3d7db263b9
VoENetwork is kept for now, but is not really used anylonger. webrtcvoiceengine is changed to have the same behavior for unsignaled ssrc as video has, which is reflected by disabling one test case and this will be discussed and followed up. BUG=webrtc:5079 TBR=tommi Review-Url: https://codereview.webrtc.org/1909333002 Cr-Commit-Position: refs/heads/master@{#12555}
95 lines
3.4 KiB
C++
95 lines
3.4 KiB
C++
/*
|
|
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
|
|
#define WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
|
|
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/base/thread_checker.h"
|
|
#include "webrtc/voice_engine/channel_manager.h"
|
|
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
|
|
|
|
#include <memory>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
namespace webrtc {
|
|
|
|
class AudioSinkInterface;
|
|
class PacketRouter;
|
|
class RtpPacketSender;
|
|
class Transport;
|
|
class TransportFeedbackObserver;
|
|
|
|
namespace voe {
|
|
|
|
class Channel;
|
|
|
|
// This class provides the "view" of a voe::Channel that we need to implement
|
|
// webrtc::AudioSendStream and webrtc::AudioReceiveStream. It serves two
|
|
// purposes:
|
|
// 1. Allow mocking just the interfaces used, instead of the entire
|
|
// voe::Channel class.
|
|
// 2. Provide a refined interface for the stream classes, including assumptions
|
|
// on return values and input adaptation.
|
|
class ChannelProxy {
|
|
public:
|
|
ChannelProxy();
|
|
explicit ChannelProxy(const ChannelOwner& channel_owner);
|
|
virtual ~ChannelProxy();
|
|
|
|
virtual void SetRTCPStatus(bool enable);
|
|
virtual void SetLocalSSRC(uint32_t ssrc);
|
|
virtual void SetRTCP_CNAME(const std::string& c_name);
|
|
virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
|
|
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
|
|
virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
|
|
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
|
|
virtual void EnableSendTransportSequenceNumber(int id);
|
|
virtual void EnableReceiveTransportSequenceNumber(int id);
|
|
virtual void RegisterSenderCongestionControlObjects(
|
|
RtpPacketSender* rtp_packet_sender,
|
|
TransportFeedbackObserver* transport_feedback_observer,
|
|
PacketRouter* packet_router);
|
|
virtual void RegisterReceiverCongestionControlObjects(
|
|
PacketRouter* packet_router);
|
|
virtual void ResetCongestionControlObjects();
|
|
|
|
virtual CallStatistics GetRTCPStatistics() const;
|
|
virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
|
|
virtual NetworkStatistics GetNetworkStatistics() const;
|
|
virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
|
|
virtual int32_t GetSpeechOutputLevelFullRange() const;
|
|
virtual uint32_t GetDelayEstimate() const;
|
|
|
|
virtual bool SetSendTelephoneEventPayloadType(int payload_type);
|
|
virtual bool SendTelephoneEventOutband(int event, int duration_ms);
|
|
virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
|
|
|
|
virtual void RegisterExternalTransport(Transport* transport);
|
|
virtual void DeRegisterExternalTransport();
|
|
virtual bool ReceivedRTPPacket(const uint8_t* packet,
|
|
size_t length,
|
|
const PacketTime& packet_time);
|
|
virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
|
|
|
|
private:
|
|
Channel* channel() const;
|
|
|
|
rtc::ThreadChecker thread_checker_;
|
|
ChannelOwner channel_owner_;
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
|
|
};
|
|
} // namespace voe
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
|