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As the synchronous version only posts a task to recreate the encoder later, it is not possible to catch errors and state changes that could appear then. The asynchronous version of SetParameters() aims to solve this by providing a callback to wait for the completion of the encoder reconfiguration, allowing any error to be propagate and subsequent getParameters() call to have up to date information. Bug: webrtc:11607 Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38627}
123 lines
4.9 KiB
C++
123 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 <memory>
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#include <string>
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#include <vector>
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#include "absl/functional/any_invocable.h"
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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/frame_transformer_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/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 "api/video_codecs/video_encoder_factory.h"
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#include "rtc_base/ref_count.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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using SetParametersCallback = absl::AnyInvocable<void(RTCError) &&>;
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class RTC_EXPORT 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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virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0;
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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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// Sets the IDs of the media streams 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 void SetStreams(const std::vector<std::string>& stream_ids) = 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 = 0;
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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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virtual void SetParametersAsync(const RtpParameters& parameters,
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SetParametersCallback callback);
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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) = 0;
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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()
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const = 0;
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virtual void SetEncoderToPacketizerFrameTransformer(
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rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0;
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// Sets a user defined encoder selector.
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// Overrides selector that is (optionally) provided by VideoEncoderFactory.
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virtual void SetEncoderSelector(
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std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
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encoder_selector) = 0;
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// TODO(crbug.com/1354101): make pure virtual again after Chrome roll.
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virtual RTCError GenerateKeyFrame(const std::vector<std::string>& rids) {
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return RTCError::OK();
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}
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protected:
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~RtpSenderInterface() override = default;
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};
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} // namespace webrtc
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#endif // API_RTP_SENDER_INTERFACE_H_
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