7623ce4aeb
Reason for revert: Bot breakage caused by TickTime::UseFakeClock has been removed. Original issue's description: > Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ ) > > Reason for revert: > Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots. > > Original issue's description: > > Merge webrtc/video_engine/ into webrtc/video/ > > > > BUG=webrtc:1695 > > R=mflodman@webrtc.org > > > > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646 > > Cr-Commit-Position: refs/heads/master@{#10926} > > TBR=mflodman@webrtc.org,pbos@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:1695 > > Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518 > Cr-Commit-Position: refs/heads/master@{#10937} BUG=webrtc:1695 TBR=mflodman@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1510183002 . Cr-Commit-Position: refs/heads/master@{#10948}
86 lines
2.8 KiB
C++
86 lines
2.8 KiB
C++
/*
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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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