3c12f4dadb
Reason for revert: Caused static initializers. BUG=chromium:556866 TBR=tommi@webrtc.org Original issue's description: > Create rtc::AtomicInt POD struct. > > Prevents accidental non-atomic reads, increments and stores since > "volatile int" doesn't enforce atomic usage. > > BUG= > R=kwiberg@webrtc.org, tommi@webrtc.org > > Committed: https://crrev.com/b27f590ece487819c3d1fda400315e582fb975b6 > Cr-Commit-Position: refs/heads/master@{#10657} TBR=kwiberg@webrtc.org,tommi@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG= Review URL: https://codereview.webrtc.org/1453093002 Cr-Commit-Position: refs/heads/master@{#10669}
67 lines
2.1 KiB
C++
67 lines
2.1 KiB
C++
/*
|
|
* 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_PACKET_ROUTER_H_
|
|
#define WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
|
|
|
|
#include <list>
|
|
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/base/criticalsection.h"
|
|
#include "webrtc/base/scoped_ptr.h"
|
|
#include "webrtc/base/thread_annotations.h"
|
|
#include "webrtc/common_types.h"
|
|
#include "webrtc/modules/pacing/paced_sender.h"
|
|
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RtpRtcp;
|
|
namespace rtcp {
|
|
class TransportFeedback;
|
|
} // namespace rtcp
|
|
|
|
// PacketRouter routes outgoing data to the correct sending RTP module, based
|
|
// on the simulcast layer in RTPVideoHeader.
|
|
class PacketRouter : public PacedSender::Callback,
|
|
public TransportSequenceNumberAllocator {
|
|
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;
|
|
|
|
void SetTransportWideSequenceNumber(uint16_t sequence_number);
|
|
uint16_t AllocateSequenceNumber() override;
|
|
|
|
// Send transport feedback packet to send-side.
|
|
virtual bool SendFeedback(rtcp::TransportFeedback* packet);
|
|
|
|
private:
|
|
rtc::CriticalSection modules_lock_;
|
|
// Map from ssrc to sending rtp module.
|
|
std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_lock_);
|
|
|
|
volatile int transport_seq_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
|