| /* |
| * Copyright (c) 2013 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 TEST_DIRECT_TRANSPORT_H_ |
| #define TEST_DIRECT_TRANSPORT_H_ |
| |
| #include <memory> |
| |
| #include "api/call/transport.h" |
| #include "api/test/simulated_network.h" |
| #include "call/call.h" |
| #include "call/simulated_packet_receiver.h" |
| #include "rtc_base/sequenced_task_checker.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "test/single_threaded_task_queue.h" |
| |
| namespace webrtc { |
| |
| class Clock; |
| class PacketReceiver; |
| |
| namespace test { |
| class Demuxer { |
| public: |
| explicit Demuxer(const std::map<uint8_t, MediaType>& payload_type_map); |
| ~Demuxer() = default; |
| MediaType GetMediaType(const uint8_t* packet_data, |
| const size_t packet_length) const; |
| const std::map<uint8_t, MediaType> payload_type_map_; |
| RTC_DISALLOW_COPY_AND_ASSIGN(Demuxer); |
| }; |
| |
| // Objects of this class are expected to be allocated and destroyed on the |
| // same task-queue - the one that's passed in via the constructor. |
| class DirectTransport : public Transport { |
| public: |
| DirectTransport(SingleThreadedTaskQueueForTesting* task_queue, |
| std::unique_ptr<SimulatedPacketReceiverInterface> pipe, |
| Call* send_call, |
| const std::map<uint8_t, MediaType>& payload_type_map); |
| |
| ~DirectTransport() override; |
| |
| RTC_DEPRECATED void StopSending(); |
| |
| // TODO(holmer): Look into moving this to the constructor. |
| virtual void SetReceiver(PacketReceiver* receiver); |
| |
| bool SendRtp(const uint8_t* data, |
| size_t length, |
| const PacketOptions& options) override; |
| bool SendRtcp(const uint8_t* data, size_t length) override; |
| |
| int GetAverageDelayMs(); |
| |
| private: |
| void ProcessPackets() RTC_EXCLUSIVE_LOCKS_REQUIRED(&process_lock_); |
| void SendPacket(const uint8_t* data, size_t length); |
| void Start(); |
| |
| Call* const send_call_; |
| Clock* const clock_; |
| |
| SingleThreadedTaskQueueForTesting* const task_queue_; |
| |
| rtc::CriticalSection process_lock_; |
| absl::optional<SingleThreadedTaskQueueForTesting::TaskId> next_process_task_ |
| RTC_GUARDED_BY(&process_lock_); |
| |
| const Demuxer demuxer_; |
| const std::unique_ptr<SimulatedPacketReceiverInterface> fake_network_; |
| |
| }; |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // TEST_DIRECT_TRANSPORT_H_ |