_ = sender_call;
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receiver_call_ = receiver_call;
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}
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void PerformTest() override {
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EXPECT_TRUE(Wait()) << "Timed out while waiting for "
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"non-zero bandwidth stats.";
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}
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private:
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Call* sender_call_;
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Call* receiver_call_;
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bool has_seen_pacer_delay_;
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const bool send_side_bwe_;
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};
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TEST_P(BandwidthEndToEndTest, VerifySendSideBweStats) {
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BandwidthStatsTest test(true);
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RunBaseTest(&test);
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}
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TEST_P(BandwidthEndToEndTest, VerifyRecvSideBweStats) {
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BandwidthStatsTest test(false);
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RunBaseTest(&test);
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}
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// Verifies that it's possible to limit the send BWE by sending a REMB.
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// This is verified by allowing the send BWE to ramp-up to >1000 kbps,
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// then have the test generate a REMB of 500 kbps and verify that the send BWE
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// is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
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// test verifies that the send BWE ramps back up to exactly 1000 kbps.
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TEST_P(BandwidthEndToEndTest, RembWithSendSideBwe) {
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class BweObserver : public test::EndToEndTest {
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public:
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BweObserver()
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: EndToEndTest(kDefaultTimeoutMs),
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sender_call_(nullptr),
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clock_(Clock::GetRealTimeClock()),
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sender_ssrc_(0),
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remb_bitrate_bps_(1000000),
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receive_transport_(nullptr),
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stop_event_(false, false),
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poller_thread_(&BitrateStatsPollingThread,
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this,
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"BitrateStatsPollingThread"),
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state_(kWaitForFirstRampUp),
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retransmission_rate_limiter_(clock_, 1000) {}
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~BweObserver() {}
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test::PacketTransport* CreateReceiveTransport(
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test::SingleThreadedTaskQueueForTesting* task_queue) override {
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receive_transport_ = new test::PacketTransport(
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task_queue, nullptr, this, test::PacketTransport::kReceiver,
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payload_type_map_, FakeNetworkPipe::Config());
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return receive_transport_;
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}
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Call::Config GetSenderCallConfig() override {
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Call::Config config(event_log_.get());
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// Set a high start bitrate to reduce the test completion time.
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config.bitrate_config.start_bitrate_bps = remb_bitrate_bps_;
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return config;
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}
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void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) override {
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ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
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sender_ssrc_ = send_config->rtp.ssrcs[0];
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encoder_config->max_bitrate_bps = 2000000;
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ASSERT_EQ(1u, receive_configs->size());
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RtpRtcp::Configuration config;
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config.receiver_only = true;
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config.clock = clock_;
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config.outgoing_transport = receive_transport_;
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config.retransmission_rate_limiter = &retransmission_rate_limiter_;
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rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
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rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
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rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
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rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
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}
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void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
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sender_call_ = sender_call;
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}
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static void BitrateStatsPollingThread(void* obj) {
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static_cast<BweObserver*>(obj)->PollStats();
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}
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void PollStats() {
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do {
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if (sender_call_) {
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Call::Stats stats = sender_call_->GetStats();
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switch (state_) {
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case kWaitForFirstRampUp:
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if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
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state_ = kWaitForRemb;
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remb_bitrate_bps_ /= 2;
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rtp_rtcp_->SetRemb(
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remb_bitrate_bps_,
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std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
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rtp_rtcp_->SendRTCP(kRtcpRr);
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}
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break;
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case kWaitForRemb:
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if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
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state_ = kWaitForSecondRampUp;
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remb_bitrate_bps_ *= 2;
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rtp_rtcp_->SetRemb(
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remb_bitrate_bps_,
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std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
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rtp_rtcp_->SendRTCP(kRtcpRr);
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}
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break;
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case kWaitForSecondRampUp:
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if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
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observation_complete_.Set();
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}
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break;
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}
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}
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} while (!stop_event_.Wait(1000));
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}
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void PerformTest() override {
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poller_thread_.Start();
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EXPECT_TRUE(Wait())
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<< "Timed out while waiting for bitrate to change according to REMB.";
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stop_event_.Set();
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poller_thread_.Stop();
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}
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private:
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enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };
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Call* sender_call_;
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Clock* const clock_;
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uint32_t sender_ssrc_;
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int remb_bitrate_bps_;
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std::unique_ptr<RtpRtcp> rtp_rtcp_;
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test::PacketTransport* receive_transport_;
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rtc::Event stop_event_;
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rtc::PlatformThread poller_thread_;
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TestState state_;
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RateLimiter retransmission_rate_limiter_;
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} test;
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RunBaseTest(&test);
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}
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TEST_P(BandwidthEndToEndTest, ReportsSetEncoderRates) {
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class EncoderRateStatsTest : public test::EndToEndTest,
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public test::FakeEncoder {
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public:
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explicit EncoderRateStatsTest(
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test::SingleThreadedTaskQueueForTesting* task_queue)
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: EndToEndTest(kDefaultTimeoutMs),
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FakeEncoder(Clock::GetRealTimeClock()),
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task_queue_(task_queue),
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send_stream_(nullptr),
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bitrate_kbps_(0) {}
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void OnVideoStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStream*>& receive_streams) override {
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send_stream_ = send_stream;
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}
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void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) override {
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send_config->encoder_settings.encoder = this;
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RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
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}
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int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
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uint32_t framerate) override {
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// Make sure not to trigger on any default zero bitrates.
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if (rate_allocation.get_sum_bps() == 0)
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return 0;
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rtc::CritScope lock(&crit_);
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bitrate_kbps_ = rate_allocation.get_sum_kbps();
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observation_complete_.Set();
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return 0;
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}
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void PerformTest() override {
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ASSERT_TRUE(Wait())
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<< "Timed out while waiting for encoder SetRates() call.";
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task_queue_->SendTask([this]() {
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WaitForEncoderTargetBitrateMatchStats();
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send_stream_->Stop();
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WaitForStatsReportZeroTargetBitrate();
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send_stream_->Start();
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WaitForEncoderTargetBitrateMatchStats();
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});
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}
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void WaitForEncoderTargetBitrateMatchStats() {
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for (int i = 0; i < kDefaultTimeoutMs; ++i) {
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VideoSendStream::Stats stats = send_stream_->GetStats();
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{
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rtc::CritScope lock(&crit_);
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if ((stats.target_media_bitrate_bps + 500) / 1000 ==
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static_cast<int>(bitrate_kbps_)) {
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return;
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}
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}
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SleepMs(1);
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}
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FAIL()
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<< "Timed out waiting for stats reporting the currently set bitrate.";
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}
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void WaitForStatsReportZeroTargetBitrate() {
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for (int i = 0; i < kDefaultTimeoutMs; ++i) {
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if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
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return;
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}
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SleepMs(1);
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}
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FAIL() << "Timed out waiting for stats reporting zero bitrate.";
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}
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private:
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test::SingleThreadedTaskQueueForTesting* const task_queue_;
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rtc::CriticalSection crit_;
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VideoSendStream* send_stream_;
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uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
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} test(&task_queue_);
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RunBaseTest(&test);
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}
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} // namespace webrtc
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