60 lines
1.9 KiB
C
60 lines
1.9 KiB
C
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
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#define WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
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#include <list>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/pacing/include/paced_sender.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RTPFragmentationHeader;
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class RtpRtcp;
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struct RTPVideoHeader;
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// PacketRouter routes outgoing data to the correct sending RTP module, based
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// on the simulcast layer in RTPVideoHeader.
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class PacketRouter : public PacedSender::Callback {
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public:
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PacketRouter();
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virtual ~PacketRouter();
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void AddRtpModule(RtpRtcp* rtp_module);
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void RemoveRtpModule(RtpRtcp* rtp_module);
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// Implements PacedSender::Callback.
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bool TimeToSendPacket(uint32_t ssrc,
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uint16_t sequence_number,
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int64_t capture_timestamp,
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bool retransmission) override;
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size_t TimeToSendPadding(size_t bytes) override;
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private:
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// TODO(holmer): When the new video API has launched, remove crit_ and
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// assume rtp_modules_ will never change during a call. We should then also
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// switch rtp_modules_ to a map from ssrc to rtp module.
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rtc::scoped_ptr<CriticalSectionWrapper> crit_;
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// Map from ssrc to sending rtp module.
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std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
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DISALLOW_COPY_AND_ASSIGN(PacketRouter);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_PACING_INCLUDE_PACKET_ROUTER_H_
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