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diff --git a/call/receive_stream.h b/call/receive_stream.h
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+/*
+ *  Copyright (c) 2021 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.
+ */
+
+#ifndef CALL_RECEIVE_STREAM_H_
+#define CALL_RECEIVE_STREAM_H_
+
+#include <vector>
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/media_types.h"
+#include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
+
+namespace webrtc {
+
+// Common base interface for MediaReceiveStream based classes and
+// FlexfecReceiveStream.
+class ReceiveStream {
+ public:
+  // Receive-stream specific RTP settings.
+  struct RtpConfig {
+    // Synchronization source (stream identifier) to be received.
+    // This member will not change mid-stream and can be assumed to be const
+    // post initialization.
+    uint32_t remote_ssrc = 0;
+
+    // Sender SSRC used for sending RTCP (such as receiver reports).
+    // This value may change mid-stream and must be done on the same thread
+    // that the value is read on (i.e. packet delivery).
+    uint32_t local_ssrc = 0;
+
+    // Enable feedback for send side bandwidth estimation.
+    // See
+    // https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions
+    // for details.
+    // This value may change mid-stream and must be done on the same thread
+    // that the value is read on (i.e. packet delivery).
+    bool transport_cc = false;
+
+    // RTP header extensions used for the received stream.
+    // This value may change mid-stream and must be done on the same thread
+    // that the value is read on (i.e. packet delivery).
+    std::vector<RtpExtension> extensions;
+  };
+
+  // Called on the packet delivery thread since some members of the config may
+  // change mid-stream (e.g. the local ssrc). All mutation must also happen on
+  // the packet delivery thread. Return value can be assumed to
+  // only be used in the calling context (on the stack basically).
+  virtual const RtpConfig& rtp_config() const = 0;
+
+ protected:
+  virtual ~ReceiveStream() {}
+};
+
+// Either an audio or video receive stream.
+class MediaReceiveStream : public ReceiveStream {
+ public:
+  // Starts stream activity.
+  // When a stream is active, it can receive, process and deliver packets.
+  virtual void Start() = 0;
+
+  // Stops stream activity. Must be called to match with a previous call to
+  // `Start()`. When a stream has been stopped, it won't receive, decode,
+  // process or deliver packets to downstream objects such as callback pointers
+  // set in the config struct.
+  virtual void Stop() = 0;
+
+  virtual void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<webrtc::FrameTransformerInterface>
+          frame_transformer) = 0;
+
+  virtual void SetFrameDecryptor(
+      rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) = 0;
+
+  virtual std::vector<RtpSource> GetSources() const = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_RECEIVE_STREAM_H_