8ef59a431f
Give internal test tools access to FakeNetworkPipe data members by adding a set of access methods. Also deleted copy assignment operator for NetworkPacket. Bug: None Change-Id: I451a21e0cc6ec82ea830cf197c7a4cef0789623c Reviewed-on: https://webrtc-review.googlesource.com/63301 Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22515}
468 lines
16 KiB
C++
468 lines
16 KiB
C++
/*
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* Copyright (c) 2012 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 <assert.h>
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#include <math.h>
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#include <string.h>
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#include <algorithm>
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#include <cmath>
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#include <utility>
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#include "call/call.h"
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#include "call/fake_network_pipe.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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namespace {
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constexpr int64_t kDefaultProcessIntervalMs = 5;
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struct PacketArrivalTimeComparator {
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bool operator()(const NetworkPacket& p1, const NetworkPacket& p2) {
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return p1.arrival_time() < p2.arrival_time();
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}
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};
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} // namespace
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NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet,
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int64_t send_time,
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int64_t arrival_time,
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rtc::Optional<PacketOptions> packet_options,
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bool is_rtcp,
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MediaType media_type,
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rtc::Optional<PacketTime> packet_time)
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: packet_(std::move(packet)),
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send_time_(send_time),
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arrival_time_(arrival_time),
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packet_options_(packet_options),
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is_rtcp_(is_rtcp),
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media_type_(media_type),
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packet_time_(packet_time) {}
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NetworkPacket::NetworkPacket(NetworkPacket&& o)
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: packet_(std::move(o.packet_)),
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send_time_(o.send_time_),
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arrival_time_(o.arrival_time_),
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packet_options_(o.packet_options_),
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is_rtcp_(o.is_rtcp_),
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media_type_(o.media_type_),
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packet_time_(o.packet_time_) {}
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NetworkPacket& NetworkPacket::operator=(NetworkPacket&& o) {
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packet_ = std::move(o.packet_);
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send_time_ = o.send_time_;
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arrival_time_ = o.arrival_time_;
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packet_options_ = o.packet_options_;
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is_rtcp_ = o.is_rtcp_;
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media_type_ = o.media_type_;
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packet_time_ = o.packet_time_;
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return *this;
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}
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DemuxerImpl::DemuxerImpl(const std::map<uint8_t, MediaType>& payload_type_map)
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: packet_receiver_(nullptr), payload_type_map_(payload_type_map) {}
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void DemuxerImpl::SetReceiver(PacketReceiver* receiver) {
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packet_receiver_ = receiver;
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}
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void DemuxerImpl::DeliverPacket(const NetworkPacket* packet,
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const PacketTime& packet_time) {
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// No packet receiver means that this demuxer will terminate the flow of
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// packets.
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if (!packet_receiver_)
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return;
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const uint8_t* const packet_data = packet->data();
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const size_t packet_length = packet->data_length();
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MediaType media_type = MediaType::ANY;
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if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
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RTC_CHECK_GE(packet_length, 2);
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const uint8_t payload_type = packet_data[1] & 0x7f;
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std::map<uint8_t, MediaType>::const_iterator it =
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payload_type_map_.find(payload_type);
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RTC_CHECK(it != payload_type_map_.end())
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<< "payload type " << static_cast<int>(payload_type) << " unknown.";
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media_type = it->second;
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}
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packet_receiver_->DeliverPacket(
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media_type, rtc::CopyOnWriteBuffer(packet_data, packet_length),
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packet_time);
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}
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FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
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const FakeNetworkPipe::Config& config)
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: FakeNetworkPipe(clock, config, nullptr, 1) {}
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FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
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const FakeNetworkPipe::Config& config,
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std::unique_ptr<Demuxer> demuxer)
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: FakeNetworkPipe(clock, config, std::move(demuxer), 1) {}
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FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
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const FakeNetworkPipe::Config& config,
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std::unique_ptr<Demuxer> demuxer,
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uint64_t seed)
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: clock_(clock),
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demuxer_(std::move(demuxer)),
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receiver_(nullptr),
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transport_(nullptr),
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random_(seed),
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config_(),
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dropped_packets_(0),
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sent_packets_(0),
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total_packet_delay_(0),
