text: /*
   *  Copyright 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.
   */
  
  // This file contains interfaces for RtpReceivers
  // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
  
  #ifndef API_RTP_RECEIVER_INTERFACE_H_
  #define API_RTP_RECEIVER_INTERFACE_H_
  
  #include <string>
  #include <vector>
  
  #include "api/crypto/frame_decryptor_interface.h"
  #include "api/dtls_transport_interface.h"
  #include "api/media_stream_interface.h"
  #include "api/media_types.h"
  #include "api/proxy.h"
  #include "api/rtp_parameters.h"
  #include "api/scoped_refptr.h"
  #include "rtc_base/ref_count.h"
  
  namespace webrtc {
  
  enum class RtpSourceType {
    SSRC,
    CSRC,
  };
  
  class RtpSource {
   public:
    RtpSource() = delete;
    RtpSource(int64_t timestamp_ms,
              uint32_t source_id,
              RtpSourceType source_type);
    RtpSource(int64_t timestamp_ms,
              uint32_t source_id,
              RtpSourceType source_type,
              uint8_t audio_level);
    RtpSource(const RtpSource&);
    RtpSource& operator=(const RtpSource&);
    ~RtpSource();
  
    int64_t timestamp_ms() const { return timestamp_ms_; }
    void update_timestamp_ms(int64_t timestamp_ms) {
      RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
      timestamp_ms_ = timestamp_ms;
    }
  
    // The identifier of the source can be the CSRC or the SSRC.
    uint32_t source_id() const { return source_id_; }
  
    // The source can be either a contributing source or a synchronization source.
    RtpSourceType source_type() const { return source_type_; }
  
    absl::optional<uint8_t> audio_level() const { return audio_level_; }
    void set_audio_level(const absl::optional<uint8_t>& level) {
      audio_level_ = level;
    }
  
    bool operator==(const RtpSource& o) const {
      return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
             source_type_ == o.source_type() && audio_level_ == o.audio_level_;
    }
  
   private:
    int64_t timestamp_ms_;
    uint32_t source_id_;
    RtpSourceType source_type_;
    absl::optional<uint8_t> audio_level_;
  };
  
  class RtpReceiverObserverInterface {
   public:
    // Note: Currently if there are multiple RtpReceivers of the same media type,
    // they will all call OnFirstPacketReceived at once.
    //
    // In the future, it's likely that an RtpReceiver will only call
    // OnFirstPacketReceived when a packet is received specifically for its
    // SSRC/mid.
    virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
  
   protected:
    virtual ~RtpReceiverObserverInterface() {}
  };
  
  class RtpReceiverInterface : public rtc::RefCountInterface {
   public:
    virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
  
    // The dtlsTransport attribute exposes the DTLS transport on which the
    // media is received. It may be null.
    // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport
    // TODO(https://bugs.webrtc.org/907849) remove default implementation
    virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
  
    // The list of streams that |track| is associated with. This is the same as
    // the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
    // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
    // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
    // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of
    // stream_ids() as soon as downstream projects are no longer dependent on
    // stream objects.
    virtual std::vector<std::string> stream_ids() const;
    virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
  
    // Audio or video receiver?
    virtual cricket::MediaType media_type() const = 0;
  
    // Not to be confused with "mid", this is a field we can temporarily use
    // to uniquely identify a receiver until we implement Unified Plan SDP.
    virtual std::string id() const = 0;
  
    // The WebRTC specification only defines RTCRtpParameters in terms of senders,
    // but this API also applies them to receivers, similar to ORTC:
    // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
    virtual RtpParameters GetParameters() const = 0;
    // Currently, doesn't support changing any parameters, but may in the future.
    virtual bool SetParameters(const RtpParameters& parameters) = 0;
  
    // Does not take ownership of observer.
    // Must call SetObserver(nullptr) before the observer is destroyed.
    virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
  
    // Sets the jitter buffer minimum delay until media playout. Actual observed
    // delay may differ depending on the congestion control. |delay_seconds| is a
    // positive value including 0.0 measured in seconds. |nullopt| means default
    // value must be used. TODO(kuddai): remove the default implmenetation once
    // the subclasses in Chromium implement this.
    virtual void SetJitterBufferMinimumDelay(
        absl::optional<double> delay_seconds);
  
    // TODO(zhihuang): Remove the default implementation once the subclasses
    // implement this. Currently, the only relevant subclass is the
    // content::FakeRtpReceiver in Chromium.
    virtual std::vector<RtpSource> GetSources() const;
  
    // Sets a user defined frame decryptor that will decrypt the entire frame
    // before it is sent across the network. This will decrypt the entire frame
    // using the user provided decryption mechanism regardless of whether SRTP is
    // enabled or not.
    virtual void SetFrameDecryptor(
        rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
  
    // Returns a pointer to the frame decryptor set previously by the
    // user. This can be used to update the state of the object.
    virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;
  
   protected:
    ~RtpReceiverInterface() override = default;
  };
  
  // Define proxy for RtpReceiverInterface.
  // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
  // are called on is an implementation detail.
  BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
  PROXY_SIGNALING_THREAD_DESTRUCTOR()
  PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
  PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
  PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
                     streams)
  PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
  PROXY_CONSTMETHOD0(std::string, id)
  PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
  PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
  PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
  PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>)
  PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
  PROXY_METHOD1(void,
                SetFrameDecryptor,
                rtc::scoped_refptr<FrameDecryptorInterface>)
  PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
                     GetFrameDecryptor)
  END_PROXY_MAP()
  
  }  // namespace webrtc
  
  #endif  // API_RTP_RECEIVER_INTERFACE_H_
  