Fixing WebRTC after moving from src/webrtc to src/

In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
diff --git a/modules/video_coding/jitter_buffer.cc b/modules/video_coding/jitter_buffer.cc
index a556080..0b65333 100644
--- a/modules/video_coding/jitter_buffer.cc
+++ b/modules/video_coding/jitter_buffer.cc
@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "webrtc/modules/video_coding/jitter_buffer.h"
+#include "modules/video_coding/jitter_buffer.h"
 
 #include <assert.h>
 
@@ -15,21 +15,21 @@
 #include <limits>
 #include <utility>
 
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/modules/video_coding/frame_buffer.h"
-#include "webrtc/modules/video_coding/include/video_coding.h"
-#include "webrtc/modules/video_coding/inter_frame_delay.h"
-#include "webrtc/modules/video_coding/internal_defines.h"
-#include "webrtc/modules/video_coding/jitter_buffer_common.h"
-#include "webrtc/modules/video_coding/jitter_estimator.h"
-#include "webrtc/modules/video_coding/packet.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/trace_event.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
-#include "webrtc/system_wrappers/include/metrics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/video_coding/frame_buffer.h"
+#include "modules/video_coding/include/video_coding.h"
+#include "modules/video_coding/inter_frame_delay.h"
+#include "modules/video_coding/internal_defines.h"
+#include "modules/video_coding/jitter_buffer_common.h"
+#include "modules/video_coding/jitter_estimator.h"
+#include "modules/video_coding/packet.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
 
 namespace webrtc {
 // Interval for updating SS data.