Repo created

TMessagesProj/jni/voip/webrtc/pc/rtp_receiver.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 classes that implement RtpReceiverInterface.
+// An RtpReceiver associates a MediaStreamTrackInterface with an underlying
+// transport (provided by cricket::VoiceChannel/cricket::VideoChannel)
+
+#ifndef PC_RTP_RECEIVER_H_
+#define PC_RTP_RECEIVER_H_
+
+#include <stdint.h>
+
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/dtls_transport_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "api/video/video_source_interface.h"
+#include "media/base/media_channel.h"
+#include "media/base/video_broadcaster.h"
+#include "pc/video_track_source.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// Internal class used by PeerConnection.
+class RtpReceiverInternal : public RtpReceiverInterface {
+ public:
+  // Call on the signaling thread, to let the receiver know that the the
+  // embedded source object should enter a stopped/ended state and the track's
+  // state set to `kEnded`, a final state that cannot be reversed.
+  virtual void Stop() = 0;
+
+  // Sets the underlying MediaEngine channel associated with this RtpSender.
+  // A VoiceMediaChannel should be used for audio RtpSenders and
+  // a VideoMediaChannel should be used for video RtpSenders.
+  // NOTE:
+  // * SetMediaChannel(nullptr) must be called before the media channel is
+  //   destroyed.
+  // * This method must be invoked on the worker thread.
+  virtual void SetMediaChannel(
+      cricket::MediaReceiveChannelInterface* media_channel) = 0;
+
+  // Configures the RtpReceiver with the underlying media channel, with the
+  // given SSRC as the stream identifier.
+  virtual void SetupMediaChannel(uint32_t ssrc) = 0;
+
+  // Configures the RtpReceiver with the underlying media channel to receive an
+  // unsignaled receive stream.
+  virtual void SetupUnsignaledMediaChannel() = 0;
+
+  virtual void set_transport(
+      rtc::scoped_refptr<DtlsTransportInterface> dtls_transport) = 0;
+  // This SSRC is used as an identifier for the receiver between the API layer
+  // and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
+  virtual absl::optional<uint32_t> ssrc() const = 0;
+
+  // Call this to notify the RtpReceiver when the first packet has been received
+  // on the corresponding channel.
+  virtual void NotifyFirstPacketReceived() = 0;
+
+  // Set the associated remote media streams for this receiver. The remote track
+  // will be removed from any streams that are no longer present and added to
+  // any new streams.
+  virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
+  // TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
+  // set_stream_ids() as soon as downstream projects are no longer dependent on
+  // stream objects.
+  virtual void SetStreams(
+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0;
+
+  // Returns an ID that changes if the attached track changes, but
+  // otherwise remains constant. Used to generate IDs for stats.
+  // The special value zero means that no track is attached.
+  virtual int AttachmentId() const = 0;
+
+ protected:
+  static int GenerateUniqueId();
+
+  static std::vector<rtc::scoped_refptr<MediaStreamInterface>>
+  CreateStreamsFromIds(std::vector<std::string> stream_ids);
+};
+
+}  // namespace webrtc
+
+#endif  // PC_RTP_RECEIVER_H_