| /* |
| * Copyright (c) 2012 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 <assert.h> |
| #include <math.h> |
| #include <string.h> |
| |
| #include <algorithm> |
| #include <cmath> |
| #include <utility> |
| |
| #include "call/call.h" |
| #include "call/fake_network_pipe.h" |
| #include "modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "rtc_base/logging.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| constexpr int64_t kDefaultProcessIntervalMs = 5; |
| struct PacketArrivalTimeComparator { |
| bool operator()(const NetworkPacket& p1, const NetworkPacket& p2) { |
| return p1.arrival_time() < p2.arrival_time(); |
| } |
| }; |
| } // namespace |
| |
| NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet, |
| int64_t send_time, |
| int64_t arrival_time, |
| rtc::Optional<PacketOptions> packet_options, |
| bool is_rtcp, |
| MediaType media_type, |
| rtc::Optional<PacketTime> packet_time) |
| : packet_(std::move(packet)), |
| send_time_(send_time), |
| arrival_time_(arrival_time), |
| packet_options_(packet_options), |
| is_rtcp_(is_rtcp), |
| media_type_(media_type), |
| packet_time_(packet_time) {} |
| |
| NetworkPacket::NetworkPacket(NetworkPacket&& o) |
| : packet_(std::move(o.packet_)), |
| send_time_(o.send_time_), |
| arrival_time_(o.arrival_time_), |
| packet_options_(o.packet_options_), |
| is_rtcp_(o.is_rtcp_), |
| media_type_(o.media_type_), |
| packet_time_(o.packet_time_) {} |
| |
| NetworkPacket& NetworkPacket::operator=(NetworkPacket&& o) { |
| packet_ = std::move(o.packet_); |
| send_time_ = o.send_time_; |
| arrival_time_ = o.arrival_time_; |
| packet_options_ = o.packet_options_; |
| is_rtcp_ = o.is_rtcp_; |
| media_type_ = o.media_type_; |
| packet_time_ = o.packet_time_; |
| |
| return *this; |
| } |
| |
| DemuxerImpl::DemuxerImpl(const std::map<uint8_t, MediaType>& payload_type_map) |
| : packet_receiver_(nullptr), payload_type_map_(payload_type_map) {} |
| |
| void DemuxerImpl::SetReceiver(PacketReceiver* receiver) { |
| packet_receiver_ = receiver; |
| } |
| |
| void DemuxerImpl::DeliverPacket(const NetworkPacket* packet, |
| const PacketTime& packet_time) { |
| // No packet receiver means that this demuxer will terminate the flow of |
| // packets. |
| if (!packet_receiver_) |
| return; |
| const uint8_t* const packet_data = packet->data(); |
| const size_t packet_length = packet->data_length(); |
| MediaType media_type = MediaType::ANY; |
| if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) { |
| RTC_CHECK_GE(packet_length, 2); |
| const uint8_t payload_type = packet_data[1] & 0x7f; |
| std::map<uint8_t, MediaType>::const_iterator it = |
| payload_type_map_.find(payload_type); |
| RTC_CHECK(it != payload_type_map_.end()) |
| << "payload type " << static_cast<int>(payload_type) << " unknown."; |
| media_type = it->second; |
| } |
| packet_receiver_->DeliverPacket( |
| media_type, rtc::CopyOnWriteBuffer(packet_data, packet_length), |
| packet_time); |
| } |
| |
| FakeNetworkPipe::FakeNetworkPipe(Clock* clock, |
| const FakeNetworkPipe::Config& config) |
| : FakeNetworkPipe(clock, config, nullptr, 1) {} |
| |
| FakeNetworkPipe::FakeNetworkPipe(Clock* clock, |
| const FakeNetworkPipe::Config& config, |
| std::unique_ptr<Demuxer> demuxer) |
| : FakeNetworkPipe(clock, config, std::move(demuxer), 1) {} |
| |
| FakeNetworkPipe::FakeNetworkPipe(Clock* clock, |
| const FakeNetworkPipe::Config& config, |
