| /* |
| * Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h" |
| |
| #include <algorithm> |
| #include <cstdint> |
| |
| #include "absl/memory/memory.h" |
| #include "modules/congestion_controller/goog_cc/delay_based_bwe.h" |
| #include "rtc_base/checks.h" |
| |
| namespace webrtc { |
| constexpr size_t kMtu = 1200; |
| constexpr uint32_t kAcceptedBitrateErrorBps = 50000; |
| |
| // Number of packets needed before we have a valid estimate. |
| constexpr int kNumInitialPackets = 2; |
| |
| constexpr int kInitialProbingPackets = 5; |
| |
| namespace test { |
| |
| void TestBitrateObserver::OnReceiveBitrateChanged( |
| uint32_t bitrate) { |
| latest_bitrate_ = bitrate; |
| updated_ = true; |
| } |
| |
| RtpStream::RtpStream(int fps, int bitrate_bps) |
| : fps_(fps), bitrate_bps_(bitrate_bps), next_rtp_time_(0) { |
| RTC_CHECK_GT(fps_, 0); |
| } |
| |
| // Generates a new frame for this stream. If called too soon after the |
| // previous frame, no frame will be generated. The frame is split into |
| // packets. |
| int64_t RtpStream::GenerateFrame(int64_t time_now_us, |
| std::vector<PacketResult>* packets) { |
| if (time_now_us < next_rtp_time_) { |
| return next_rtp_time_; |
| } |
| RTC_CHECK(packets != NULL); |
| size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_; |
| size_t n_packets = |
| std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u); |
| size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets); |
| for (size_t i = 0; i < n_packets; ++i) { |
| PacketResult packet; |
| packet.sent_packet.send_time = |
| Timestamp::us(time_now_us + kSendSideOffsetUs); |
| packet.sent_packet.size = DataSize::bytes(payload_size); |
| packets->push_back(packet); |
| } |
| next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_; |
| return next_rtp_time_; |
| } |
| |
| // The send-side time when the next frame can be generated. |
| int64_t RtpStream::next_rtp_time() const { |
| return next_rtp_time_; |
| } |
| |
| void RtpStream::set_bitrate_bps(int bitrate_bps) { |
| ASSERT_GE(bitrate_bps, 0); |
| bitrate_bps_ = bitrate_bps; |
| } |
| |
| int RtpStream::bitrate_bps() const { |
| return bitrate_bps_; |
| } |
| |
| bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs, |
| const std::unique_ptr<RtpStream>& rhs) { |
| return lhs->next_rtp_time_ < rhs->next_rtp_time_; |
| } |
| |
| StreamGenerator::StreamGenerator(int capacity, int64_t time_now) |
| : capacity_(capacity), prev_arrival_time_us_(time_now) {} |
| |
| StreamGenerator::~StreamGenerator() = default; |
| |
| // Add a new stream. |
| void StreamGenerator::AddStream(RtpStream* stream) { |
| streams_.push_back(std::unique_ptr<RtpStream>(stream)); |
| } |
| |
| // Set the link capacity. |
| void StreamGenerator::set_capacity_bps(int capacity_bps) { |
| ASSERT_GT(capacity_bps, 0); |
| capacity_ = capacity_bps; |
| } |
| |
| // Divides |bitrate_bps| among all streams. The allocated bitrate per stream |
| // is decided by the current allocation ratios. |
| void StreamGenerator::SetBitrateBps(int bitrate_bps) { |
| ASSERT_GE(streams_.size(), 0u); |
| int total_bitrate_before = 0; |
| for (const auto& stream : streams_) { |
| total_bitrate_before += stream->bitrate_bps(); |
| } |
| int64_t bitrate_before = 0; |
| int total_bitrate_after = 0; |
| for (const auto& stream : streams_) { |
| bitrate_before += stream->bitrate_bps(); |
| int64_t bitrate_after = |
| (bitrate_before * bitrate_bps + total_bitrate_before / 2) / |
| total_bitrate_before; |
| stream->set_bitrate_bps(bitrate_after - total_bitrate_after); |
| total_bitrate_after += stream->bitrate_bps(); |
| } |
| ASSERT_EQ(bitrate_before, total_bitrate_before); |
| EXPECT_EQ(total_bitrate_after, bitrate_bps); |
| } |
| |
| // TODO(holmer): Break out the channel simulation part from this class to make |
| // it possible to simulate different types of channels. |
| int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets, |
| int64_t time_now_us) { |
| RTC_CHECK(packets != NULL); |
| RTC_CHECK(packets->empty()); |
| RTC_CHECK_GT(capacity_, 0); |
| auto it = |
| std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare); |
