mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 13:50:40 +01:00
fbb7ce8a93
This reverts commit 021512b76a.
Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.
Original change's description:
> rtpsender interface: make pure virtual again
>
> after providing default implementations in Chromium tests
>
> BUG=None
>
> Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37941}
Bug: None
Change-Id: I40f27c36819365fadae32032521f7e11184bee62
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
Owners-Override: Andrey Logvin <landrey@google.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37947}
113 lines
4.5 KiB
C++
113 lines
4.5 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 RtpSenders
|
|
// http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
|
|
|
|
#ifndef API_RTP_SENDER_INTERFACE_H_
|
|
#define API_RTP_SENDER_INTERFACE_H_
|
|
|
|
#include <memory>
|
|
#include <string>
|
|
#include <vector>
|
|
|
|
#include "api/crypto/frame_encryptor_interface.h"
|
|
#include "api/dtls_transport_interface.h"
|
|
#include "api/dtmf_sender_interface.h"
|
|
#include "api/frame_transformer_interface.h"
|
|
#include "api/media_stream_interface.h"
|
|
#include "api/media_types.h"
|
|
#include "api/rtc_error.h"
|
|
#include "api/rtp_parameters.h"
|
|
#include "api/scoped_refptr.h"
|
|
#include "api/video_codecs/video_encoder_factory.h"
|
|
#include "rtc_base/ref_count.h"
|
|
#include "rtc_base/system/rtc_export.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RTC_EXPORT RtpSenderInterface : public rtc::RefCountInterface {
|
|
public:
|
|
// Returns true if successful in setting the track.
|
|
// Fails if an audio track is set on a video RtpSender, or vice-versa.
|
|
virtual bool SetTrack(MediaStreamTrackInterface* track) = 0;
|
|
virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
|
|
|
|
// The dtlsTransport attribute exposes the DTLS transport on which the
|
|
// media is sent. It may be null.
|
|
// https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-transport
|
|
// TODO(https://bugs.webrtc.org/907849) remove default implementation
|
|
virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
|
|
|
|
// Returns primary SSRC used by this sender for sending media.
|
|
// Returns 0 if not yet determined.
|
|
// TODO(deadbeef): Change to absl::optional.
|
|
// TODO(deadbeef): Remove? With GetParameters this should be redundant.
|
|
virtual uint32_t ssrc() const = 0;
|
|
|
|
// Audio or video sender?
|
|
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;
|
|
|
|
// Returns a list of media stream ids associated with this sender's track.
|
|
// These are signalled in the SDP so that the remote side can associate
|
|
// tracks.
|
|
virtual std::vector<std::string> stream_ids() const = 0;
|
|
|
|
// Sets the IDs of the media streams associated with this sender's track.
|
|
// These are signalled in the SDP so that the remote side can associate
|
|
// tracks.
|
|
virtual void SetStreams(const std::vector<std::string>& stream_ids) {}
|
|
|
|
// Returns the list of encoding parameters that will be applied when the SDP
|
|
// local description is set. These initial encoding parameters can be set by
|
|
// PeerConnection::AddTransceiver, and later updated with Get/SetParameters.
|
|
// TODO(orphis): Make it pure virtual once Chrome has updated
|
|
virtual std::vector<RtpEncodingParameters> init_send_encodings() const;
|
|
|
|
virtual RtpParameters GetParameters() const = 0;
|
|
// Note that only a subset of the parameters can currently be changed. See
|
|
// rtpparameters.h
|
|
// The encodings are in increasing quality order for simulcast.
|
|
virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
|
|
|
|
// Returns null for a video sender.
|
|
virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
|
|
|
|
// Sets a user defined frame encryptor that will encrypt the entire frame
|
|
// before it is sent across the network. This will encrypt the entire frame
|
|
// using the user provided encryption mechanism regardless of whether SRTP is
|
|
// enabled or not.
|
|
virtual void SetFrameEncryptor(
|
|
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor);
|
|
|
|
// Returns a pointer to the frame encryptor set previously by the
|
|
// user. This can be used to update the state of the object.
|
|
virtual rtc::scoped_refptr<FrameEncryptorInterface> GetFrameEncryptor() const;
|
|
|
|
virtual void SetEncoderToPacketizerFrameTransformer(
|
|
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
|
|
|
|
// Sets a user defined encoder selector.
|
|
// Overrides selector that is (optionally) provided by VideoEncoderFactory.
|
|
virtual void SetEncoderSelector(
|
|
std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
|
|
encoder_selector) {}
|
|
|
|
protected:
|
|
~RtpSenderInterface() override = default;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // API_RTP_SENDER_INTERFACE_H_
|