| std::unique_ptr<Demuxer> demuxer, |
| uint64_t seed) |
| : clock_(clock), |
| demuxer_(std::move(demuxer)), |
| receiver_(nullptr), |
| transport_(nullptr), |
| random_(seed), |
| config_(), |
| dropped_packets_(0), |
| sent_packets_(0), |
| total_packet_delay_(0), |
| bursting_(false), |
| next_process_time_(clock_->TimeInMilliseconds()), |
| last_log_time_(clock_->TimeInMilliseconds()) { |
| SetConfig(config); |
| } |
| |
| FakeNetworkPipe::FakeNetworkPipe(Clock* clock, |
| const FakeNetworkPipe::Config& config, |
| Transport* transport) |
| : clock_(clock), |
| receiver_(nullptr), |
| transport_(transport), |
| random_(1), |
| config_(), |
| dropped_packets_(0), |
| sent_packets_(0), |
| total_packet_delay_(0), |
| bursting_(false), |
| next_process_time_(clock_->TimeInMilliseconds()), |
| last_log_time_(clock_->TimeInMilliseconds()) { |
| SetConfig(config); |
| } |
| |
| FakeNetworkPipe::~FakeNetworkPipe() = default; |
| |
| void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) { |
| rtc::CritScope crit(&config_lock_); |
| if (demuxer_) |
| demuxer_->SetReceiver(receiver); |
| receiver_ = receiver; |
| } |
| |
| bool FakeNetworkPipe::SendRtp(const uint8_t* packet, |
| size_t length, |
| const PacketOptions& options) { |
| RTC_DCHECK(HasTransport()); |
| EnqueuePacket(rtc::CopyOnWriteBuffer(packet, length), options, false, |
| MediaType::ANY, rtc::nullopt); |
| return true; |
| } |
| |
| bool FakeNetworkPipe::SendRtcp(const uint8_t* packet, size_t length) { |
| RTC_DCHECK(HasTransport()); |
| EnqueuePacket(rtc::CopyOnWriteBuffer(packet, length), rtc::nullopt, true, |
| MediaType::ANY, rtc::nullopt); |
| return true; |
| } |
| |
| PacketReceiver::DeliveryStatus FakeNetworkPipe::DeliverPacket( |
| MediaType media_type, |
| rtc::CopyOnWriteBuffer packet, |
| const PacketTime& packet_time) { |
| return EnqueuePacket(std::move(packet), rtc::nullopt, false, media_type, |
| packet_time) |
| ? PacketReceiver::DELIVERY_OK |
| : PacketReceiver::DELIVERY_PACKET_ERROR; |
| } |
| |
| void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) { |
| rtc::CritScope crit(&config_lock_); |
| config_ = config; // Shallow copy of the struct. |
| double prob_loss = config.loss_percent / 100.0; |
| if (config_.avg_burst_loss_length == -1) { |
| // Uniform loss |
| prob_loss_bursting_ = prob_loss; |
| prob_start_bursting_ = prob_loss; |
| } else { |
| // Lose packets according to a gilbert-elliot model. |
| int avg_burst_loss_length = config.avg_burst_loss_length; |
| int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss)); |
| |
| RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length) |
| << "For a total packet loss of " << config.loss_percent << "%% then" |
| << " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1 |
| << " or higher."; |
| |
| prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length); |
| prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length; |
| } |
| } |
| |
| void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) { |
| RTC_DCHECK(HasDemuxer()); |
| EnqueuePacket(rtc::CopyOnWriteBuffer(data, data_length), rtc::nullopt, false, |
| MediaType::ANY, rtc::nullopt); |
| } |
| |
| bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet, |
| rtc::Optional<PacketOptions> options, |
| bool is_rtcp, |
| MediaType media_type, |
| rtc::Optional<PacketTime> packet_time) { |
| Config config; |
