/*
 *  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_RTPRECEIVERINTERFACE_H_
#define API_RTPRECEIVERINTERFACE_H_

#include <string>
#include <vector>

#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/proxy.h"
#include "api/rtpparameters.h"
#include "rtc_base/refcount.h"
#include "rtc_base/scoped_ref_ptr.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)
      : timestamp_ms_(timestamp_ms),
        source_id_(source_id),
        source_type_(source_type) {}

  RtpSource(int64_t timestamp_ms,
            uint32_t source_id,
            RtpSourceType source_type,
            uint8_t audio_level)
      : timestamp_ms_(timestamp_ms),
        source_id_(source_id),
        source_type_(source_type),
        audio_level_(audio_level) {}

  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_; }

  rtc::Optional<uint8_t> audio_level() const { return audio_level_; }
  void set_audio_level(const rtc::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_;
  rtc::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 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-x%5B%5Bassociatedremotemediastreams%5D%5D
  // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
  virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
      const {
    return std::vector<rtc::scoped_refptr<MediaStreamInterface>>();
  }

  // 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;

  // 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 {
    return std::vector<RtpSource>();
  }
  // TODO(hta): Remove default implementation or move function to
  // an internal interface. content::FakeRtpReceiver in Chromium needs this.

  // Returns an ID that changes if the attached track changes, but
  // otherwise remains constant. Used to generate IDs for stats.
  // The special value zero means that no track is attached.
  virtual int AttachmentId() const { return 0; }

 protected:
  virtual ~RtpReceiverInterface() {}
};

// 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(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_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
  PROXY_CONSTMETHOD0(int, AttachmentId);
  END_PROXY_MAP()

}  // namespace webrtc

#endif  // API_RTPRECEIVERINTERFACE_H_