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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/api/rtp_sender_interface.h | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/api/rtp_sender_interface.h | 123 |
1 files changed, 123 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/rtp_sender_interface.h b/third_party/libwebrtc/api/rtp_sender_interface.h new file mode 100644 index 0000000000..98ee91b1cc --- /dev/null +++ b/third_party/libwebrtc/api/rtp_sender_interface.h @@ -0,0 +1,123 @@ +/* + * 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 "absl/functional/any_invocable.h" +#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" + +#include "api/rtp_sender_setparameters_callback.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 + virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const = 0; + + // 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) = 0; + + // 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 = 0; + + 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; + virtual void SetParametersAsync(const RtpParameters& parameters, + SetParametersCallback callback); + + // 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) = 0; + + // 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 = 0; + + virtual void SetEncoderToPacketizerFrameTransformer( + rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) = 0; + + // Sets a user defined encoder selector. + // Overrides selector that is (optionally) provided by VideoEncoderFactory. + virtual void SetEncoderSelector( + std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface> + encoder_selector) = 0; + + // TODO(crbug.com/1354101): make pure virtual again after Chrome roll. + virtual RTCError GenerateKeyFrame(const std::vector<std::string>& rids) { + return RTCError::OK(); + } + + protected: + ~RtpSenderInterface() override = default; +}; + +} // namespace webrtc + +#endif // API_RTP_SENDER_INTERFACE_H_ |