ae4237e5db
Followup to cl https://webrtc-review.googlesource.com/c/src/+/103640. Set the rtcp_send_transport at construction time, delete RegisterTransport, and the proxying of transport methods. In addition, delete the unused RtcpRtpStats argument from the constructor. Bug: webrtc:9801 Change-Id: I80f25bc08dc2130386053568ddce4ef91654caeb Reviewed-on: https://webrtc-review.googlesource.com/c/103803 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25010}
110 lines
4.3 KiB
C++
110 lines
4.3 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_RECEIVE_PROXY_H_
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#define AUDIO_CHANNEL_RECEIVE_PROXY_H_
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#include <map>
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#include <memory>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "api/rtpreceiverinterface.h"
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#include "audio/channel_receive.h"
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#include "call/rtp_packet_sink_interface.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 AudioSinkInterface;
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class PacketRouter;
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class RtpPacketReceived;
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class Transport;
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namespace voe {
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class ChannelSendProxy;
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// This class provides the "view" of a voe::Channel that we need to implement
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// webrtc::AudioReceiveStream. 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 ChannelReceiveProxy : public RtpPacketSinkInterface {
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public:
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ChannelReceiveProxy();
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explicit ChannelReceiveProxy(std::unique_ptr<ChannelReceive> channel);
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virtual ~ChannelReceiveProxy();
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// Shared with ChannelSendProxy
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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 CallReceiveStatistics GetRTCPStatistics() const;
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virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
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virtual void RegisterReceiverCongestionControlObjects(
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PacketRouter* packet_router);
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virtual void ResetReceiverCongestionControlObjects();
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virtual NetworkStatistics GetNetworkStatistics() const;
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virtual AudioDecodingCallStats GetDecodingCallStatistics() 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 void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
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virtual void SetSink(AudioSinkInterface* sink);
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// Implements RtpPacketSinkInterface
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void OnRtpPacket(const RtpPacketReceived& packet) override;
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virtual void SetChannelOutputVolumeScaling(float scaling);
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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 PreferredSampleRate() const;
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virtual void AssociateSendChannel(const ChannelSendProxy& send_channel_proxy);
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virtual void DisassociateSendChannel();
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// Produces the transport-related timestamps; current_delay_ms is left unset.
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absl::optional<Syncable::Info> GetSyncInfo() 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 bool GetRecCodec(CodecInst* codec_inst) const;
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virtual std::vector<webrtc::RtpSource> GetSources() const;
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virtual void StartPlayout();
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virtual void StopPlayout();
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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<ChannelReceive> channel_;
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RTC_DISALLOW_COPY_AND_ASSIGN(ChannelReceiveProxy);
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};
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} // namespace voe
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} // namespace webrtc
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#endif // AUDIO_CHANNEL_RECEIVE_PROXY_H_
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