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/*
* Copyright (c) 2015 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_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
#define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
#include <list>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/pacing/include/paced_sender.h"
namespace webrtc {
class CriticalSectionWrapper;
class RTPFragmentationHeader;
class RtpRtcp;
struct RTPVideoHeader;
// PacketRouter routes outgoing data to the correct sending RTP module, based
// on the simulcast layer in RTPVideoHeader.
class PacketRouter : public PacedSender::Callback {
public:
PacketRouter();
virtual ~PacketRouter();
void AddRtpModule(RtpRtcp* rtp_module);
void RemoveRtpModule(RtpRtcp* rtp_module);
// Implements PacedSender::Callback.
bool TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission) override;
size_t TimeToSendPadding(size_t bytes) override;
private:
// TODO(holmer): When the new video API has launched, remove crit_ and
// assume rtp_modules_ will never change during a call. We should then also
// switch rtp_modules_ to a map from ssrc to rtp module.
rtc::scoped_ptr<CriticalSectionWrapper> crit_;
// Map from ssrc to sending rtp module.
std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
DISALLOW_COPY_AND_ASSIGN(PacketRouter);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
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