| { |
| rtc::CritScope crit(&config_lock_); |
| config = config_; |
| } |
| rtc::CritScope crit(&process_lock_); |
| if (config.queue_length_packets > 0 && |
| capacity_link_.size() >= config.queue_length_packets) { |
| // Too many packet on the link, drop this one. |
| ++dropped_packets_; |
| return false; |
| } |
| |
| int64_t time_now = clock_->TimeInMilliseconds(); |
| |
| // Delay introduced by the link capacity. |
| int64_t capacity_delay_ms = 0; |
| if (config.link_capacity_kbps > 0) { |
| const int bytes_per_millisecond = config.link_capacity_kbps / 8; |
| // To round to the closest millisecond we add half a milliseconds worth of |
| // bytes to the delay calculation. |
| capacity_delay_ms = (packet.size() + capacity_delay_error_bytes_ + |
| bytes_per_millisecond / 2) / |
| bytes_per_millisecond; |
| capacity_delay_error_bytes_ += |
| packet.size() - capacity_delay_ms * bytes_per_millisecond; |
| } |
| int64_t network_start_time = time_now; |
| |
| // Check if there already are packets on the link and change network start |
| // time forward if there is. |
| if (!capacity_link_.empty() && |
| network_start_time < capacity_link_.back().arrival_time()) |
| network_start_time = capacity_link_.back().arrival_time(); |
| |
| int64_t arrival_time = network_start_time + capacity_delay_ms; |
| capacity_link_.emplace(std::move(packet), time_now, arrival_time, options, |
| is_rtcp, media_type, packet_time); |
| return true; |
| } |
| |
| float FakeNetworkPipe::PercentageLoss() { |
| rtc::CritScope crit(&process_lock_); |
| if (sent_packets_ == 0) |
| return 0; |
| |
| return static_cast<float>(dropped_packets_) / |
| (sent_packets_ + dropped_packets_); |
| } |
| |
| int FakeNetworkPipe::AverageDelay() { |
| rtc::CritScope crit(&process_lock_); |
| if (sent_packets_ == 0) |
| return 0; |
| |
| return static_cast<int>(total_packet_delay_ / |
| static_cast<int64_t>(sent_packets_)); |
| } |
| |
| size_t FakeNetworkPipe::DroppedPackets() { |
| rtc::CritScope crit(&process_lock_); |
| return dropped_packets_; |
| } |
| |
| size_t FakeNetworkPipe::SentPackets() { |
| rtc::CritScope crit(&process_lock_); |
| return sent_packets_; |
| } |
| |
| void FakeNetworkPipe::Process() { |
| int64_t time_now = clock_->TimeInMilliseconds(); |
| std::queue<NetworkPacket> packets_to_deliver; |
| Config config; |
| double prob_loss_bursting; |
| double prob_start_bursting; |
| { |
| rtc::CritScope crit(&config_lock_); |
| config = config_; |
| prob_loss_bursting = prob_loss_bursting_; |
| prob_start_bursting = prob_start_bursting_; |
| } |
| { |
| rtc::CritScope crit(&process_lock_); |
| if (time_now - last_log_time_ > 5000) { |
| int64_t queueing_delay_ms = 0; |
| if (!capacity_link_.empty()) { |
| queueing_delay_ms = time_now - capacity_link_.front().send_time(); |
| } |
| RTC_LOG(LS_INFO) << "Network queue: " << queueing_delay_ms << " ms."; |
| last_log_time_ = time_now; |
| } |
| |
| // Check the capacity link first. |
| if (!capacity_link_.empty()) { |
| int64_t last_arrival_time = |
| delay_link_.empty() ? -1 : delay_link_.back().arrival_time(); |
| bool needs_sort = false; |
| while (!capacity_link_.empty() && |
| time_now >= capacity_link_.front().arrival_time()) { |
| // Time to get this packet. |
| NetworkPacket packet = std::move(capacity_link_.front()); |
| capacity_link_.pop(); |
| |
| // Drop packets at an average rate of |config_.loss_percent| with |
