20a4b3fb2a
These are the changes made to WebRtcVoiceEngine and surrounding code. It still contains some things that are inelegant, like how AudioCodecSpec and AudioFormatInfo is ferried around in SendCodecSpec. This should probably be resolved before landing. There are also a few test still that are disabled. They should be removed or fixed, as the case may be. I've put this CL up to get a better overview of the changes made and how reviewable they are. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2705093002 Cr-Commit-Position: refs/heads/master@{#17904}
145 lines
6.0 KiB
C++
145 lines
6.0 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/rtpreceiverinterface.h"
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/race_checker.h"
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#include "webrtc/base/thread_checker.h"
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#include "webrtc/modules/audio_coding/codecs/audio_encoder.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 {
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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 int GetSpeechOutputLevel() const;
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virtual int 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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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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virtual void OnRtpPacket(const RtpPacketReceived& packet);
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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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