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_