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bursting_(false),
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next_process_time_(clock_->TimeInMilliseconds()),
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last_log_time_(clock_->TimeInMilliseconds()) {
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SetConfig(config);
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}
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FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
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const FakeNetworkPipe::Config& config,
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Transport* transport)
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: clock_(clock),
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receiver_(nullptr),
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transport_(transport),
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random_(1),
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config_(),
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dropped_packets_(0),
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sent_packets_(0),
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total_packet_delay_(0),
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bursting_(false),
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next_process_time_(clock_->TimeInMilliseconds()),
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last_log_time_(clock_->TimeInMilliseconds()) {
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SetConfig(config);
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}
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FakeNetworkPipe::~FakeNetworkPipe() = default;
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void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
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rtc::CritScope crit(&config_lock_);
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if (demuxer_)
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demuxer_->SetReceiver(receiver);
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receiver_ = receiver;
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}
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bool FakeNetworkPipe::SendRtp(const uint8_t* packet,
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size_t length,
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const PacketOptions& options) {
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RTC_DCHECK(HasTransport());
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EnqueuePacket(rtc::CopyOnWriteBuffer(packet, length), options, false,
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MediaType::ANY, rtc::nullopt);
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return true;
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}
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bool FakeNetworkPipe::SendRtcp(const uint8_t* packet, size_t length) {
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RTC_DCHECK(HasTransport());
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EnqueuePacket(rtc::CopyOnWriteBuffer(packet, length), rtc::nullopt, true,
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MediaType::ANY, rtc::nullopt);
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return true;
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}
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PacketReceiver::DeliveryStatus FakeNetworkPipe::DeliverPacket(
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MediaType media_type,
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rtc::CopyOnWriteBuffer packet,
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const PacketTime& packet_time) {
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return EnqueuePacket(std::move(packet), rtc::nullopt, false, media_type,
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packet_time)
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? PacketReceiver::DELIVERY_OK
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: PacketReceiver::DELIVERY_PACKET_ERROR;
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}
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void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
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rtc::CritScope crit(&config_lock_);
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config_ = config; // Shallow copy of the struct.
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double prob_loss = config.loss_percent / 100.0;
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if (config_.avg_burst_loss_length == -1) {
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// Uniform loss
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prob_loss_bursting_ = prob_loss;
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prob_start_bursting_ = prob_loss;
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} else {
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// Lose packets according to a gilbert-elliot model.
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int avg_burst_loss_length = config.avg_burst_loss_length;
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int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));
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RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
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<< "For a total packet loss of " << config.loss_percent << "%% then"
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<< " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1
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<< " or higher.";
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prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length);
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prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length;
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}
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}
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void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
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RTC_DCHECK(HasDemuxer());
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EnqueuePacket(rtc::CopyOnWriteBuffer(data, data_length), rtc::nullopt, false,
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MediaType::ANY, rtc::nullopt);
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}
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bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
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rtc::Optional<PacketOptions> options,
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bool is_rtcp,
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MediaType media_type,
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rtc::Optional<PacketTime> packet_time) {
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Config config;
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{
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rtc::CritScope crit(&config_lock_);
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config = config_;
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}
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rtc::CritScope crit(&process_lock_);
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if (config.queue_length_packets > 0 &&
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capacity_link_.size() >= config.queue_length_packets) {
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// Too many packet on the link, drop this one.
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++dropped_packets_;
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return false;
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}
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int64_t time_now = clock_->TimeInMilliseconds();
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// Delay introduced by the link capacity.
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int64_t capacity_delay_ms = 0;
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if (config.link_capacity_kbps > 0) {
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const int bytes_per_millisecond = config.link_capacity_kbps / 8;
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// To round to the closest millisecond we add half a milliseconds worth of
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// bytes to the delay calculation.
