| /* | 
 |  *  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 <cstddef> | 
 | #include <cstdint> | 
 | #include <deque> | 
 | #include <map> | 
 | #include <memory> | 
 | #include <optional> | 
 |  | 
 | #include "api/array_view.h" | 
 | #include "api/call/transport.h" | 
 | #include "api/test/simulated_network.h" | 
 | #include "call/simulated_packet_receiver.h" | 
 | #include "modules/rtp_rtcp/source/rtp_packet_received.h" | 
 | #include "rtc_base/copy_on_write_buffer.h" | 
 | #include "rtc_base/synchronization/mutex.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, | 
 |                 std::optional<PacketOptions> packet_options, | 
 |                 bool is_rtcp, | 
 |                 MediaType media_type, | 
 |                 std::optional<int64_t> packet_time_us, | 
 |                 Transport* transport); | 
 |  | 
 |   NetworkPacket(RtpPacketReceived packet, | 
 |                 MediaType media_type, | 
 |                 int64_t send_time, | 
 |                 int64_t arrival_time); | 
 |  | 
 |   // 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_; } | 
 |   std::optional<int64_t> packet_time_us() const { return packet_time_us_; } | 
 |   RtpPacketReceived* packet_received() { | 
 |     return packet_received_ ? &packet_received_.value() : nullptr; | 
 |   } | 
 |   std::optional<RtpPacketReceived> packet_received() const { | 
 |     return packet_received_; | 
 |   } | 
 |   Transport* transport() const { return transport_; } | 
 |  | 
 |  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. | 
 |   std::optional<PacketOptions> packet_options_; | 
 |   bool is_rtcp_; | 
 |   // If using a PacketReceiver for incoming degradation, populate with | 
 |   // appropriate MediaType and packet time. This type/timing will be kept and | 
 |   // forwarded. The packet time might be altered to reflect time spent in fake | 
 |   // network pipe. | 
 |   MediaType media_type_; | 
 |   std::optional<int64_t> packet_time_us_; | 
 |   std::optional<RtpPacketReceived> packet_received_; | 
 |   Transport* transport_; | 
 | }; | 
 |  | 
 | // Class faking a network link, internally is uses an implementation of a | 
 | // SimulatedNetworkInterface to simulate network behavior. | 
 | class FakeNetworkPipe : public SimulatedPacketReceiverInterface { | 
 |  public: | 
 |   // Will keep `network_behavior` alive while pipe is alive itself. | 
 |   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); | 
 |  | 
 |   ~FakeNetworkPipe() override; | 
 |  | 
 |   FakeNetworkPipe(const FakeNetworkPipe&) = delete; | 
 |   FakeNetworkPipe& operator=(const FakeNetworkPipe&) = delete; | 
 |  | 
 |   void SetClockOffset(int64_t offset_ms); | 
 |  | 
 |   // Must not be called in parallel with DeliverPacket or Process. | 
 |   void SetReceiver(PacketReceiver* receiver) override; | 
 |  | 
 |   // Adds/subtracts references to Transport instances. If a Transport is | 
 |   // destroyed we cannot use to forward a potential delayed packet, these | 
 |   // methods are used to maintain a map of which instances are live. | 
 |   void AddActiveTransport(Transport* transport); | 
 |   void RemoveActiveTransport(Transport* transport); | 
 |  | 
 |   // Methods for use with Transport interface. When/if packets are delivered, | 
 |   // they will be passed to the instance specified by the `transport` parameter. | 
 |   // Note that that instance must be in the map of active transports. | 
 |   bool SendRtp(rtc::ArrayView<const uint8_t> packet, | 
 |                const PacketOptions& options, | 
 |                Transport* transport); | 
 |   bool SendRtcp(rtc::ArrayView<const uint8_t> packet, Transport* transport); | 
 |  | 
 |   // Implements the PacketReceiver interface. When/if packets are delivered, | 
 |   // they will be passed directly to the receiver instance given in | 
 |   // SetReceiver(). The receive time will be increased by the amount of time the | 
 |   // packet spent in the fake network pipe. | 
 |   void DeliverRtpPacket( | 
 |       MediaType media_type, | 
 |       RtpPacketReceived packet, | 
 |       OnUndemuxablePacketHandler undemuxable_packet_handler) override; | 
 |   void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override; | 
 |  | 
 |   // Processes the network queues and trigger PacketReceiver::IncomingPacket for | 
 |   // packets ready to be delivered. | 
 |   void Process() override; | 
 |   std::optional<int64_t> TimeUntilNextProcess() 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. Use this method | 
 |   // when enqueueing packets that should be received by PacketReceiver instance. | 
 |   bool EnqueuePacket(rtc::CopyOnWriteBuffer packet, | 
 |                      std::optional<PacketOptions> options, | 
 |                      bool is_rtcp, | 
 |                      MediaType media_type, | 
 |                      std::optional<int64_t> packet_time_us); | 
 |  | 
 |   // Returns true if enqueued, or false if packet was dropped. Use this method | 
 |   // when enqueueing packets that should be received by Transport instance. | 
 |   bool EnqueuePacket(rtc::CopyOnWriteBuffer packet, | 
 |                      std::optional<PacketOptions> options, | 
 |                      bool is_rtcp, | 
 |                      Transport* transport); | 
 |  | 
 |   bool EnqueuePacket(NetworkPacket&& net_packet) | 
 |       RTC_EXCLUSIVE_LOCKS_REQUIRED(process_lock_); | 
 |  | 
 |   void DeliverNetworkPacket(NetworkPacket* packet) | 
 |       RTC_EXCLUSIVE_LOCKS_REQUIRED(config_lock_); | 
 |   bool HasReceiver() const; | 
 |  | 
 |   Clock* const clock_; | 
 |   // `config_lock` guards the mostly constant things like the callbacks. | 
 |   mutable Mutex config_lock_; | 
 |   const std::unique_ptr<NetworkBehaviorInterface> network_behavior_; | 
 |   PacketReceiver* receiver_ RTC_GUARDED_BY(config_lock_); | 
 |  | 
 |   // `process_lock` guards the data structures involved in delay and loss | 
 |   // processes, such as the packet queues. | 
 |   Mutex 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 last_log_time_us_; | 
 |  | 
 |   std::map<Transport*, size_t> active_transports_ RTC_GUARDED_BY(config_lock_); | 
 | }; | 
 |  | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // CALL_FAKE_NETWORK_PIPE_H_ |