Move, almost, all receive side references to RTP to RtpStreamReceiver.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
index 301cb84765e41a13ba5b6f4c60fcb19cc5083ed6..66589888bda4263496fdc99887bb2c56c4fe1a2d 100644
--- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -540,6 +540,9 @@ class RtpRtcp : public Module {
 
     /*
     *   Send NACK for the packets specified.
+    *
+    *   Note: This assumes the caller keeps track of timing and doesn't rely on
+    *   the RTP module to do this.
     */
     virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;