50e28166af
This cl just moves the logic form the default module SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch in size between trate the vector and rtp modules. This was the same in the default module and is quite hard to protect from before we have the new video API. I also removed some test form rtp_rtcp_impl_unittest that were affected by this change. The test tests code that isn't implemented, hence the DISABLED_, and this will never be implemented in the RTP module, rather the payload router in the future. BUG=769 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42419004 Cr-Commit-Position: refs/heads/master@{#8453} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
96 lines
3.2 KiB
C++
96 lines
3.2 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_ENGINE_PAYLOAD_ROUTER_H_
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#define WEBRTC_VIDEO_ENGINE_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/thread_annotations.h"
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#include "webrtc/common_types.h"
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#include "webrtc/system_wrappers/interface/atomic32.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.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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// Called when it's time to send a stored packet.
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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);
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// Called when it's time to send padding, returns the number of bytes actually
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// sent.
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size_t TimeToSendPadding(size_t bytes);
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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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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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DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_PAYLOAD_ROUTER_H_
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