Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 123 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/rtp_receiver_interface.h b/third_party/libwebrtc/api/rtp_receiver_interface.h
new file mode 100644
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+++ b/third_party/libwebrtc/api/rtp_receiver_interface.h
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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 RtpReceivers
+// http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
+
+#ifndef API_RTP_RECEIVER_INTERFACE_H_
+#define API_RTP_RECEIVER_INTERFACE_H_
+
+#include <string>
+#include <vector>
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/dtls_transport_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
+#include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+class RtpReceiverObserverInterface {
+ public:
+  // Note: Currently if there are multiple RtpReceivers of the same media type,
+  // they will all call OnFirstPacketReceived at once.
+  //
+  // In the future, it's likely that an RtpReceiver will only call
+  // OnFirstPacketReceived when a packet is received specifically for its
+  // SSRC/mid.
+  virtual void OnFirstPacketReceived(cricket::MediaType media_type) = 0;
+
+ protected:
+  virtual ~RtpReceiverObserverInterface() {}
+};
+
+class RTC_EXPORT RtpReceiverInterface : public rtc::RefCountInterface {
+ public:
+  virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
+
+  // The dtlsTransport attribute exposes the DTLS transport on which the
+  // media is received. It may be null.
+  // https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-transport
+  // TODO(https://bugs.webrtc.org/907849) remove default implementation
+  virtual rtc::scoped_refptr<DtlsTransportInterface> dtls_transport() const;
+
+  // The list of streams that `track` is associated with. This is the same as
+  // the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
+  // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
+  // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
+  // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of
+  // stream_ids() as soon as downstream projects are no longer dependent on
+  // stream objects.
+  virtual std::vector<std::string> stream_ids() const;
+  virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
+
+  // Audio or video receiver?
+  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;
+
+  // The WebRTC specification only defines RTCRtpParameters in terms of senders,
+  // but this API also applies them to receivers, similar to ORTC:
+  // http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+  virtual RtpParameters GetParameters() const = 0;
+  // TODO(dinosaurav): Delete SetParameters entirely after rolling to Chromium.
+  // Currently, doesn't support changing any parameters.
+  virtual bool SetParameters(const RtpParameters& parameters) { return false; }
+
+  // Does not take ownership of observer.
+  // Must call SetObserver(nullptr) before the observer is destroyed.
+  virtual void SetObserver(RtpReceiverObserverInterface* observer) = 0;
+
+  // Sets the jitter buffer minimum delay until media playout. Actual observed
+  // delay may differ depending on the congestion control. `delay_seconds` is a
+  // positive value including 0.0 measured in seconds. `nullopt` means default
+  // value must be used.
+  virtual void SetJitterBufferMinimumDelay(
+      absl::optional<double> delay_seconds) = 0;
+
+  // TODO(zhihuang): Remove the default implementation once the subclasses
+  // implement this. Currently, the only relevant subclass is the
+  // content::FakeRtpReceiver in Chromium.
+  virtual std::vector<RtpSource> GetSources() const;
+
+  // Sets a user defined frame decryptor that will decrypt the entire frame
+  // before it is sent across the network. This will decrypt the entire frame
+  // using the user provided decryption mechanism regardless of whether SRTP is
+  // enabled or not.
+  // TODO(bugs.webrtc.org/12772): Remove.
+  virtual void SetFrameDecryptor(
+      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
+
+  // Returns a pointer to the frame decryptor set previously by the
+  // user. This can be used to update the state of the object.
+  // TODO(bugs.webrtc.org/12772): Remove.
+  virtual rtc::scoped_refptr<FrameDecryptorInterface> GetFrameDecryptor() const;
+
+  // Sets a frame transformer between the depacketizer and the decoder to enable
+  // client code to transform received frames according to their own processing
+  // logic.
+  virtual void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+
+ protected:
+  ~RtpReceiverInterface() override = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_RTP_RECEIVER_INTERFACE_H_