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cc189177a6
This reverts commit df5731e44d.
Reason for revert: Breaks WebRTC in Chrome FYI for all platforms.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/2966
Original change's description:
> Improve spec compliance of SetStreamIDs in RtpSenderInterface
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> This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded
> event if needed and exposes the method on RtpSenderInterface.
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> This is a spec-compliance change.
>
> Bug: webrtc:10129
> Change-Id: I2b98b92665c847102838b094516a79b24de0e47e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27974}
TBR=steveanton@webrtc.org,hbos@webrtc.org,guidou@webrtc.org
# Passing all bots except for flaky webrtc_perf_tests
NOTRY=True
Bug: webrtc:10129
Change-Id: If97317f7a01b34465685fcebbeea0d7576ed7328
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137431
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27988}
119 lines
4.9 KiB
C++
119 lines
4.9 KiB
C++
/*
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* Copyright 2015 The WebRTC project authors. All Rights Reserved.
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*
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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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*/
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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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#ifndef API_RTP_SENDER_INTERFACE_H_
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#define API_RTP_SENDER_INTERFACE_H_
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#include <string>
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#include <vector>
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#include "api/crypto/frame_encryptor_interface.h"
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#include "api/dtls_transport_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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#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 "api/scoped_refptr.h"
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#include "rtc_base/ref_count.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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// The dtlsTransport attribute exposes the DTLS transport on which the
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// media is sent. It may be null.
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// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
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// TODO(https://bugs.webrtc.org/907849) remove default implementation
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virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
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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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// TODO(deadbeef): Change to absl::optional.
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// TODO(deadbeef): Remove? With GetParameters this should be redundant.
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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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// 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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// 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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// 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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virtual RtpParameters GetParameters() const = 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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// Returns null for a video sender.
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virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
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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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protected:
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~RtpSenderInterface() override = default;
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};
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// Define proxy for RtpSenderInterface.
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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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BEGIN_SIGNALING_PROXY_MAP(RtpSender)
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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(rtc::scoped_refptr<DtlsTransportInterface>, dtls_transport)
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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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PROXY_CONSTMETHOD0(std::vector<RtpEncodingParameters>, init_send_encodings)
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PROXY_CONSTMETHOD0(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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END_PROXY_MAP()
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} // namespace webrtc
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#endif // API_RTP_SENDER_INTERFACE_H_
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