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+/*
+ * 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_