149 lines
6.3 KiB
C++
149 lines
6.3 KiB
C++
/*
|
|
* 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/proxy.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.
|
|
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.
|
|
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;
|
|
};
|
|
|
|
// Define proxy for RtpReceiverInterface.
|
|
// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
|
|
// are called on is an implementation detail.
|
|
BEGIN_PRIMARY_PROXY_MAP(RtpReceiver)
|
|
PROXY_PRIMARY_THREAD_DESTRUCTOR()
|
|
PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
|
|
PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
|
|
PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
|
|
PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
|
|
streams)
|
|
BYPASS_PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
|
|
BYPASS_PROXY_CONSTMETHOD0(std::string, id)
|
|
PROXY_CONSTMETHOD0(RtpParameters, GetParameters)
|
|
PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*)
|
|
PROXY_METHOD1(void, SetJitterBufferMinimumDelay, absl::optional<double>)
|
|
PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources)
|
|
PROXY_METHOD1(void,
|
|
SetFrameDecryptor,
|
|
rtc::scoped_refptr<FrameDecryptorInterface>)
|
|
PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameDecryptorInterface>,
|
|
GetFrameDecryptor)
|
|
PROXY_METHOD1(void,
|
|
SetDepacketizerToDecoderFrameTransformer,
|
|
rtc::scoped_refptr<FrameTransformerInterface>)
|
|
END_PROXY_MAP()
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_RTP_RECEIVER_INTERFACE_H_
|