/*
* 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 RtpSenderObserverInterface {
public:
// The observer is called when the first media packet is sent for the observed
// sender. It is called immediately if the first packet was already sent.
virtual void OnFirstPacketSent(cricket::MediaType media_type) = 0;
protected:
virtual ~RtpSenderObserverInterface() {}
};
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 std::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);
// Sets an observer which gets a callback when the first media packet is sent
// for this sender.
// Does not take ownership of observer.
// Must call SetObserver(nullptr) before the observer is destroyed.
virtual void SetObserver(RtpSenderObserverInterface*
/* observer */) {}
// 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_
