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/call/video_send_stream.h b/call/video_send_stream.h
index a176709..a458203 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -17,9 +17,11 @@
#include <vector>
#include "webrtc/api/call/transport.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/call/rtp_config.h"
+#include "webrtc/call/video_config.h"
#include "webrtc/common_types.h"
#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/config.h"
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/media/base/videosourceinterface.h"
#include "webrtc/rtc_base/platform_file.h"
