webrtc_m130/call/fake_network_pipe.cc
Sebastian Jansson 7e85d67031 Added SetClockOffset on FakeNetworkPipe.
Added functionality on the FakeNetworkPipe to introduce arbitrary
clock offsets. This offset is added to the reported receive time of
all packets. This prepares for a later CL using this to test correction
of receive time stamps.

Bug: webrtc:9054
Change-Id: I811b3aa8359bc917f59443088d8a418368242db9
Reviewed-on: https://webrtc-review.googlesource.com/64726
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22763}
2018-04-06 09:02:12 +00:00

476 lines
16 KiB
C++

/*
* 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),
clock_offset_ms_(0),
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),
clock_offset_ms_(0),
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::SetClockOffset(int64_t offset_ms) {
rtc::CritScope crit(&config_lock_);
clock_offset_ms_ = offset_ms;
}
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);
packet_time.timestamp += (clock_offset_ms_ * 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;
}
void FakeNetworkPipe::DeliverPacketWithLock(NetworkPacket* packet) {
rtc::CritScope crit(&config_lock_);
DeliverPacket(packet);
}
void FakeNetworkPipe::ResetStats() {
rtc::CritScope crit(&process_lock_);
dropped_packets_ = 0;
sent_packets_ = 0;
total_packet_delay_ = 0;
}
int FakeNetworkPipe::GetConfigCapacityKbps() const {
rtc::CritScope crit(&config_lock_);
return config_.link_capacity_kbps;
}
void FakeNetworkPipe::AddToPacketDropCount() {
rtc::CritScope crit(&process_lock_);
++dropped_packets_;
}
void FakeNetworkPipe::AddToPacketSentCount(int count) {
rtc::CritScope crit(&process_lock_);
sent_packets_ += count;
}
void FakeNetworkPipe::AddToTotalDelay(int delay_ms) {
rtc::CritScope crit(&process_lock_);
total_packet_delay_ += delay_ms;
}
int64_t FakeNetworkPipe::GetTimeInMilliseconds() const {
return clock_->TimeInMilliseconds();
}
bool FakeNetworkPipe::IsRandomLoss(double prob_loss) {
return random_.Rand<double>() < prob_loss;
}
bool FakeNetworkPipe::ShouldProcess(int64_t time_now) const {
return time_now >= next_process_time_;
}
void FakeNetworkPipe::SetTimeToNextProcess(int64_t skip_ms) {
next_process_time_ += skip_ms;
}
} // namespace webrtc