/*
 *  Copyright 2019 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 PC_AUDIO_RTP_RECEIVER_H_
#define PC_AUDIO_RTP_RECEIVER_H_

#include <stdint.h>
#include <string>
#include <vector>

#include "absl/types/optional.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "media/base/media_channel.h"
#include "pc/jitter_buffer_delay_interface.h"
#include "pc/remote_audio_source.h"
#include "pc/rtp_receiver.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/thread.h"

namespace webrtc {

class AudioRtpReceiver : public ObserverInterface,
                         public AudioSourceInterface::AudioObserver,
                         public rtc::RefCountedObject<RtpReceiverInternal> {
 public:
  AudioRtpReceiver(rtc::Thread* worker_thread,
                   std::string receiver_id,
                   std::vector<std::string> stream_ids);
  // TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
  AudioRtpReceiver(
      rtc::Thread* worker_thread,
      const std::string& receiver_id,
      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
  virtual ~AudioRtpReceiver();

  // ObserverInterface implementation
  void OnChanged() override;

  // AudioSourceInterface::AudioObserver implementation
  void OnSetVolume(double volume) override;

  rtc::scoped_refptr<AudioTrackInterface> audio_track() const {
    return track_.get();
  }

  // RtpReceiverInterface implementation
  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
    return track_.get();
  }
  rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override {
    return dtls_transport_;
  }
  std::vector<std::string> stream_ids() const override;
  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
      const override {
    return streams_;
  }

  cricket::MediaType media_type() const override {
    return cricket::MEDIA_TYPE_AUDIO;
  }

  std::string id() const override { return id_; }

  RtpParameters GetParameters() const override;
  bool SetParameters(const RtpParameters& parameters) override;

  void SetFrameDecryptor(
      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) override;

  rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor()
      const override;

  // RtpReceiverInternal implementation.
  void Stop() override;
  void SetupMediaChannel(uint32_t ssrc) override;
  uint32_t ssrc() const override { return ssrc_.value_or(0); }
  void NotifyFirstPacketReceived() override;
  void set_stream_ids(std::vector<std::string> stream_ids) override;
  void set_transport(
      rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override {
    dtls_transport_ = dtls_transport;
  }
  void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
                      streams) override;
  void SetObserver(RtpReceiverObserverInterface* observer) override;

  void SetJitterBufferMinimumDelay(
      absl::optional<double> delay_seconds) override;

  void SetMediaChannel(cricket::MediaChannel* media_channel) override;

  std::vector<RtpSource> GetSources() const override;
  int AttachmentId() const override { return attachment_id_; }

 private:
  void Reconfigure();
  bool SetOutputVolume(double volume);

  rtc::Thread* const worker_thread_;
  const std::string id_;
  const rtc::scoped_refptr<RemoteAudioSource> source_;
  const rtc::scoped_refptr<AudioTrackInterface> track_;
  cricket::VoiceMediaChannel* media_channel_ = nullptr;
  absl::optional<uint32_t> ssrc_;
  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
  bool cached_track_enabled_;
  double cached_volume_ = 1;
  bool stopped_ = false;
  RtpReceiverObserverInterface* observer_ = nullptr;
  bool received_first_packet_ = false;
  int attachment_id_ = 0;
  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_;
  rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_;
  // Allows to thread safely change playout delay. Handles caching cases if
  // |SetJitterBufferMinimumDelay| is called before start.
  rtc::scoped_refptr<JitterBufferDelayInterface> delay_;
};

}  // namespace webrtc

#endif  // PC_AUDIO_RTP_RECEIVER_H_