/* * 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 #include "api/call/transport.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/test/simulated_network.h" #include "call/call.h" #include "call/simulated_packet_receiver.h" #include "rtc_base/synchronization/mutex.h" #include "rtc_base/task_utils/repeating_task.h" #include "rtc_base/thread_annotations.h" namespace webrtc { class PacketReceiver; namespace test { class Demuxer { public: explicit Demuxer(const std::map& payload_type_map); ~Demuxer() = default; Demuxer(const Demuxer&) = delete; Demuxer& operator=(const Demuxer&) = delete; MediaType GetMediaType(const uint8_t* packet_data, size_t packet_length) const; const std::map payload_type_map_; }; // 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(TaskQueueBase* task_queue, std::unique_ptr pipe, Call* send_call, const std::map& payload_type_map, rtc::ArrayView audio_extensions, rtc::ArrayView video_extensions); ~DirectTransport() override; // 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 LegacySendPacket(const uint8_t* data, size_t length); void Start(); Call* const send_call_; TaskQueueBase* const task_queue_; Mutex process_lock_; RepeatingTaskHandle next_process_task_ RTC_GUARDED_BY(&process_lock_); const Demuxer demuxer_; const std::unique_ptr fake_network_; const RtpHeaderExtensionMap audio_extensions_; const RtpHeaderExtensionMap video_extensions_; }; } // namespace test } // namespace webrtc #endif // TEST_DIRECT_TRANSPORT_H_