| (*it)->GenerateFrame(time_now_us, packets); |
| int i = 0; |
| for (PacketResult& packet : *packets) { |
| int capacity_bpus = capacity_ / 1000; |
| int64_t required_network_time_us = |
| (8 * 1000 * packet.sent_packet.size.bytes() + capacity_bpus / 2) / |
| capacity_bpus; |
| prev_arrival_time_us_ = |
| std::max(time_now_us + required_network_time_us, |
| prev_arrival_time_us_ + required_network_time_us); |
| packet.receive_time = Timestamp::us(prev_arrival_time_us_); |
| ++i; |
| } |
| it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare); |
| return std::max((*it)->next_rtp_time(), time_now_us); |
| } |
| } // namespace test |
| |
| DelayBasedBweTest::DelayBasedBweTest() |
| : field_trial(), |
| clock_(100000000), |
| acknowledged_bitrate_estimator_( |
| absl::make_unique<AcknowledgedBitrateEstimator>( |
| &field_trial_config_)), |
| probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)), |
| bitrate_estimator_( |
| new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)), |
| stream_generator_(new test::StreamGenerator(1e6, // Capacity. |
| clock_.TimeInMicroseconds())), |
| arrival_time_offset_ms_(0), |
| first_update_(true) {} |
| |
| DelayBasedBweTest::DelayBasedBweTest(const std::string& field_trial_string) |
| : field_trial( |
| absl::make_unique<test::ScopedFieldTrials>(field_trial_string)), |
| clock_(100000000), |
| acknowledged_bitrate_estimator_( |
| absl::make_unique<AcknowledgedBitrateEstimator>( |
| &field_trial_config_)), |
| probe_bitrate_estimator_(new ProbeBitrateEstimator(nullptr)), |
| bitrate_estimator_( |
| new DelayBasedBwe(&field_trial_config_, nullptr, nullptr)), |
| stream_generator_(new test::StreamGenerator(1e6, // Capacity. |
| clock_.TimeInMicroseconds())), |
| arrival_time_offset_ms_(0), |
| first_update_(true) {} |
| |
| DelayBasedBweTest::~DelayBasedBweTest() {} |
| |
| void DelayBasedBweTest::AddDefaultStream() { |
| stream_generator_->AddStream(new test::RtpStream(30, 3e5)); |
| } |
| |
| const uint32_t DelayBasedBweTest::kDefaultSsrc = 0; |
| |
| void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms, |
| int64_t send_time_ms, |
| size_t payload_size) { |
| IncomingFeedback(arrival_time_ms, send_time_ms, payload_size, |
| PacedPacketInfo()); |
| } |
| |
| void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms, |
| int64_t send_time_ms, |
| size_t payload_size, |
| const PacedPacketInfo& pacing_info) { |
| RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0); |
| PacketResult packet; |
| packet.receive_time = |
| Timestamp::ms(arrival_time_ms + arrival_time_offset_ms_); |
| packet.sent_packet.send_time = Timestamp::ms(send_time_ms); |
| packet.sent_packet.size = DataSize::bytes(payload_size); |
| packet.sent_packet.pacing_info = pacing_info; |
| if (packet.sent_packet.pacing_info.probe_cluster_id != |
| PacedPacketInfo::kNotAProbe) |
| probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet); |
| |
| TransportPacketsFeedback msg; |
| msg.feedback_time = Timestamp::ms(clock_.TimeInMilliseconds()); |
| msg.packet_feedbacks.push_back(packet); |
| acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector( |
| msg.SortedByReceiveTime()); |
| DelayBasedBwe::Result result = |
| bitrate_estimator_->IncomingPacketFeedbackVector( |
| msg, acknowledged_bitrate_estimator_->bitrate(), |
| probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(), |
| /*network_estimate*/ absl::nullopt, /*in_alr*/ false); |
| if (result.updated) { |
| bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps()); |
| } |
| } |
| |
| // Generates a frame of packets belonging to a stream at a given bitrate and |
| // with a given ssrc. The stream is pushed through a very simple simulated |
| // network, and is then given to the receive-side bandwidth estimator. |
| // Returns true if an over-use was seen, false otherwise. |
| // The StreamGenerator::updated() should be used to check for any changes in |
| // target bitrate after the call to this function. |
| bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc, |
| uint32_t bitrate_bps) { |
| stream_generator_->SetBitrateBps(bitrate_bps); |
