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This CL activates unit tests for the task queue based send side congestion controller that will replace the current one in the future. To be able to have the tests run side by side with the tests of the old congestion controller, the old tests have been prefixed with "Legacy". This CL also contains some minor fixes to the new congestion controller code. Bug: webrtc:8415 Change-Id: I5e7474d42f17fcbfef402e26f638846fa3424695 Reviewed-on: https://webrtc-review.googlesource.com/55381 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22229}
514 lines
20 KiB
C++
514 lines
20 KiB
C++
/*
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* Copyright (c) 2016 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/congestion_controller/delay_based_bwe_unittest_helper.h"
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#include <algorithm>
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#include <limits>
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#include <utility>
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#include "modules/congestion_controller/delay_based_bwe.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/ptr_util.h"
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namespace webrtc {
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constexpr size_t kMtu = 1200;
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constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
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// Number of packets needed before we have a valid estimate.
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constexpr int kNumInitialPackets = 2;
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constexpr int kInitialProbingPackets = 5;
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namespace test {
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void TestBitrateObserver::OnReceiveBitrateChanged(
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const std::vector<uint32_t>& ssrcs,
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uint32_t bitrate) {
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latest_bitrate_ = bitrate;
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updated_ = true;
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}
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RtpStream::RtpStream(int fps, int bitrate_bps)
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: fps_(fps),
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bitrate_bps_(bitrate_bps),
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next_rtp_time_(0),
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sequence_number_(0) {
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RTC_CHECK_GT(fps_, 0);
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}
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// Generates a new frame for this stream. If called too soon after the
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// previous frame, no frame will be generated. The frame is split into
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// packets.
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int64_t RtpStream::GenerateFrame(int64_t time_now_us,
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std::vector<PacketFeedback>* packets) {
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if (time_now_us < next_rtp_time_) {
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return next_rtp_time_;
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}
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RTC_CHECK(packets != NULL);
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size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
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size_t n_packets =
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std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
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size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
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for (size_t i = 0; i < n_packets; ++i) {
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PacketFeedback packet(-1, sequence_number_++);
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packet.send_time_ms = (time_now_us + kSendSideOffsetUs) / 1000;
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packet.payload_size = payload_size;
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packets->push_back(packet);
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}
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next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
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return next_rtp_time_;
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}
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// The send-side time when the next frame can be generated.
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int64_t RtpStream::next_rtp_time() const {
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return next_rtp_time_;
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}
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void RtpStream::set_bitrate_bps(int bitrate_bps) {
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ASSERT_GE(bitrate_bps, 0);
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bitrate_bps_ = bitrate_bps;
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}
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int RtpStream::bitrate_bps() const {
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return bitrate_bps_;
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}
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bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
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const std::unique_ptr<RtpStream>& rhs) {
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return lhs->next_rtp_time_ < rhs->next_rtp_time_;
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}
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StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
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: capacity_(capacity), prev_arrival_time_us_(time_now) {}
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// Add a new stream.
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void StreamGenerator::AddStream(RtpStream* stream) {
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streams_.push_back(std::unique_ptr<RtpStream>(stream));
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}
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// Set the link capacity.
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void StreamGenerator::set_capacity_bps(int capacity_bps) {
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ASSERT_GT(capacity_bps, 0);
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capacity_ = capacity_bps;
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}
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// Divides |bitrate_bps| among all streams. The allocated bitrate per stream
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// is decided by the current allocation ratios.
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void StreamGenerator::SetBitrateBps(int bitrate_bps) {
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ASSERT_GE(streams_.size(), 0u);
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int total_bitrate_before = 0;
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for (const auto& stream : streams_) {
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total_bitrate_before += stream->bitrate_bps();
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}
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int64_t bitrate_before = 0;
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int total_bitrate_after = 0;
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for (const auto& stream : streams_) {
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bitrate_before += stream->bitrate_bps();
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int64_t bitrate_after =
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(bitrate_before * bitrate_bps + total_bitrate_before / 2) /
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total_bitrate_before;
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stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
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total_bitrate_after += stream->bitrate_bps();
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}
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ASSERT_EQ(bitrate_before, total_bitrate_before);
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EXPECT_EQ(total_bitrate_after, bitrate_bps);
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}
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// TODO(holmer): Break out the channel simulation part from this class to make
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// it possible to simulate different types of channels.
