Use rtp_header_extension_map.h instead of rtp_header_extension.h
Finish renaming started in the https://chromium-review.googlesource.com/c/520947/
Bug: webrtc:5565
Change-Id: If420e05165ef7c110b7d38f53dbe73c21a4059bc
Reviewed-on: https://chromium-review.googlesource.com/528095
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#18538}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 84b4d2c1c2db7f65c3d67238ce4af0042885ac97diff --git a/call/call.cc b/call/call.cc
index f31e114..c4ce716 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -42,9 +42,9 @@
#include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/modules/utility/include/process_thread.h"
#include "webrtc/system_wrappers/include/clock.h"
diff --git a/call/rtx_receive_stream_unittest.cc b/call/rtx_receive_stream_unittest.cc
index 6bb067b..91ed2ca 100644
--- a/call/rtx_receive_stream_unittest.cc
+++ b/call/rtx_receive_stream_unittest.cc
@@ -9,9 +9,9 @@
*/
#include "webrtc/call/rtx_receive_stream.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index 58e0dc2..cd431bf 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -17,8 +17,8 @@
#include "webrtc/base/ignore_wundef.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_send_stream.h"
diff --git a/logging/rtc_event_log/rtc_event_log_unittest.cc b/logging/rtc_event_log/rtc_event_log_unittest.cc
index b5c91fa..07b394a 100644
--- a/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -24,9 +24,9 @@
#include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "webrtc/test/gtest.h"
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 22b614c..dc623ce 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -110,7 +110,6 @@
"source/rtp_format_vp8.h",
"source/rtp_format_vp9.cc",
"source/rtp_format_vp9.h",
- "source/rtp_header_extension.h",
"source/rtp_header_extension_map.cc",
"source/rtp_header_extensions.cc",
"source/rtp_header_extensions.h",
diff --git a/modules/rtp_rtcp/source/rtp_header_extension.h b/modules/rtp_rtcp/source/rtp_header_extension.h
deleted file mode 100644
index 2a03fec..0000000
--- a/modules/rtp_rtcp/source/rtp_header_extension.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
-
-// DEPRECATED, use include below instead.
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSION_H_
diff --git a/test/fuzzers/rtp_packet_fuzzer.cc b/test/fuzzers/rtp_packet_fuzzer.cc
index 64ab6ec..da62948 100644
--- a/test/fuzzers/rtp_packet_fuzzer.cc
+++ b/test/fuzzers/rtp_packet_fuzzer.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
