2015-09-24 16:47:53 -07:00
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/*
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2016-02-10 07:54:43 -08:00
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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2015-09-24 16:47:53 -07:00
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*
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2016-02-10 07:54:43 -08:00
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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2015-09-24 16:47:53 -07:00
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*/
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2015-09-28 16:53:55 -07:00
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// This file contains interfaces for RtpSenders
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// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
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2019-01-11 09:11:00 -08:00
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#ifndef API_RTP_SENDER_INTERFACE_H_
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#define API_RTP_SENDER_INTERFACE_H_
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2015-09-28 16:53:55 -07:00
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#include <string>
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2016-06-06 14:27:39 -07:00
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#include <vector>
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2015-09-28 16:53:55 -07:00
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2019-01-11 09:11:00 -08:00
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#include "api/crypto/frame_encryptor_interface.h"
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#include "api/dtmf_sender_interface.h"
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#include "api/media_stream_interface.h"
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#include "api/media_types.h"
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2017-09-15 06:47:31 +02:00
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#include "api/proxy.h"
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#include "api/rtc_error.h"
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#include "api/rtp_parameters.h"
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#include "rtc_base/ref_count.h"
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2017-09-15 06:47:31 +02:00
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#include "rtc_base/scoped_ref_ptr.h"
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namespace webrtc {
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class RtpSenderInterface : public rtc::RefCountInterface {
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public:
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// Returns true if successful in setting the track.
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// Fails if an audio track is set on a video RtpSender, or vice-versa.
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virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
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virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
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2016-06-06 14:27:39 -07:00
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// Returns primary SSRC used by this sender for sending media.
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// Returns 0 if not yet determined.
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2018-06-21 13:32:56 +02:00
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// TODO(deadbeef): Change to absl::optional.
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// TODO(deadbeef): Remove? With GetParameters this should be redundant.
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2015-11-25 11:26:01 -08:00
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virtual uint32_t ssrc() const = 0;
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// Audio or video sender?
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virtual cricket::MediaType media_type() const = 0;
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2015-09-28 16:53:55 -07:00
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// Not to be confused with "mid", this is a field we can temporarily use
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// to uniquely identify a receiver until we implement Unified Plan SDP.
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virtual std::string id() const = 0;
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2018-04-02 16:31:36 -07:00
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// Returns a list of media stream ids associated with this sender's track.
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// These are signalled in the SDP so that the remote side can associate
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// tracks.
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virtual std::vector<std::string> stream_ids() const = 0;
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2018-10-01 22:47:20 +02:00
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// Returns the list of encoding parameters that will be applied when the SDP
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// local description is set. These initial encoding parameters can be set by
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// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
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// TODO(orphis): Make it pure virtual once Chrome has updated
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virtual std::vector<RtpEncodingParameters> init_send_encodings() const;
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2018-06-04 16:01:22 +02:00
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virtual RtpParameters GetParameters() = 0;
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// Note that only a subset of the parameters can currently be changed. See
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// rtpparameters.h
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// The encodings are in increasing quality order for simulcast.
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virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
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2017-02-01 20:27:00 -08:00
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// Returns null for a video sender.
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virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
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2018-08-29 17:02:10 -07:00
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// Sets a user defined frame encryptor that will encrypt the entire frame
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// before it is sent across the network. This will encrypt the entire frame
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// using the user provided encryption mechanism regardless of whether SRTP is
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// enabled or not.
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virtual void SetFrameEncryptor(
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rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
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// Returns a pointer to the frame encryptor set previously by the
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// user. This can be used to update the state of the object.
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virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const;
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2015-09-28 16:53:55 -07:00
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protected:
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~RtpSenderInterface() override = default;
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};
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// Define proxy for RtpSenderInterface.
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2017-02-08 01:38:21 -08:00
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// TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
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// are called on is an implementation detail.
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2016-04-15 03:49:07 -07:00
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BEGIN_SIGNALING_PROXY_MAP(RtpSender)
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2017-01-18 08:55:23 -08:00
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PROXY_SIGNALING_THREAD_DESTRUCTOR()
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PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
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PROXY_CONSTMETHOD0(uint32_t, ssrc)
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PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
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PROXY_CONSTMETHOD0(std::string, id)
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PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
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2018-10-01 22:47:20 +02:00
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PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
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2018-05-03 15:31:53 +02:00
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PROXY_METHOD0(RtpParameters, GetParameters);
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PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
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PROXY_METHOD1(void,
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SetFrameEncryptor,
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rtc::scoped_refptr<FrameEncryptorInterface>);
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PROXY_CONSTMETHOD0(rtc::scoped_refptr<FrameEncryptorInterface>,
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GetFrameEncryptor);
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2018-01-11 17:18:19 +01:00
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END_PROXY_MAP()
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2015-09-28 16:53:55 -07:00
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} // namespace webrtc
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2019-01-11 09:11:00 -08:00
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#endif // API_RTP_SENDER_INTERFACE_H_
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