9e03c3b372
Reason for revert: Reverting all CLs related to moving the eventlog, as they break Chromium tests. Original issue's description: > Move RtcEventLog object from inside VoiceEngine to Call. > > In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced. > The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface. > > BUG=webrtc:4741,webrtc:5603,chromium:609749 > R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org > > Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016 > Cr-Commit-Position: refs/heads/master@{#13321} TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741,webrtc:5603,chromium:609749 Review-Url: https://codereview.webrtc.org/2111813002 Cr-Commit-Position: refs/heads/master@{#13340}
102 lines
3.7 KiB
C++
102 lines
3.7 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_VOICE_ENGINE_CHANNEL_PROXY_H_
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#define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/voice_engine/channel_manager.h"
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#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
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#include <memory>
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#include <string>
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#include <vector>
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namespace webrtc {
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class AudioSinkInterface;
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class PacketRouter;
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class RtpPacketSender;
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class Transport;
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class TransportFeedbackObserver;
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namespace voe {
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class Channel;
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// This class provides the "view" of a voe::Channel that we need to implement
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// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
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// purposes:
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// 1. Allow mocking just the interfaces used, instead of the entire
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// voe::Channel class.
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// 2. Provide a refined interface for the stream classes, including assumptions
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// on return values and input adaptation.
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class ChannelProxy {
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public:
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ChannelProxy();
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explicit ChannelProxy(const ChannelOwner& channel_owner);
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virtual ~ChannelProxy();
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virtual void SetRTCPStatus(bool enable);
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virtual void SetLocalSSRC(uint32_t ssrc);
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virtual void SetRTCP_CNAME(const std::string& c_name);
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virtual void SetNACKStatus(bool enable, int max_packets);
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virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
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virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
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virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
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virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
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virtual void EnableSendTransportSequenceNumber(int id);
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virtual void EnableReceiveTransportSequenceNumber(int id);
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virtual void RegisterSenderCongestionControlObjects(
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RtpPacketSender* rtp_packet_sender,
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TransportFeedbackObserver* transport_feedback_observer,
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PacketRouter* packet_router);
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virtual void RegisterReceiverCongestionControlObjects(
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PacketRouter* packet_router);
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virtual void ResetCongestionControlObjects();
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virtual CallStatistics GetRTCPStatistics() const;
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virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
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virtual NetworkStatistics GetNetworkStatistics() const;
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virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
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virtual int32_t GetSpeechOutputLevelFullRange() const;
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virtual uint32_t GetDelayEstimate() const;
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virtual bool SetSendTelephoneEventPayloadType(int payload_type);
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virtual bool SendTelephoneEventOutband(int event, int duration_ms);
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virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
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virtual void SetInputMute(bool muted);
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virtual void RegisterExternalTransport(Transport* transport);
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virtual void DeRegisterExternalTransport();
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virtual bool ReceivedRTPPacket(const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time);
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virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
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virtual const rtc::scoped_refptr<AudioDecoderFactory>&
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GetAudioDecoderFactory() const;
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virtual void SetChannelOutputVolumeScaling(float scaling);
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private:
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Channel* channel() const;
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rtc::ThreadChecker thread_checker_;
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ChannelOwner channel_owner_;
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RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
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};
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} // namespace voe
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
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