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capacity_delay_ms = (packet.size() + capacity_delay_error_bytes_ +
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bytes_per_millisecond / 2) /
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bytes_per_millisecond;
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capacity_delay_error_bytes_ +=
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packet.size() - capacity_delay_ms * bytes_per_millisecond;
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}
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int64_t network_start_time = time_now;
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// Check if there already are packets on the link and change network start
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// time forward if there is.
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if (!capacity_link_.empty() &&
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network_start_time < capacity_link_.back().arrival_time())
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network_start_time = capacity_link_.back().arrival_time();
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int64_t arrival_time = network_start_time + capacity_delay_ms;
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capacity_link_.emplace(std::move(packet), time_now, arrival_time, options,
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is_rtcp, media_type, packet_time);
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return true;
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}
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float FakeNetworkPipe::PercentageLoss() {
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rtc::CritScope crit(&process_lock_);
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if (sent_packets_ == 0)
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return 0;
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return static_cast<float>(dropped_packets_) /
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(sent_packets_ + dropped_packets_);
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}
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int FakeNetworkPipe::AverageDelay() {
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rtc::CritScope crit(&process_lock_);
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if (sent_packets_ == 0)
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return 0;
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return static_cast<int>(total_packet_delay_ /
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static_cast<int64_t>(sent_packets_));
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}
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size_t FakeNetworkPipe::DroppedPackets() {
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rtc::CritScope crit(&process_lock_);
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return dropped_packets_;
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}
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size_t FakeNetworkPipe::SentPackets() {
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rtc::CritScope crit(&process_lock_);
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return sent_packets_;
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}
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void FakeNetworkPipe::Process() {
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int64_t time_now = clock_->TimeInMilliseconds();
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std::queue<NetworkPacket> packets_to_deliver;
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Config config;
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double prob_loss_bursting;
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double prob_start_bursting;
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{
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rtc::CritScope crit(&config_lock_);
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config = config_;
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prob_loss_bursting = prob_loss_bursting_;
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prob_start_bursting = prob_start_bursting_;
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}
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{
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rtc::CritScope crit(&process_lock_);
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if (time_now - last_log_time_ > 5000) {
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int64_t queueing_delay_ms = 0;
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if (!capacity_link_.empty()) {
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queueing_delay_ms = time_now - capacity_link_.front().send_time();
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}
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RTC_LOG(LS_INFO) << "Network queue: " << queueing_delay_ms << " ms.";
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last_log_time_ = time_now;
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}
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// Check the capacity link first.
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if (!capacity_link_.empty()) {
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int64_t last_arrival_time =
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delay_link_.empty() ? -1 : delay_link_.back().arrival_time();
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bool needs_sort = false;
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while (!capacity_link_.empty() &&
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time_now >= capacity_link_.front().arrival_time()) {
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// Time to get this packet.
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NetworkPacket packet = std::move(capacity_link_.front());
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capacity_link_.pop();
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// Drop packets at an average rate of |config_.loss_percent| with
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// and average loss burst length of |config_.avg_burst_loss_length|.
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if ((bursting_ && random_.Rand<double>() < prob_loss_bursting) ||
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(!bursting_ && random_.Rand<double>() < prob_start_bursting)) {
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bursting_ = true;
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continue;
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} else {
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bursting_ = false;
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}
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int arrival_time_jitter = random_.Gaussian(
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config.queue_delay_ms, config.delay_standard_deviation_ms);
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// If reordering is not allowed then adjust arrival_time_jitter
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// to make sure all packets are sent in order.
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if (!config.allow_reordering && !delay_link_.empty() &&
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packet.arrival_time() + arrival_time_jitter < last_arrival_time) {
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arrival_time_jitter = last_arrival_time - packet.arrival_time();
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}
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packet.IncrementArrivalTime(arrival_time_jitter);
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if (packet.arrival_time() >= last_arrival_time) {
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last_arrival_time = packet.arrival_time();
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} else {
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needs_sort = true;
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}
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delay_link_.emplace_back(std::move(packet));
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}
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if (needs_sort) {
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// Packet(s) arrived out of order, make sure list is sorted.