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int64_t StreamGenerator::GenerateFrame(std::vector<PacketFeedback>* packets,
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int64_t time_now_us) {
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RTC_CHECK(packets != NULL);
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RTC_CHECK(packets->empty());
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RTC_CHECK_GT(capacity_, 0);
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auto it =
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std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
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(*it)->GenerateFrame(time_now_us, packets);
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int i = 0;
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for (PacketFeedback& packet : *packets) {
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int capacity_bpus = capacity_ / 1000;
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int64_t required_network_time_us =
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(8 * 1000 * packet.payload_size + capacity_bpus / 2) / capacity_bpus;
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prev_arrival_time_us_ =
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std::max(time_now_us + required_network_time_us,
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prev_arrival_time_us_ + required_network_time_us);
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packet.arrival_time_ms = prev_arrival_time_us_ / 1000;
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++i;
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}
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it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
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return std::max((*it)->next_rtp_time(), time_now_us);
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}
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} // namespace test
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LegacyDelayBasedBweTest::LegacyDelayBasedBweTest()
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: clock_(100000000),
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acknowledged_bitrate_estimator_(
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rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
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bitrate_estimator_(new DelayBasedBwe(nullptr, &clock_)),
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stream_generator_(new test::StreamGenerator(1e6, // Capacity.
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clock_.TimeInMicroseconds())),
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arrival_time_offset_ms_(0),
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first_update_(true) {}
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LegacyDelayBasedBweTest::~LegacyDelayBasedBweTest() {}
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void LegacyDelayBasedBweTest::AddDefaultStream() {
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stream_generator_->AddStream(new test::RtpStream(30, 3e5));
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}
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const uint32_t LegacyDelayBasedBweTest::kDefaultSsrc = 0;
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void LegacyDelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
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int64_t send_time_ms,
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uint16_t sequence_number,
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size_t payload_size) {
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IncomingFeedback(arrival_time_ms, send_time_ms, sequence_number, payload_size,
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PacedPacketInfo());
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}
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void LegacyDelayBasedBweTest::IncomingFeedback(
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int64_t arrival_time_ms,
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int64_t send_time_ms,
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uint16_t sequence_number,
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size_t payload_size,
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const PacedPacketInfo& pacing_info) {
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RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
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PacketFeedback packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
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sequence_number, payload_size, pacing_info);
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std::vector<PacketFeedback> packets;
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packets.push_back(packet);
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acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
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DelayBasedBwe::Result result =
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bitrate_estimator_->IncomingPacketFeedbackVector(
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packets, acknowledged_bitrate_estimator_->bitrate_bps());
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const uint32_t kDummySsrc = 0;
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if (result.updated) {
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bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
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result.target_bitrate_bps);
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}
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}
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// Generates a frame of packets belonging to a stream at a given bitrate and
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// with a given ssrc. The stream is pushed through a very simple simulated
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// network, and is then given to the receive-side bandwidth estimator.
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// Returns true if an over-use was seen, false otherwise.
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// The StreamGenerator::updated() should be used to check for any changes in
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// target bitrate after the call to this function.
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bool LegacyDelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
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uint32_t bitrate_bps) {
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stream_generator_->SetBitrateBps(bitrate_bps);
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std::vector<PacketFeedback> packets;
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int64_t next_time_us =
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stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
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if (packets.empty())
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return false;
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bool overuse = false;
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bitrate_observer_.Reset();
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clock_.AdvanceTimeMicroseconds(1000 * packets.back().arrival_time_ms -
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clock_.TimeInMicroseconds());
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for (auto& packet : packets) {
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RTC_CHECK_GE(packet.arrival_time_ms + arrival_time_offset_ms_, 0);
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packet.arrival_time_ms += arrival_time_offset_ms_;
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}
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acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
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DelayBasedBwe::Result result =
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bitrate_estimator_->IncomingPacketFeedbackVector(
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packets, acknowledged_bitrate_estimator_->bitrate_bps());
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const uint32_t kDummySsrc = 0;
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if (result.updated) {
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bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
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result.target_bitrate_bps);
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if (!first_update_ && result.target_bitrate_bps < bitrate_bps)
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overuse = true;
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first_update_ = false;
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}
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clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
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return overuse;
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}
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// Run the bandwidth estimator with a stream of |number_of_frames| frames, or
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// until it reaches |target_bitrate|.
