28a44564c9
This reverts commit 825eb58d59940a4c3c9837595c4b3b07059c93ca. This Relands the cl reviewed in https://codereview.webrtc.org/1917793002/ patchset #1 is a pure reland. patchset #2 fix an overflow in BitrateProber that caused WebRtcVideoChannel2BaseTest.TwoStreamsSendAndReceive to fail. Original cl description: Remove SendPacer from ViEEncoder This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator) R=stefan@webrtc.org TBR=mflodman@webrtc.org BUG=webrtc:5687 Review URL: https://codereview.webrtc.org/1947873002 . Cr-Commit-Position: refs/heads/master@{#12630}
67 lines
2.1 KiB
C++
67 lines
2.1 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_MODULES_PACING_PACKET_ROUTER_H_
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#define WEBRTC_MODULES_PACING_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/criticalsection.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/pacing/paced_sender.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class RtpRtcp;
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namespace rtcp {
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class TransportFeedback;
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} // namespace rtcp
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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::PacketSender,
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public TransportSequenceNumberAllocator {
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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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void SetTransportWideSequenceNumber(uint16_t sequence_number);
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uint16_t AllocateSequenceNumber() override;
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// Send transport feedback packet to send-side.
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virtual bool SendFeedback(rtcp::TransportFeedback* packet);
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private:
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rtc::ThreadChecker pacer_thread_checker_;
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rtc::CriticalSection modules_crit_;
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std::list<RtpRtcp*> rtp_modules_ GUARDED_BY(modules_crit_);
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volatile int transport_seq_;
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RTC_DISALLOW_COPY_AND_ASSIGN(PacketRouter);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_PACING_PACKET_ROUTER_H_
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