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std::sort(delay_link_.begin(), delay_link_.end(),
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PacketArrivalTimeComparator());
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}
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}
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// Check the extra delay queue.
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while (!delay_link_.empty() &&
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time_now >= delay_link_.front().arrival_time()) {
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// Deliver this packet.
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NetworkPacket packet(std::move(delay_link_.front()));
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delay_link_.pop_front();
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// |time_now| might be later than when the packet should have arrived, due
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// to NetworkProcess being called too late. For stats, use the time it
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// should have been on the link.
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total_packet_delay_ += packet.arrival_time() - packet.send_time();
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packets_to_deliver.push(std::move(packet));
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}
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sent_packets_ += packets_to_deliver.size();
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}
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rtc::CritScope crit(&config_lock_);
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while (!packets_to_deliver.empty()) {
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NetworkPacket packet = std::move(packets_to_deliver.front());
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packets_to_deliver.pop();
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DeliverPacket(&packet);
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}
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next_process_time_ = !delay_link_.empty()
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? delay_link_.begin()->arrival_time()
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: time_now + kDefaultProcessIntervalMs;
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}
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void FakeNetworkPipe::DeliverPacket(NetworkPacket* packet) {
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if (demuxer_) {
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demuxer_->DeliverPacket(packet, PacketTime());
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} else if (transport_) {
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if (packet->is_rtcp()) {
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transport_->SendRtcp(packet->data(), packet->data_length());
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} else {
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transport_->SendRtp(packet->data(), packet->data_length(),
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packet->packet_options());
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}
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} else if (receiver_) {
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PacketTime packet_time = packet->packet_time();
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if (packet_time.timestamp != -1) {
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int64_t queue_time = packet->arrival_time() - packet->send_time();
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RTC_CHECK(queue_time >= 0);
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packet_time.timestamp += (queue_time * 1000);
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}
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receiver_->DeliverPacket(packet->media_type(),
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std::move(*packet->raw_packet()), packet_time);
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}
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}
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int64_t FakeNetworkPipe::TimeUntilNextProcess() {
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rtc::CritScope crit(&process_lock_);
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return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
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0);
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}
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bool FakeNetworkPipe::HasTransport() const {
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rtc::CritScope crit(&config_lock_);
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return transport_ != nullptr;
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}
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bool FakeNetworkPipe::HasDemuxer() const {
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rtc::CritScope crit(&config_lock_);
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return demuxer_ != nullptr;
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}
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void FakeNetworkPipe::DeliverPacketWithLock(NetworkPacket* packet) {
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rtc::CritScope crit(&config_lock_);
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DeliverPacket(packet);
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}
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void FakeNetworkPipe::ResetStats() {
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rtc::CritScope crit(&process_lock_);
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dropped_packets_ = 0;
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sent_packets_ = 0;
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total_packet_delay_ = 0;
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}
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int FakeNetworkPipe::GetConfigCapacityKbps() const {
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rtc::CritScope crit(&config_lock_);
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return config_.link_capacity_kbps;
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}
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void FakeNetworkPipe::AddToPacketDropCount() {
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rtc::CritScope crit(&process_lock_);
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++dropped_packets_;
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}
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void FakeNetworkPipe::AddToPacketSentCount(int count) {
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rtc::CritScope crit(&process_lock_);
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sent_packets_ += count;
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}
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void FakeNetworkPipe::AddToTotalDelay(int delay_ms) {
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rtc::CritScope crit(&process_lock_);
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total_packet_delay_ += delay_ms;
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}
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int64_t FakeNetworkPipe::GetTimeInMilliseconds() const {
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return clock_->TimeInMilliseconds();
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}
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bool FakeNetworkPipe::IsRandomLoss(double prob_loss) {
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return random_.Rand<double>() < prob_loss;
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}
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bool FakeNetworkPipe::ShouldProcess(int64_t time_now) const {
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return time_now >= next_process_time_;
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}
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void FakeNetworkPipe::SetTimeToNextProcess(int64_t skip_ms) {
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next_process_time_ += skip_ms;
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}
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} // namespace webrtc
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