call/fake_network_pipe.h - src.git - Git at Google

 /*
  *  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;
   void ProcessThreadAttached(ProcessThread* process_thread) override;

   // Get statistics.
   float PercentageLoss();
   int AverageDelay() override;
   size_t DroppedPackets();
   size_t SentPackets();
   void ResetStats();

  protected:
   void DeliverPacketWithLock(NetworkPacket* packet);
   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_;

   rtc::CriticalSection process_thread_lock_;
   ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_) = nullptr;

   // 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 last_log_time_us_;

   RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
 };

 }  // namespace webrtc

 #endif  // CALL_FAKE_NETWORK_PIPE_H_
	/*
	* 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;
	void ProcessThreadAttached(ProcessThread* process_thread) override;

	// Get statistics.
	float PercentageLoss();
	int AverageDelay() override;
	size_t DroppedPackets();
	size_t SentPackets();
	void ResetStats();

	protected:
	void DeliverPacketWithLock(NetworkPacket* packet);
	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_;

	rtc::CriticalSection process_thread_lock_;
	ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_) = nullptr;

	// 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 last_log_time_us_;

	RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
	};

	} // namespace webrtc

	#endif // CALL_FAKE_NETWORK_PIPE_H_