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/*
* 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/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<uint8_t, MediaType>& 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<uint8_t, MediaType> 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<SimulatedPacketReceiverInterface> pipe,
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map,
rtc::ArrayView<const RtpExtension> audio_extensions,
rtc::ArrayView<const RtpExtension> 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<SimulatedPacketReceiverInterface> fake_network_;
const RtpHeaderExtensionMap audio_extensions_;
const RtpHeaderExtensionMap video_extensions_;
};
} // namespace test
} // namespace webrtc
#endif // TEST_DIRECT_TRANSPORT_H_
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