tor-browser

rtp_receiver_interface.h (5612B)

---

/*
*  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_RTP_RECEIVER_INTERFACE_H_
#define API_RTP_RECEIVER_INTERFACE_H_

#include <optional>
#include <string>
#include <utility>
#include <vector>

#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/ref_count.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "rtc_base/system/rtc_export.h"

namespace webrtc {

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(MediaType media_type) = 0;

protected:
 virtual ~RtpReceiverObserverInterface() {}
};

class RTC_EXPORT RtpReceiverInterface : public RefCountInterface,
                                       public FrameTransformerHost {
public:
 virtual scoped_refptr<MediaStreamTrackInterface> track() const = 0;

 // The dtlsTransport attribute exposes the DTLS transport on which the
 // media is received. It may be null.
 // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport
 // TODO(https://bugs.webrtc.org/907849) remove default implementation
 virtual scoped_refptr<DtlsTransportInterface> dtls_transport() const;

 // 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-associatedremotemediastreams
 // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
 // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of
 // stream_ids() as soon as downstream projects are no longer dependent on
 // stream objects.
 virtual std::vector<std::string> stream_ids() const;
 virtual std::vector<scoped_refptr<MediaStreamInterface>> streams() const;

 // Audio or video receiver?
 virtual 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;
 // TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium.
 // Currently, doesn't support changing any parameters.
 virtual bool SetParameters(const RtpParameters& /* parameters */) {
   return false;
 }

 // Does not take ownership of observer.
 // Must call SetObserver(nullptr) before the observer is destroyed.
 virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;

 // Sets the jitter buffer minimum delay until media playout. Actual observed
 // delay may differ depending on the congestion control. `delay_seconds` is a
 // positive value including 0.0 measured in seconds. `nullopt` means default
 // value must be used.
 virtual void SetJitterBufferMinimumDelay(
     std::optional<double> delay_seconds) = 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;

 // Sets a user defined frame decryptor that will decrypt the entire frame
 // before it is sent across the network. This will decrypt the entire frame
 // using the user provided decryption mechanism regardless of whether SRTP is
 // enabled or not.
 // TODO(bugs.webrtc.org/12772): Remove.
 virtual void SetFrameDecryptor(
     scoped_refptr<FrameDecryptorInterface> frame_decryptor);

 // Returns a pointer to the frame decryptor set previously by the
 // user. This can be used to update the state of the object.
 // TODO(bugs.webrtc.org/12772): Remove.
 virtual scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;

 // Sets a frame transformer between the depacketizer and the decoder to enable
 // client code to transform received frames according to their own processing
 // logic.
 [[deprecated("Use SetFrameTransformer")]] virtual void
 SetDepacketizerToDecoderFrameTransformer(
     scoped_refptr<FrameTransformerInterface> frame_transformer) {
   SetFrameTransformer(std::move(frame_transformer));
 }

 // Default implementation of SetFrameTransformer.
 // TODO: bugs.webrtc.org/15929 - Make pure virtual.
 void SetFrameTransformer(
     scoped_refptr<FrameTransformerInterface> frame_transformer) override;

protected:
 ~RtpReceiverInterface() override = default;
};

}  // namespace webrtc

#endif  // API_RTP_RECEIVER_INTERFACE_H_