| std::vector<PacketResult> packets; |
| |
| int64_t next_time_us = |
| stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds()); |
| if (packets.empty()) |
| return false; |
| |
| bool overuse = false; |
| bitrate_observer_.Reset(); |
| clock_.AdvanceTimeMicroseconds(packets.back().receive_time.us() - |
| clock_.TimeInMicroseconds()); |
| for (auto& packet : packets) { |
| RTC_CHECK_GE(packet.receive_time.ms() + arrival_time_offset_ms_, 0); |
| packet.receive_time += TimeDelta::ms(arrival_time_offset_ms_); |
| |
| if (packet.sent_packet.pacing_info.probe_cluster_id != |
| PacedPacketInfo::kNotAProbe) |
| probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet); |
| } |
| |
| acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets); |
| TransportPacketsFeedback msg; |
| msg.packet_feedbacks = packets; |
| msg.feedback_time = Timestamp::ms(clock_.TimeInMilliseconds()); |
| |
| DelayBasedBwe::Result result = |
| bitrate_estimator_->IncomingPacketFeedbackVector( |
| msg, acknowledged_bitrate_estimator_->bitrate(), |
| probe_bitrate_estimator_->FetchAndResetLastEstimatedBitrate(), |
| /*network_estimate*/ absl::nullopt, /*in_alr*/ false); |
| if (result.updated) { |
| bitrate_observer_.OnReceiveBitrateChanged(result.target_bitrate.bps()); |
| if (!first_update_ && result.target_bitrate.bps() < bitrate_bps) |
| overuse = true; |
| first_update_ = false; |
| } |
| |
| clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds()); |
| return overuse; |
| } |
| |
| // Run the bandwidth estimator with a stream of |number_of_frames| frames, or |
| // until it reaches |target_bitrate|. |
| // Can for instance be used to run the estimator for some time to get it |
| // into a steady state. |
| uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc, |
| int max_number_of_frames, |
| uint32_t start_bitrate, |
| uint32_t min_bitrate, |
| uint32_t max_bitrate, |
| uint32_t target_bitrate) { |
| uint32_t bitrate_bps = start_bitrate; |
| bool bitrate_update_seen = false; |
| // Produce |number_of_frames| frames and give them to the estimator. |
| for (int i = 0; i < max_number_of_frames; ++i) { |
| bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps); |
| if (overuse) { |
| EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate); |
| EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate); |
| bitrate_bps = bitrate_observer_.latest_bitrate(); |
| bitrate_update_seen = true; |
| } else if (bitrate_observer_.updated()) { |
| bitrate_bps = bitrate_observer_.latest_bitrate(); |
| bitrate_observer_.Reset(); |
| } |
| if (bitrate_update_seen && bitrate_bps > target_bitrate) { |
| break; |
| } |
| } |
| EXPECT_TRUE(bitrate_update_seen); |
| return bitrate_bps; |
| } |
| |
| void DelayBasedBweTest::InitialBehaviorTestHelper( |
| uint32_t expected_converge_bitrate) { |
| const int kFramerate = 50; // 50 fps to avoid rounding errors. |
| const int kFrameIntervalMs = 1000 / kFramerate; |
| const PacedPacketInfo kPacingInfo(0, 5, 5000); |
| DataRate bitrate = DataRate::Zero(); |
| int64_t send_time_ms = 0; |
| std::vector<uint32_t> ssrcs; |
| EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate)); |
| EXPECT_EQ(0u, ssrcs.size()); |
| clock_.AdvanceTimeMilliseconds(1000); |
| EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate)); |
| EXPECT_FALSE(bitrate_observer_.updated()); |
| bitrate_observer_.Reset(); |
| clock_.AdvanceTimeMilliseconds(1000); |
| // Inserting packets for 5 seconds to get a valid estimate. |
| for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) { |
| // NOTE!!! If the following line is moved under the if case then this test |
| // wont work on windows realease bots. |
| PacedPacketInfo pacing_info = |
| i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo(); |
| |
| if (i == kNumInitialPackets) { |
| EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate)); |
| EXPECT_EQ(0u, ssrcs.size()); |
| EXPECT_FALSE(bitrate_observer_.updated()); |
| bitrate_observer_.Reset(); |
| } |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu, |
| pacing_info); |
| clock_.AdvanceTimeMilliseconds(1000 / kFramerate); |
