e1198e068b
This CL only wires up the new stats but does not set the values yet. This will be added in a follow-up CL. BUG=webrtc:8127 Review-Url: https://codereview.webrtc.org/3011623002 Cr-Commit-Position: refs/heads/master@{#19751}
153 lines
6.5 KiB
C++
153 lines
6.5 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/api/audio/audio_mixer.h"
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#include "webrtc/api/audio_codecs/audio_encoder.h"
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#include "webrtc/api/rtpreceiverinterface.h"
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#include "webrtc/call/rtp_packet_sink_interface.h"
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#include "webrtc/rtc_base/constructormagic.h"
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#include "webrtc/rtc_base/race_checker.h"
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#include "webrtc/rtc_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 RtcpBandwidthObserver;
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class RtcpRttStats;
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class RtpPacketSender;
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class RtpPacketReceived;
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class RtpReceiver;
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class RtpRtcp;
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class RtpTransportControllerSendInterface;
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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 : public RtpPacketSinkInterface {
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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 bool SetEncoder(int payload_type,
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std::unique_ptr<AudioEncoder> encoder);
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virtual void ModifyEncoder(
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rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier);
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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 SetSendAudioLevelIndicationStatus(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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RtpTransportControllerSendInterface* transport,
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RtcpBandwidthObserver* bandwidth_observer);
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virtual void RegisterReceiverCongestionControlObjects(
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PacketRouter* packet_router);
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virtual void ResetSenderCongestionControlObjects();
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virtual void ResetReceiverCongestionControlObjects();
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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 ANAStats GetANAStatistics() const;
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virtual int GetSpeechOutputLevel() const;
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virtual int GetSpeechOutputLevelFullRange() const;
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// See description of "totalAudioEnergy" in the WebRTC stats spec:
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// https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
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virtual double GetTotalOutputEnergy() const;
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virtual double GetTotalOutputDuration() const;
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virtual uint32_t GetDelayEstimate() const;
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virtual bool SetSendTelephoneEventPayloadType(int payload_type,
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int payload_frequency);
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virtual bool SendTelephoneEventOutband(int event, int duration_ms);
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virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
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virtual void SetRecPayloadType(int payload_type,
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const SdpAudioFormat& format);
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virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
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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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// Implements RtpPacketSinkInterface
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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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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virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
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int sample_rate_hz,
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AudioFrame* audio_frame);
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virtual int NeededFrequency() const;
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virtual void SetTransportOverhead(int transport_overhead_per_packet);
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virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
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virtual void DisassociateSendChannel();
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virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
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RtpReceiver** rtp_receiver) const;
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virtual uint32_t GetPlayoutTimestamp() const;
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virtual void SetMinimumPlayoutDelay(int delay_ms);
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virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
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virtual bool GetRecCodec(CodecInst* codec_inst) const;
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virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
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virtual void OnRecoverableUplinkPacketLossRate(
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float recoverable_packet_loss_rate);
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virtual void RegisterLegacyReceiveCodecs();
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virtual std::vector<webrtc::RtpSource> GetSources() const;
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private:
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Channel* channel() const;
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// Thread checkers document and lock usage of some methods on voe::Channel to
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// specific threads we know about. The goal is to eventually split up
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// voe::Channel into parts with single-threaded semantics, and thereby reduce
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// the need for locks.
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rtc::ThreadChecker worker_thread_checker_;
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rtc::ThreadChecker module_process_thread_checker_;
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// Methods accessed from audio and video threads are checked for sequential-
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// only access. We don't necessarily own and control these threads, so thread
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// checkers cannot be used. E.g. Chromium may transfer "ownership" from one
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// audio thread to another, but access is still sequential.
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rtc::RaceChecker audio_thread_race_checker_;
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rtc::RaceChecker video_capture_thread_race_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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