Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Original-Commit-Position: refs/heads/master@{#19639}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 1acbd68718476b2754a7872fb72e3a8a74166eb9
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 4678b8f..0dbb2f7 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -14,8 +14,8 @@
#include <utility>
#include <vector>
+#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/modules/pacing/packet_router.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