| send_time_ms += kFrameIntervalMs; |
| } |
| EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate)); |
| ASSERT_EQ(1u, ssrcs.size()); |
| EXPECT_EQ(kDefaultSsrc, ssrcs.front()); |
| EXPECT_NEAR(expected_converge_bitrate, bitrate.bps(), |
| kAcceptedBitrateErrorBps); |
| EXPECT_TRUE(bitrate_observer_.updated()); |
| bitrate_observer_.Reset(); |
| EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate.bps()); |
| } |
| |
| void DelayBasedBweTest::RateIncreaseReorderingTestHelper( |
| uint32_t expected_bitrate_bps) { |
| const int kFramerate = 50; // 50 fps to avoid rounding errors. |
| const int kFrameIntervalMs = 1000 / kFramerate; |
| const PacedPacketInfo kPacingInfo(0, 5, 5000); |
| int64_t send_time_ms = 0; |
| // Inserting packets for five seconds to get a valid estimate. |
| for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) { |
| // NOTE!!! If the following line is moved under the if case then this test |
| // wont work on windows realease bots. |
| PacedPacketInfo pacing_info = |
| i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo(); |
| |
| // TODO(sprang): Remove this hack once the single stream estimator is gone, |
| // as it doesn't do anything in Process(). |
| if (i == kNumInitialPackets) { |
| // Process after we have enough frames to get a valid input rate estimate. |
| |
| EXPECT_FALSE(bitrate_observer_.updated()); // No valid estimate. |
| } |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, kMtu, |
| pacing_info); |
| clock_.AdvanceTimeMilliseconds(kFrameIntervalMs); |
| send_time_ms += kFrameIntervalMs; |
| } |
| EXPECT_TRUE(bitrate_observer_.updated()); |
| EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(), |
| kAcceptedBitrateErrorBps); |
| for (int i = 0; i < 10; ++i) { |
| clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs); |
| send_time_ms += 2 * kFrameIntervalMs; |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000); |
| IncomingFeedback(clock_.TimeInMilliseconds(), |
| send_time_ms - kFrameIntervalMs, 1000); |
| } |
| EXPECT_TRUE(bitrate_observer_.updated()); |
| EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(), |
| kAcceptedBitrateErrorBps); |
| } |
| |
| // Make sure we initially increase the bitrate as expected. |
| void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper( |
| int expected_iterations) { |
| // This threshold corresponds approximately to increasing linearly with |
| // bitrate(i) = 1.04 * bitrate(i-1) + 1000 |
| // until bitrate(i) > 500000, with bitrate(1) ~= 30000. |
| uint32_t bitrate_bps = 30000; |
| int iterations = 0; |
| AddDefaultStream(); |
| // Feed the estimator with a stream of packets and verify that it reaches |
| // 500 kbps at the expected time. |
| while (bitrate_bps < 5e5) { |
| bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps); |
| if (overuse) { |
| EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps); |
| bitrate_bps = bitrate_observer_.latest_bitrate(); |
| bitrate_observer_.Reset(); |
| } else if (bitrate_observer_.updated()) { |
| bitrate_bps = bitrate_observer_.latest_bitrate(); |
| bitrate_observer_.Reset(); |
| } |
| ++iterations; |
| } |
| ASSERT_EQ(expected_iterations, iterations); |
| } |
| |
| void DelayBasedBweTest::CapacityDropTestHelper( |
| int number_of_streams, |
| bool wrap_time_stamp, |
| uint32_t expected_bitrate_drop_delta, |
| int64_t receiver_clock_offset_change_ms) { |
| const int kFramerate = 30; |
| const int kStartBitrate = 900e3; |
| const int kMinExpectedBitrate = 800e3; |
| const int kMaxExpectedBitrate = 1100e3; |
| const uint32_t kInitialCapacityBps = 1000e3; |
| const uint32_t kReducedCapacityBps = 500e3; |
| |
| int steady_state_time = 0; |
| if (number_of_streams <= 1) { |
| steady_state_time = 10; |
| AddDefaultStream(); |
| } else { |
| steady_state_time = 10 * number_of_streams; |
| int bitrate_sum = 0; |
| int kBitrateDenom = number_of_streams * (number_of_streams - 1); |
| for (int i = 0; i < number_of_streams; i++) { |
| // First stream gets half available bitrate, while the rest share the |
| // remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up) |
| int bitrate = kStartBitrate / 2; |
