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/*
* 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.
*/
#ifndef CALL_FAKE_NETWORK_PIPE_H_
#define CALL_FAKE_NETWORK_PIPE_H_
#include <deque>
#include <map>
#include <memory>
#include <queue>
#include <set>
#include <string>
#include <vector>
#include "api/call/transport.h"
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/simulated_packet_receiver.h"
#include "common_types.h" // NOLINT(build/include)
#include "rtc_base/constructormagic.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class Clock;
class PacketReceiver;
enum class MediaType;
class NetworkPacket {
public:
NetworkPacket(rtc::CopyOnWriteBuffer packet,
int64_t send_time,
int64_t arrival_time,
absl::optional<PacketOptions> packet_options,
bool is_rtcp,
MediaType media_type,
absl::optional<int64_t> packet_time_us);
// Disallow copy constructor and copy assignment (no deep copies of |data_|).
NetworkPacket(const NetworkPacket&) = delete;
~NetworkPacket();
NetworkPacket& operator=(const NetworkPacket&) = delete;
// Allow move constructor/assignment, so that we can use in stl containers.
NetworkPacket(NetworkPacket&&);
NetworkPacket& operator=(NetworkPacket&&);
const uint8_t* data() const { return packet_.data(); }
size_t data_length() const { return packet_.size(); }
rtc::CopyOnWriteBuffer* raw_packet() { return &packet_; }
int64_t send_time() const { return send_time_; }
int64_t arrival_time() const { return arrival_time_; }
void IncrementArrivalTime(int64_t extra_delay) {
arrival_time_ += extra_delay;
}
PacketOptions packet_options() const {
return packet_options_.value_or(PacketOptions());
}
bool is_rtcp() const { return is_rtcp_; }
MediaType media_type() const { return media_type_; }
absl::optional<int64_t> packet_time_us() const { return packet_time_us_; }
private:
rtc::CopyOnWriteBuffer packet_;
// The time the packet was sent out on the network.
int64_t send_time_;
// The time the packet should arrive at the receiver.
int64_t arrival_time_;
// If using a Transport for outgoing degradation, populate with
// PacketOptions (transport-wide sequence number) for RTP.
absl::optional<PacketOptions> packet_options_;
bool is_rtcp_;
// If using a PacketReceiver for incoming degradation, populate with
// appropriate MediaType and PacketTime. This type/timing will be kept and
// forwarded. The PacketTime might be altered to reflect time spent in fake
// network pipe.
MediaType media_type_;
absl::optional<int64_t> packet_time_us_;
};
// Class faking a network link, internally is uses an implementation of a
// SimulatedNetworkInterface to simulate network behavior.
class FakeNetworkPipe : public webrtc::SimulatedPacketReceiverInterface,
public Transport {
public:
// Will keep |network_behavior| alive while pipe is alive itself.
// Use these constructors if you plan to insert packets using DeliverPacket().
FakeNetworkPipe(Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior);
FakeNetworkPipe(Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior,
PacketReceiver* receiver);
FakeNetworkPipe(Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior,
PacketReceiver* receiver,
uint64_t seed);
// Use this constructor if you plan to insert packets using SendRt[c?]p().
FakeNetworkPipe(Clock* clock,
std::unique_ptr<NetworkBehaviorInterface> network_behavior,
Transport* transport);
~FakeNetworkPipe() override;
void SetClockOffset(int64_t offset_ms);
// Must not be called in parallel with DeliverPacket or Process.
void SetReceiver(PacketReceiver* receiver) override;
// Implements Transport interface. When/if packets are delivered, they will
// be passed to the transport instance given in SetReceiverTransport(). These
// methods should only be called if a Transport instance was provided in the
// constructor.
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override;
bool SendRtcp(const uint8_t* packet, size_t length) override;
// Implements the PacketReceiver interface. When/if packets are delivered,
// they will be passed directly to the receiver instance given in
// SetReceiver(), without passing through a Demuxer. The receive time in
// PacketTime will be increased by the amount of time the packet spent in the
// fake network pipe.
PacketReceiver::DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) override;
// TODO(bugs.webrtc.org/9584): Needed to inherit the alternative signature for
// this method.
using PacketReceiver::DeliverPacket;
// Processes the network queues and trigger PacketReceiver::IncomingPacket for
// packets ready to be delivered.
void Process() override;
int64_t TimeUntilNextProcess() override;
// Get statistics.
float PercentageLoss();
int AverageDelay() override;
size_t DroppedPackets();
size_t SentPackets();
void ResetStats();
protected:
void DeliverPacketWithLock(NetworkPacket* packet);
void AddToPacketDropCount();
void AddToPacketSentCount(int count);
void AddToTotalDelay(int delay_us);
int64_t GetTimeInMicroseconds() const;
bool ShouldProcess(int64_t time_now_us) const;
void SetTimeToNextProcess(int64_t skip_us);
private:
struct StoredPacket {
NetworkPacket packet;
bool removed = false;
explicit StoredPacket(NetworkPacket&& packet);
StoredPacket(StoredPacket&&) = default;
StoredPacket(const StoredPacket&) = delete;
StoredPacket& operator=(const StoredPacket&) = delete;
StoredPacket() = delete;
};
// Returns true if enqueued, or false if packet was dropped.
virtual bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
absl::optional<PacketOptions> options,
bool is_rtcp,
MediaType media_type,
absl::optional<int64_t> packet_time_us);
bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
absl::optional<PacketOptions> options,
bool is_rtcp,
MediaType media_type) {
return EnqueuePacket(packet, options, is_rtcp, media_type, absl::nullopt);
}
void DeliverNetworkPacket(NetworkPacket* packet)
RTC_EXCLUSIVE_LOCKS_REQUIRED(config_lock_);
bool HasTransport() const;
bool HasReceiver() const;
Clock* const clock_;
// |config_lock| guards the mostly constant things like the callbacks.
rtc::CriticalSection config_lock_;
const std::unique_ptr<NetworkBehaviorInterface> network_behavior_;
PacketReceiver* receiver_ RTC_GUARDED_BY(config_lock_);
Transport* const transport_ RTC_GUARDED_BY(config_lock_);
// |process_lock| guards the data structures involved in delay and loss
// processes, such as the packet queues.
rtc::CriticalSection process_lock_;
// Packets are added at the back of the deque, this makes the deque ordered
// by increasing send time. The common case when removing packets from the
// deque is removing early packets, which will be close to the front of the
// deque. This makes finding the packets in the deque efficient in the common
// case.
std::deque<StoredPacket> packets_in_flight_ RTC_GUARDED_BY(process_lock_);
int64_t clock_offset_ms_ RTC_GUARDED_BY(config_lock_);
// Statistics.
size_t dropped_packets_ RTC_GUARDED_BY(process_lock_);
size_t sent_packets_ RTC_GUARDED_BY(process_lock_);
int64_t total_packet_delay_us_ RTC_GUARDED_BY(process_lock_);
int64_t next_process_time_us_;
int64_t last_log_time_us_;
RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
};
} // namespace webrtc
#endif // CALL_FAKE_NETWORK_PIPE_H_