| // and average loss burst length of |config_.avg_burst_loss_length|. |
| if ((bursting_ && random_.Rand<double>() < prob_loss_bursting) || |
| (!bursting_ && random_.Rand<double>() < prob_start_bursting)) { |
| bursting_ = true; |
| continue; |
| } else { |
| bursting_ = false; |
| } |
| |
| int arrival_time_jitter = random_.Gaussian( |
| config.queue_delay_ms, config.delay_standard_deviation_ms); |
| |
| // If reordering is not allowed then adjust arrival_time_jitter |
| // to make sure all packets are sent in order. |
| if (!config.allow_reordering && !delay_link_.empty() && |
| packet.arrival_time() + arrival_time_jitter < last_arrival_time) { |
| arrival_time_jitter = last_arrival_time - packet.arrival_time(); |
| } |
| packet.IncrementArrivalTime(arrival_time_jitter); |
| if (packet.arrival_time() >= last_arrival_time) { |
| last_arrival_time = packet.arrival_time(); |
| } else { |
| needs_sort = true; |
| } |
| delay_link_.emplace_back(std::move(packet)); |
| } |
| |
| if (needs_sort) { |
| // Packet(s) arrived out of order, make sure list is sorted. |
| std::sort(delay_link_.begin(), delay_link_.end(), |
| PacketArrivalTimeComparator()); |
| } |
| } |
| |
| // Check the extra delay queue. |
| while (!delay_link_.empty() && |
| time_now >= delay_link_.front().arrival_time()) { |
| // Deliver this packet. |
| NetworkPacket packet(std::move(delay_link_.front())); |
| delay_link_.pop_front(); |
| // |time_now| might be later than when the packet should have arrived, due |
| // to NetworkProcess being called too late. For stats, use the time it |
| // should have been on the link. |
| total_packet_delay_ += packet.arrival_time() - packet.send_time(); |
| packets_to_deliver.push(std::move(packet)); |
| } |
| sent_packets_ += packets_to_deliver.size(); |
| } |
| |
| rtc::CritScope crit(&config_lock_); |
| while (!packets_to_deliver.empty()) { |
| NetworkPacket packet = std::move(packets_to_deliver.front()); |
| packets_to_deliver.pop(); |
| DeliverPacket(&packet); |
| } |
| |
| next_process_time_ = !delay_link_.empty() |
| ? delay_link_.begin()->arrival_time() |
| : time_now + kDefaultProcessIntervalMs; |
| } |
| |
| void FakeNetworkPipe::DeliverPacket(NetworkPacket* packet) { |
| if (demuxer_) { |
| demuxer_->DeliverPacket(packet, PacketTime()); |
| } else if (transport_) { |
| if (packet->is_rtcp()) { |
| transport_->SendRtcp(packet->data(), packet->data_length()); |
| } else { |
| transport_->SendRtp(packet->data(), packet->data_length(), |
| packet->packet_options()); |
| } |
| } else if (receiver_) { |
| PacketTime packet_time = packet->packet_time(); |
| if (packet_time.timestamp != -1) { |
| int64_t queue_time = packet->arrival_time() - packet->send_time(); |
| RTC_CHECK(queue_time >= 0); |
| packet_time.timestamp += (queue_time * 1000); |
| } |
| receiver_->DeliverPacket(packet->media_type(), |
| std::move(*packet->raw_packet()), packet_time); |
| } |
| } |
| |
| int64_t FakeNetworkPipe::TimeUntilNextProcess() { |
| rtc::CritScope crit(&process_lock_); |
| return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(), |
| 0); |
| } |
| |
| bool FakeNetworkPipe::HasTransport() const { |
| rtc::CritScope crit(&config_lock_); |
| return transport_ != nullptr; |
| } |
| |
| bool FakeNetworkPipe::HasDemuxer() const { |
| rtc::CritScope crit(&config_lock_); |
| return demuxer_ != nullptr; |
| } |
| |
| } // namespace webrtc |