| if (i > 0) { |
| bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom; |
| } |
| stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate)); |
| bitrate_sum += bitrate; |
| } |
| ASSERT_EQ(bitrate_sum, kStartBitrate); |
| } |
| |
| // Run in steady state to make the estimator converge. |
| stream_generator_->set_capacity_bps(kInitialCapacityBps); |
| uint32_t bitrate_bps = SteadyStateRun( |
| kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate, |
| kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps); |
| EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u); |
| bitrate_observer_.Reset(); |
| |
| // Add an offset to make sure the BWE can handle it. |
| arrival_time_offset_ms_ += receiver_clock_offset_change_ms; |
| |
| // Reduce the capacity and verify the decrease time. |
| stream_generator_->set_capacity_bps(kReducedCapacityBps); |
| int64_t overuse_start_time = clock_.TimeInMilliseconds(); |
| int64_t bitrate_drop_time = -1; |
| for (int i = 0; i < 100 * number_of_streams; ++i) { |
| GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps); |
| if (bitrate_drop_time == -1 && |
| bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) { |
| bitrate_drop_time = clock_.TimeInMilliseconds(); |
| } |
| if (bitrate_observer_.updated()) |
| bitrate_bps = bitrate_observer_.latest_bitrate(); |
| } |
| |
| EXPECT_NEAR(expected_bitrate_drop_delta, |
| bitrate_drop_time - overuse_start_time, 33); |
| } |
| |
| void DelayBasedBweTest::TestTimestampGroupingTestHelper() { |
| const int kFramerate = 50; // 50 fps to avoid rounding errors. |
| const int kFrameIntervalMs = 1000 / kFramerate; |
| int64_t send_time_ms = 0; |
| // Initial set of frames to increase the bitrate. 6 seconds to have enough |
| // time for the first estimate to be generated and for Process() to be called. |
| for (int i = 0; i <= 6 * kFramerate; ++i) { |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000); |
| |
| clock_.AdvanceTimeMilliseconds(kFrameIntervalMs); |
| send_time_ms += kFrameIntervalMs; |
| } |
| EXPECT_TRUE(bitrate_observer_.updated()); |
| EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u); |
| |
| // Insert batches of frames which were sent very close in time. Also simulate |
| // capacity over-use to see that we back off correctly. |
| const int kTimestampGroupLength = 15; |
| for (int i = 0; i < 100; ++i) { |
| for (int j = 0; j < kTimestampGroupLength; ++j) { |
| // Insert |kTimestampGroupLength| frames with just 1 timestamp ticks in |
| // between. Should be treated as part of the same group by the estimator. |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 100); |
| clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength); |
| send_time_ms += 1; |
| } |
| // Increase time until next batch to simulate over-use. |
| clock_.AdvanceTimeMilliseconds(10); |
| send_time_ms += kFrameIntervalMs - kTimestampGroupLength; |
| } |
| EXPECT_TRUE(bitrate_observer_.updated()); |
| // Should have reduced the estimate. |
| EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u); |
| } |
| |
| void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) { |
| const int kFramerate = 100; |
| const int kFrameIntervalMs = 1000 / kFramerate; |
| int64_t send_time_ms = 0; |
| |
| for (size_t i = 0; i < 3000; ++i) { |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000); |
| clock_.AdvanceTimeMilliseconds(kFrameIntervalMs); |
| send_time_ms += kFrameIntervalMs; |
| } |
| DataRate bitrate_before = DataRate::Zero(); |
| std::vector<uint32_t> ssrcs; |
| bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before); |
| |
| clock_.AdvanceTimeMilliseconds(silence_time_s * 1000); |
| send_time_ms += silence_time_s * 1000; |
| |
| for (size_t i = 0; i < 24; ++i) { |
| IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms, 1000); |
| clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs); |
| send_time_ms += kFrameIntervalMs; |
| } |
| DataRate bitrate_after = DataRate::Zero(); |
| bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after); |
| EXPECT_LT(bitrate_after, bitrate_before); |
| } |
| } // namespace webrtc |