/*
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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_VIDEO_PAYLOAD_ROUTER_H_
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#define WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
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#include <list>
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#include <vector>
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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/system_wrappers/include/atomic32.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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// PayloadRouter 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 PayloadRouter {
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public:
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PayloadRouter();
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~PayloadRouter();
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static size_t DefaultMaxPayloadLength();
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// Rtp modules are assumed to be sorted in simulcast index order.
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void SetSendingRtpModules(const std::list<RtpRtcp*>& rtp_modules);
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// PayloadRouter will only route packets if being active, all packets will be
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// dropped otherwise.
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void set_active(bool active);
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bool active();
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// Input parameters according to the signature of RtpRtcp::SendOutgoingData.
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// Returns true if the packet was routed / sent, false otherwise.
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bool RoutePayload(FrameType frame_type,
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int8_t payload_type,
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uint32_t time_stamp,
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int64_t capture_time_ms,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation,
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const RTPVideoHeader* rtp_video_hdr);
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// Configures current target bitrate per module. 'stream_bitrates' is assumed
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// to be in the same order as 'SetSendingRtpModules'.
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void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
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// Returns the maximum allowed data payload length, given the configured MTU
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// and RTP headers.
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size_t MaxPayloadLength() const;
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void AddRef() { ++ref_count_; }
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void Release() { if (--ref_count_ == 0) { delete this; } }
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private:
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// TODO(mflodman): When the new video API has launched, remove crit_ and
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// assume rtp_modules_ will never change during a call.
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rtc::scoped_ptr<CriticalSectionWrapper> crit_;
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// Active sending RTP modules, in layer order.
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std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_.get());
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bool active_ GUARDED_BY(crit_.get());
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Atomic32 ref_count_;
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RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_
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