audio_rtp_receiver.h

/*

 *  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/dtls_transport_interface.h"

#include "api/frame_transformer_interface.h"

#include "api/media_stream_interface.h"

#include "api/media_types.h"

#include "api/rtp_parameters.h"

#include "api/rtp_receiver_interface.h"

#include "api/scoped_refptr.h"

#include "api/sequence_checker.h"

#include "api/task_queue/pending_task_safety_flag.h"

#include "api/transport/rtp/rtp_source.h"

#include "media/base/media_channel.h"

#include "pc/audio_track.h"

#include "pc/jitter_buffer_delay.h"

#include "pc/media_stream_track_proxy.h"

#include "pc/remote_audio_source.h"

#include "pc/rtp_receiver.h"

#include "rtc_base/system/no_unique_address.h"

#include "rtc_base/thread.h"

#include "rtc_base/thread_annotations.h"

namespace webrtc {

class AudioRtpReceiver : public ObserverInterface,

                         public AudioSourceInterface::AudioObserver,

                         public RtpReceiverInternal {

 public:

  // The constructor supports optionally passing the voice channel to the

  // instance at construction time without having to call `SetMediaChannel()`

  // on the worker thread straight after construction.

  // However, when using that, the assumption is that right after construction,

  // a call to either `SetupUnsignaledMediaChannel` or `SetupMediaChannel`

  // will be made, which will internally start the source on the worker thread.

  AudioRtpReceiver(

      rtc::Thread* worker_thread,

      std::string receiver_id,

      std::vector<std::string> stream_ids,

      bool is_unified_plan,

      cricket::VoiceMediaReceiveChannelInterface* voice_channel = nullptr);

  // 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,

      bool is_unified_plan,

      cricket::VoiceMediaReceiveChannelInterface* media_channel = nullptr);

  virtual ~AudioRtpReceiver();

  // ObserverInterface implementation

  void OnChanged() override;

  // AudioSourceInterface::AudioObserver implementation

  void OnSetVolume(double volume) override;

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

  // RtpReceiverInterface implementation

  rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {

    return track_;

  rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const override;

  std::vector<std::string> stream_ids() const override;

  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()

      const override;

  cricket::MediaType media_type() const override {

    return cricket::MEDIA_TYPE_AUDIO;

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

  RtpParameters GetParameters() const 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;

  void SetupUnsignaledMediaChannel() override;

  absl::optional<uint32_t> ssrc() const override;

  void NotifyFirstPacketReceived() override;

  void set_stream_ids(std::vector<std::string> stream_ids) override;

  void set_transport(

      rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) override;

  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::MediaReceiveChannelInterface* media_channel) override;

  std::vector<RtpSource> GetSources() const override;

  int AttachmentId() const override { return attachment_id_; }

  void SetFrameTransformer(

      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) override;

 private:

  void RestartMediaChannel(absl::optional<uint32_t> ssrc)

      RTC_RUN_ON(&signaling_thread_checker_);

  void RestartMediaChannel_w(absl::optional<uint32_t> ssrc,

                             bool track_enabled,

                             MediaSourceInterface::SourceState state)

      RTC_RUN_ON(worker_thread_);

  void Reconfigure(bool track_enabled) RTC_RUN_ON(worker_thread_);

  void SetOutputVolume_w(double volume) RTC_RUN_ON(worker_thread_);

  RTC_NO_UNIQUE_ADDRESS SequenceChecker signaling_thread_checker_;

  rtc::Thread* const worker_thread_;

  const std::string id_;

  const rtc::scoped_refptr<RemoteAudioSource> source_;

  const rtc::scoped_refptr<AudioTrackProxyWithInternal<AudioTrack>> track_;

  cricket::VoiceMediaReceiveChannelInterface* media_channel_

      RTC_GUARDED_BY(worker_thread_) = nullptr;

  absl::optional<uint32_t> signaled_ssrc_ RTC_GUARDED_BY(worker_thread_);

  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_

      RTC_GUARDED_BY(&signaling_thread_checker_);

  bool cached_track_enabled_ RTC_GUARDED_BY(&signaling_thread_checker_);

  double cached_volume_ RTC_GUARDED_BY(worker_thread_) = 1.0;

  RtpReceiverObserverInterface* observer_

      RTC_GUARDED_BY(&signaling_thread_checker_) = nullptr;

  bool received_first_packet_ RTC_GUARDED_BY(&signaling_thread_checker_) =

      false;

  const int attachment_id_;

  rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor_

      RTC_GUARDED_BY(worker_thread_);

  rtc::scoped_refptr<DtlsTransportInterface> dtls_transport_

      RTC_GUARDED_BY(&signaling_thread_checker_);

  // Stores and updates the playout delay. Handles caching cases if

  // `SetJitterBufferMinimumDelay` is called before start.

  JitterBufferDelay delay_ RTC_GUARDED_BY(worker_thread_);

  rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_

      RTC_GUARDED_BY(worker_thread_);

  const rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_;

};

}  // namespace webrtc

#endif  // PC_AUDIO_RTP_RECEIVER_H_