14d5dbe5b3
The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon. Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/. TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org BUG=webrtc:4741, webrtc:5603, chromium:609749 Review-Url: https://codereview.webrtc.org/2110113003 Cr-Commit-Position: refs/heads/master@{#13379}
105 lines
3.7 KiB
C++
105 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 RtcEventLog;
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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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virtual void SetRtcEventLog(RtcEventLog* event_log);
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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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