| /* |
| * 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 "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_ |