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/*
* 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 <cstddef>
#include <cstdint>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_interface.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
#include "rtc_base/checks.h"
#include "test/field_trial.h"
#include "test/gtest.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,
int64_t* next_sequence_number,
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::Micros(time_now_us + kSendSideOffsetUs);
packet.sent_packet.size = DataSize::Bytes(payload_size);
packet.sent_packet.sequence_number = (*next_sequence_number)++;
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(int64_t time_now_us,
int64_t* next_sequence_number,
std::vector<PacketResult>* packets) {
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, next_sequence_number, packets);
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::Micros(prev_arrival_time_us_);
}
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(std::make_unique<test::ScopedFieldTrials>(
"WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled:true/")),
clock_(100000000),
acknowledged_bitrate_estimator_(
AcknowledgedBitrateEstimatorInterface::Create(&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),
next_sequence_number_(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);
IncomingFeedback(Timestamp::Millis(arrival_time_ms + arrival_time_offset_ms_),
Timestamp::Millis(send_time_ms), payload_size, pacing_info);
}
void DelayBasedBweTest::IncomingFeedback(Timestamp receive_time,
Timestamp send_time,
size_t payload_size,
const PacedPacketInfo& pacing_info) {
PacketResult packet;
packet.receive_time = receive_time;
packet.sent_packet.send_time = send_time;
packet.sent_packet.size = DataSize::Bytes(payload_size);
packet.sent_packet.pacing_info = pacing_info;
packet.sent_packet.sequence_number = next_sequence_number_++;
if (packet.sent_packet.pacing_info.probe_cluster_id !=
PacedPacketInfo::kNotAProbe)
probe_bitrate_estimator_->HandleProbeAndEstimateBitrate(packet);
TransportPacketsFeedback msg;
msg.feedback_time = Timestamp::Millis(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(
clock_.TimeInMicroseconds(), &next_sequence_number_, &packets);
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::Millis(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::Millis(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
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