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/*
* 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 <string>
#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/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/transport/rtp/rtp_source.h"
#include "rtc_base/ref_count.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(cricket::MediaType media_type) = 0;
protected:
virtual ~RtpReceiverObserverInterface() {}
};
class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
public:
virtual rtc::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 rtc::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<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
// Audio or video receiver?
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;
// 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(
absl::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(
rtc::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 rtc::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.
virtual void SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
protected:
~RtpReceiverInterface() override = default;
};
} // namespace webrtc
#endif // API_RTP_RECEIVER_INTERFACE_H_
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