audio/channel_receive_proxy.h

/*
 *  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 AUDIO_CHANNEL_RECEIVE_PROXY_H_
#define AUDIO_CHANNEL_RECEIVE_PROXY_H_

#include <map>
#include <memory>
#include <vector>

#include "api/audio/audio_mixer.h"
#include "api/rtpreceiverinterface.h"
#include "audio/channel.h"
#include "call/rtp_packet_sink_interface.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/thread_checker.h"

namespace webrtc {

class AudioSinkInterface;
class PacketRouter;
class RtpPacketReceived;
class Transport;

namespace voe {

class ChannelSendProxy;

// This class provides the "view" of a voe::Channel that we need to implement
// 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 ChannelReceiveProxy : public RtpPacketSinkInterface {
 public:
  ChannelReceiveProxy();
  explicit ChannelReceiveProxy(std::unique_ptr<Channel> channel);
  virtual ~ChannelReceiveProxy();

  // Shared with ChannelSendProxy
  virtual void SetLocalSSRC(uint32_t ssrc);
  virtual void SetNACKStatus(bool enable, int max_packets);
  virtual CallStatistics GetRTCPStatistics() const;
  virtual void RegisterTransport(Transport* transport);
  virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);

  virtual void RegisterReceiverCongestionControlObjects(
      PacketRouter* packet_router);
  virtual void ResetReceiverCongestionControlObjects();
  virtual NetworkStatistics GetNetworkStatistics() const;
  virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
  virtual int GetSpeechOutputLevelFullRange() const;
  // See description of "totalAudioEnergy" in the WebRTC stats spec:
  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
  virtual double GetTotalOutputEnergy() const;
  virtual double GetTotalOutputDuration() const;
  virtual uint32_t GetDelayEstimate() const;
  virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
  virtual void SetSink(AudioSinkInterface* sink);

  // Implements RtpPacketSinkInterface
  void OnRtpPacket(const RtpPacketReceived& packet) override;

  virtual void SetChannelOutputVolumeScaling(float scaling);
  virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
      int sample_rate_hz,
      AudioFrame* audio_frame);
  virtual int PreferredSampleRate() const;
  virtual void AssociateSendChannel(const ChannelSendProxy& send_channel_proxy);
  virtual void DisassociateSendChannel();

  // Produces the transport-related timestamps; current_delay_ms is left unset.
  absl::optional<Syncable::Info> GetSyncInfo() const;
  virtual uint32_t GetPlayoutTimestamp() const;
  virtual void SetMinimumPlayoutDelay(int delay_ms);
  virtual bool GetRecCodec(CodecInst* codec_inst) const;
  virtual std::vector<webrtc::RtpSource> GetSources() const;
  virtual void StartPlayout();
  virtual void StopPlayout();

 private:
  // Thread checkers document and lock usage of some methods on voe::Channel to
  // specific threads we know about. The goal is to eventually split up
  // voe::Channel into parts with single-threaded semantics, and thereby reduce
  // the need for locks.
  rtc::ThreadChecker worker_thread_checker_;
  rtc::ThreadChecker module_process_thread_checker_;
  // Methods accessed from audio and video threads are checked for sequential-
  // only access. We don't necessarily own and control these threads, so thread
  // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
  // audio thread to another, but access is still sequential.
  rtc::RaceChecker audio_thread_race_checker_;
  rtc::RaceChecker video_capture_thread_race_checker_;
  std::unique_ptr<Channel> channel_;

  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelReceiveProxy);
};
}  // namespace voe
}  // namespace webrtc

#endif  // AUDIO_CHANNEL_RECEIVE_PROXY_H_