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// Can for instance be used to run the estimator for some time to get it
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// into a steady state.
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uint32_t LegacyDelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
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int max_number_of_frames,
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uint32_t start_bitrate,
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uint32_t min_bitrate,
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uint32_t max_bitrate,
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uint32_t target_bitrate) {
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uint32_t bitrate_bps = start_bitrate;
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bool bitrate_update_seen = false;
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// Produce |number_of_frames| frames and give them to the estimator.
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for (int i = 0; i < max_number_of_frames; ++i) {
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bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
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if (overuse) {
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EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
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EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
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bitrate_bps = bitrate_observer_.latest_bitrate();
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bitrate_update_seen = true;
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} else if (bitrate_observer_.updated()) {
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bitrate_bps = bitrate_observer_.latest_bitrate();
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bitrate_observer_.Reset();
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}
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if (bitrate_update_seen && bitrate_bps > target_bitrate) {
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break;
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}
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}
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EXPECT_TRUE(bitrate_update_seen);
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return bitrate_bps;
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}
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void LegacyDelayBasedBweTest::InitialBehaviorTestHelper(
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uint32_t expected_converge_bitrate) {
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const int kFramerate = 50; // 50 fps to avoid rounding errors.
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const int kFrameIntervalMs = 1000 / kFramerate;
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const PacedPacketInfo kPacingInfo(0, 5, 5000);
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uint32_t bitrate_bps = 0;
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int64_t send_time_ms = 0;
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uint16_t sequence_number = 0;
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std::vector<uint32_t> ssrcs;
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EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
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EXPECT_EQ(0u, ssrcs.size());
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clock_.AdvanceTimeMilliseconds(1000);
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EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
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EXPECT_FALSE(bitrate_observer_.updated());
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bitrate_observer_.Reset();
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clock_.AdvanceTimeMilliseconds(1000);
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// Inserting packets for 5 seconds to get a valid estimate.
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for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
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// NOTE!!! If the following line is moved under the if case then this test
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// wont work on windows realease bots.
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PacedPacketInfo pacing_info =
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i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
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if (i == kNumInitialPackets) {
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EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
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EXPECT_EQ(0u, ssrcs.size());
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EXPECT_FALSE(bitrate_observer_.updated());
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bitrate_observer_.Reset();
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}
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IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
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sequence_number++, kMtu, pacing_info);
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clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
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send_time_ms += kFrameIntervalMs;
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}
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EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
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ASSERT_EQ(1u, ssrcs.size());
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EXPECT_EQ(kDefaultSsrc, ssrcs.front());
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EXPECT_NEAR(expected_converge_bitrate, bitrate_bps, kAcceptedBitrateErrorBps);
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EXPECT_TRUE(bitrate_observer_.updated());
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bitrate_observer_.Reset();
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EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate_bps);
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}
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void LegacyDelayBasedBweTest::RateIncreaseReorderingTestHelper(
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uint32_t expected_bitrate_bps) {
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const int kFramerate = 50; // 50 fps to avoid rounding errors.
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const int kFrameIntervalMs = 1000 / kFramerate;
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const PacedPacketInfo kPacingInfo(0, 5, 5000);
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int64_t send_time_ms = 0;
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uint16_t sequence_number = 0;
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// Inserting packets for five seconds to get a valid estimate.
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for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
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// NOTE!!! If the following line is moved under the if case then this test
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// wont work on windows realease bots.
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PacedPacketInfo pacing_info =
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i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
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// TODO(sprang): Remove this hack once the single stream estimator is gone,
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// as it doesn't do anything in Process().
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if (i == kNumInitialPackets) {
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// Process after we have enough frames to get a valid input rate estimate.
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EXPECT_FALSE(bitrate_observer_.updated()); // No valid estimate.
