rtp_sender_interface.h

Enable keyboard shortcuts

/*

 *  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 RtpSenders

// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface

#ifndef API_RTP_SENDER_INTERFACE_H_

#define API_RTP_SENDER_INTERFACE_H_

#include <cstdint>

#include <memory>

#include <string>

#include <utility>

#include <vector>

#include "absl/functional/any_invocable.h"

#include "api/crypto/frame_encryptor_interface.h"

#include "api/dtls_transport_interface.h"

#include "api/dtmf_sender_interface.h"

#include "api/frame_transformer_interface.h"

#include "api/media_stream_interface.h"

#include "api/media_types.h"

#include "api/ref_count.h"

#include "api/rtc_error.h"

#include "api/rtp_parameters.h"

#include "api/scoped_refptr.h"

#include "api/video_codecs/video_encoder_factory.h"

#include "rtc_base/system/rtc_export.h"

#include "api/rtp_sender_setparameters_callback.h"

namespace webrtc {

class RTC_EXPORT RtpSenderInterface : public webrtc::RefCountInterface,

                                      public FrameTransformerHost {

 public:

  // Returns true if successful in setting the track.

  // Fails if an audio track is set on a video RtpSender, or vice-versa.

  virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;

  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;

  // The dtlsTransport attribute exposes the DTLS transport on which the

  // media is sent. It may be null.

  // https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport

  virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;

  // Returns primary SSRC used by this sender for sending media.

  // Returns 0 if not yet determined.

  // TODO(deadbeef): Change to absl::optional.

  // TODO(deadbeef): Remove? With GetParameters this should be redundant.

  virtual uint32_t ssrc() const = 0;

  // Audio or video sender?

  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;

  // Returns a list of media stream ids associated with this sender's track.

  // These are signalled in the SDP so that the remote side can associate

  // tracks.

  virtual std::vector<std::string> stream_ids() const = 0;

  // Sets the IDs of the media streams associated with this sender's track.

  // These are signalled in the SDP so that the remote side can associate

  // tracks.

  virtual void SetStreams(const std::vector<std::string>& stream_ids) = 0;

  // Returns the list of encoding parameters that will be applied when the SDP

  // local description is set. These initial encoding parameters can be set by

  // PeerConnection::AddTransceiver, and later updated with Get/SetParameters.

  // TODO(orphis): Make it pure virtual once Chrome has updated

  virtual std::vector<RtpEncodingParameters> init_send_encodings() const = 0;

  virtual RtpParameters GetParameters() const = 0;

  // Note that only a subset of the parameters can currently be changed. See

  // rtpparameters.h

  // The encodings are in increasing quality order for simulcast.

  virtual RTCError SetParameters(const RtpParameters& parameters) = 0;

  virtual void SetParametersAsync(const RtpParameters& parameters,

                                  SetParametersCallback callback);

  // Returns null for a video sender.

  virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;

  // Sets a user defined frame encryptor that will encrypt the entire frame

  // before it is sent across the network. This will encrypt the entire frame

  // using the user provided encryption mechanism regardless of whether SRTP is

  // enabled or not.

  virtual void SetFrameEncryptor(

      rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor) = 0;

  // Returns a pointer to the frame encryptor set previously by the

  // user. This can be used to update the state of the object.

  virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor()

      const = 0;

  // TODO: bugs.webrtc.org/15929 - add [[deprecated("Use SetFrameTransformer")]]

  // when usage in Chrome is removed

  virtual void SetEncoderToPacketizerFrameTransformer(

      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {

    SetFrameTransformer(std::move(frame_transformer));

  // Sets a user defined encoder selector.

  // Overrides selector that is (optionally) provided by VideoEncoderFactory.

  virtual void SetEncoderSelector(

      std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>

          encoder_selector) = 0;

  // Default implementation of SetFrameTransformer.

  // TODO: bugs.webrtc.org/15929 - remove when all implementations are good

  void SetFrameTransformer(rtc::scoped_refptr<FrameTransformerInterface>

                               frame_transformer) override {}

 protected:

  ~RtpSenderInterface() override = default;

};

}  // namespace webrtc

#endif  // API_RTP_SENDER_INTERFACE_H_