blob: 0698992d5a871f48b266ae718cf11116af0fd3e2 [file] [log] [blame]
/*
* 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 WEBRTC_TEST_DIRECT_TRANSPORT_H_
#define WEBRTC_TEST_DIRECT_TRANSPORT_H_
#include <assert.h>
#include <deque>
#include "webrtc/api/call/transport.h"
#include "webrtc/call/call.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/event.h"
#include "webrtc/rtc_base/platform_thread.h"
#include "webrtc/test/fake_network_pipe.h"
namespace webrtc {
class Clock;
class PacketReceiver;
namespace test {
class DirectTransport : public Transport {
public:
DirectTransport(Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map);
DirectTransport(const FakeNetworkPipe::Config& config,
Call* send_call,
const std::map<uint8_t, MediaType>& payload_type_map);
DirectTransport(const FakeNetworkPipe::Config& config,
Call* send_call,
std::unique_ptr<Demuxer> demuxer);
// This deprecated variant always uses MediaType::VIDEO.
RTC_DEPRECATED explicit DirectTransport(Call* send_call)
: DirectTransport(
FakeNetworkPipe::Config(),
send_call,
std::unique_ptr<Demuxer>(new ForceDemuxer(MediaType::VIDEO))) {}
~DirectTransport() override;
void SetConfig(const FakeNetworkPipe::Config& config);
virtual 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:
// TODO(minyue): remove when the deprecated ctors of DirectTransport that
// create ForceDemuxer are removed.
class ForceDemuxer : public Demuxer {
public:
explicit ForceDemuxer(MediaType media_type);
void SetReceiver(PacketReceiver* receiver) override;
void DeliverPacket(const NetworkPacket* packet,
const PacketTime& packet_time) override;
private:
const MediaType media_type_;
PacketReceiver* packet_receiver_;
RTC_DISALLOW_COPY_AND_ASSIGN(ForceDemuxer);
};
static bool NetworkProcess(void* transport);
bool SendPackets();
rtc::CriticalSection lock_;
Call* const send_call_;
rtc::Event packet_event_;
rtc::PlatformThread thread_;
Clock* const clock_;
bool shutting_down_;
FakeNetworkPipe fake_network_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_TEST_DIRECT_TRANSPORT_H_