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}
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IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
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sequence_number++, kMtu, pacing_info);
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clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
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send_time_ms += kFrameIntervalMs;
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}
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EXPECT_TRUE(bitrate_observer_.updated());
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EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
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kAcceptedBitrateErrorBps);
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for (int i = 0; i < 10; ++i) {
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clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
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send_time_ms += 2 * kFrameIntervalMs;
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IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
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sequence_number + 2, 1000);
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IncomingFeedback(clock_.TimeInMilliseconds(),
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send_time_ms - kFrameIntervalMs, sequence_number + 1,
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1000);
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sequence_number += 2;
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}
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EXPECT_TRUE(bitrate_observer_.updated());
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EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
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kAcceptedBitrateErrorBps);
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}
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// Make sure we initially increase the bitrate as expected.
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void LegacyDelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
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int expected_iterations) {
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// This threshold corresponds approximately to increasing linearly with
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// bitrate(i) = 1.04 * bitrate(i-1) + 1000
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// until bitrate(i) > 500000, with bitrate(1) ~= 30000.
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uint32_t bitrate_bps = 30000;
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int iterations = 0;
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AddDefaultStream();
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// Feed the estimator with a stream of packets and verify that it reaches
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// 500 kbps at the expected time.
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while (bitrate_bps < 5e5) {
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bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
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if (overuse) {
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EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
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bitrate_bps = bitrate_observer_.latest_bitrate();
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bitrate_observer_.Reset();
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} else if (bitrate_observer_.updated()) {
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bitrate_bps = bitrate_observer_.latest_bitrate();
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bitrate_observer_.Reset();
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}
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++iterations;
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}
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ASSERT_EQ(expected_iterations, iterations);
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}
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void LegacyDelayBasedBweTest::CapacityDropTestHelper(
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int number_of_streams,
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bool wrap_time_stamp,
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uint32_t expected_bitrate_drop_delta,
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int64_t receiver_clock_offset_change_ms) {
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const int kFramerate = 30;
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const int kStartBitrate = 900e3;
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const int kMinExpectedBitrate = 800e3;
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const int kMaxExpectedBitrate = 1100e3;
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const uint32_t kInitialCapacityBps = 1000e3;
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const uint32_t kReducedCapacityBps = 500e3;
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int steady_state_time = 0;
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if (number_of_streams <= 1) {
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steady_state_time = 10;
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AddDefaultStream();
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} else {
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steady_state_time = 10 * number_of_streams;
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int bitrate_sum = 0;
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int kBitrateDenom = number_of_streams * (number_of_streams - 1);
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for (int i = 0; i < number_of_streams; i++) {
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// First stream gets half available bitrate, while the rest share the
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// remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
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int bitrate = kStartBitrate / 2;
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if (i > 0) {
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bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
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}
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stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
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bitrate_sum += bitrate;
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}
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ASSERT_EQ(bitrate_sum, kStartBitrate);
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}
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// Run in steady state to make the estimator converge.
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stream_generator_->set_capacity_bps(kInitialCapacityBps);
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uint32_t bitrate_bps = SteadyStateRun(
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kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
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kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
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EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
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bitrate_observer_.Reset();
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|
|
|
// Add an offset to make sure the BWE can handle it.
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|
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 LegacyDelayBasedBweTest::TestTimestampGroupingTestHelper() {
|
|
const int kFramerate = 50; // 50 fps to avoid rounding errors.
|
|
const int kFrameIntervalMs = 1000 / kFramerate;
|
|
int64_t send_time_ms = 0;
|
|
uint16_t sequence_number = 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,
|
|
sequence_number++, 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,
|
|
sequence_number++, 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 LegacyDelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
|
|
const int kFramerate = 100;
|
|
const int kFrameIntervalMs = 1000 / kFramerate;
|
|
int64_t send_time_ms = 0;
|
|
uint16_t sequence_number = 0;
|
|
|
|
for (size_t i = 0; i < 3000; ++i) {
|
|
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
|
|
sequence_number++, 1000);
|
|
clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
|
|
send_time_ms += kFrameIntervalMs;
|
|
}
|
|
uint32_t bitrate_before = 0;
|
|
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,
|
|
sequence_number++, 1000);
|
|
clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
|
|
send_time_ms += kFrameIntervalMs;
|
|
}
|
|
uint32_t bitrate_after = 0;
|
|
bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
|
|
EXPECT_LT(bitrate_after, bitrate_before);
|
|
}
|
|
} // namespace webrtc
|