Rename ReceiveStream to ReceiveStreamInterface

Bug: webrtc:7484
Change-Id: I41176a66b8399f6c8cf568630f2808eb95cf6247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262767
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36917}
diff --git a/call/call.cc b/call/call.cc
index f96bda5..078aa2e 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -85,7 +85,7 @@
          map.IsRegistered(kRtpExtensionTransportSequenceNumber02);
 }
 
-bool UseSendSideBwe(const ReceiveStream* stream) {
+bool UseSendSideBwe(const ReceiveStreamInterface* stream) {
   return stream->transport_cc() &&
          HasTransportSequenceNumber(stream->GetRtpExtensionMap());
 }
@@ -361,7 +361,7 @@
 
   bool IdentifyReceivedPacket(RtpPacketReceived& packet,
                               bool* use_send_side_bwe = nullptr);
-  bool RegisterReceiveStream(uint32_t ssrc, ReceiveStream* stream);
+  bool RegisterReceiveStream(uint32_t ssrc, ReceiveStreamInterface* stream);
   bool UnregisterReceiveStream(uint32_t ssrc);
 
   void UpdateAggregateNetworkState();
@@ -416,7 +416,7 @@
 
   // TODO(bugs.webrtc.org/11993): Move receive_rtp_config_ over to the
   // network thread.
-  std::map<uint32_t, ReceiveStream*> receive_rtp_config_
+  std::map<uint32_t, ReceiveStreamInterface*> receive_rtp_config_
       RTC_GUARDED_BY(&receive_11993_checker_);
 
   // Audio and Video send streams are owned by the client that creates them.
@@ -1710,7 +1710,8 @@
   return true;
 }
 
-bool Call::RegisterReceiveStream(uint32_t ssrc, ReceiveStream* stream) {
+bool Call::RegisterReceiveStream(uint32_t ssrc,
+                                 ReceiveStreamInterface* stream) {
   RTC_DCHECK_RUN_ON(&receive_11993_checker_);
   RTC_DCHECK(stream);
   auto inserted = receive_rtp_config_.emplace(ssrc, stream);
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index 118eb0b..31aa99d 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -25,7 +25,7 @@
 namespace webrtc {
 
 class FlexfecReceiveStream : public RtpPacketSinkInterface,
-                             public ReceiveStream {
+                             public ReceiveStreamInterface {
  public:
   ~FlexfecReceiveStream() override = default;
 
diff --git a/call/flexfec_receive_stream_impl.h b/call/flexfec_receive_stream_impl.h
index 857715a..48edf09 100644
--- a/call/flexfec_receive_stream_impl.h
+++ b/call/flexfec_receive_stream_impl.h
@@ -58,7 +58,7 @@
 
   Stats GetStats() const override;
 
-  // ReceiveStream impl.
+  // ReceiveStreamInterface impl.
   void SetRtpExtensions(std::vector<RtpExtension> extensions) override;
   RtpHeaderExtensionMap GetRtpExtensionMap() const override;
 
diff --git a/call/receive_stream.h b/call/receive_stream.h
index 0464a70..4a946d6 100644
--- a/call/receive_stream.h
+++ b/call/receive_stream.h
@@ -24,7 +24,7 @@
 
 // Common base interface for MediaReceiveStream based classes and
 // FlexfecReceiveStream.
-class ReceiveStream {
+class ReceiveStreamInterface {
  public:
   // Receive-stream specific RTP settings.
   // TODO(tommi): This struct isn't needed at this level anymore. Move it closer
@@ -68,11 +68,11 @@
   virtual bool transport_cc() const = 0;
 
  protected:
-  virtual ~ReceiveStream() {}
+  virtual ~ReceiveStreamInterface() {}
 };
 
 // Either an audio or video receive stream.
-class MediaReceiveStream : public ReceiveStream {
+class MediaReceiveStream : public ReceiveStreamInterface {
  public:
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
@@ -94,6 +94,10 @@
   virtual std::vector<RtpSource> GetSources() const = 0;
 };
 
+// TODO(bugs.webrtc.org/7484): Remove this once downstream usage of the
+// deprecated name is gone.
+using ReceiveStream [[deprecated]] = ReceiveStreamInterface;
+
 }  // namespace webrtc
 
 #endif  // CALL_RECEIVE_STREAM_H_