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/*
* 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/thread_checker.h"
#include "webrtc/voice_engine/channel_manager.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
#include <string>
#include <vector>
namespace webrtc {
class AudioSinkInterface;
class PacketRouter;
class RtpPacketSender;
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 EnableSendTransportSequenceNumber(int id);
virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
virtual void SetCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,
PacketRouter* packet_router);
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(uint8_t event, uint32_t duration_ms);
virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
private:
Channel* channel() const;
rtc::ThreadChecker thread_checker_;
ChannelOwner channel_owner_;
};
} // namespace voe
} // namespace webrtc
#endif // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_
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