84583f6183
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender then each outgoing audio payload will first pass through the provided FrameEncryptor which will have a chance to modify the payload contents for the purposes of encryption. If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming audio payload will first pass through the provided FrameDecryptor which have a chance to modify the payload contents for the purpose of decryption. While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not use it and so it is left as null. Bug: webrtc:9681 Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf Reviewed-on: https://webrtc-review.googlesource.com/c/101702 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25001}
112 lines
4.4 KiB
C++
112 lines
4.4 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 AUDIO_CHANNEL_SEND_PROXY_H_
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#define AUDIO_CHANNEL_SEND_PROXY_H_
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#include <memory>
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#include <string>
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#include <vector>
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#include "api/audio_codecs/audio_encoder.h"
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#include "audio/channel_send.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/race_checker.h"
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#include "rtc_base/thread_checker.h"
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namespace webrtc {
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class FrameEncryptorInterface;
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class RtcpBandwidthObserver;
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class RtpRtcp;
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class RtpTransportControllerSendInterface;
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class Transport;
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namespace voe {
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// This class provides the "view" of a voe::Channel that we need to implement
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// webrtc::AudioSendStream. It serves two 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 ChannelSendProxy {
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public:
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ChannelSendProxy();
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explicit ChannelSendProxy(std::unique_ptr<ChannelSend> channel);
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virtual ~ChannelSendProxy();
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// Shared with ChannelReceiveProxy
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virtual void SetLocalSSRC(uint32_t ssrc);
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virtual void SetNACKStatus(bool enable, int max_packets);
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virtual CallSendStatistics GetRTCPStatistics() const;
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virtual void RegisterTransport(Transport* transport);
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virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
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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 SetMid(const std::string& mid, int extension_id);
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virtual void SetRTCP_CNAME(const std::string& c_name);
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virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
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virtual void EnableSendTransportSequenceNumber(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 ResetSenderCongestionControlObjects();
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virtual std::vector<ReportBlock> GetRemoteRTCPReportBlocks() const;
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virtual ANAStats GetANAStatistics() 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 SetInputMute(bool muted);
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virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame);
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virtual void SetTransportOverhead(int transport_overhead_per_packet);
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virtual RtpRtcp* GetRtpRtcp() 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 StartSend();
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virtual void StopSend();
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// Needed by ChannelReceiveProxy::AssociateSendChannel.
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virtual ChannelSend* GetChannel() const;
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// E2EE Custom Audio Frame Encryption (Optional)
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virtual void SetFrameEncryptor(FrameEncryptorInterface* frame_encryptor);
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private:
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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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std::unique_ptr<ChannelSend> channel_;
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RTC_DISALLOW_COPY_AND_ASSIGN(ChannelSendProxy);
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};
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} // namespace voe
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} // namespace webrtc
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#endif // AUDIO_CHANNEL_SEND_PROXY_H_
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