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}

modules/audio_coding/BUILD.gn

@@ -871,13 +871,13 @@ if (rtc_enable_protobuf) {
     deps = [
       ":ana_config_proto",
     ]
-    proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor"
+    proto_out_dir = "modules/audio_coding/audio_network_adaptor"
   }
   proto_library("ana_config_proto") {
     sources = [
       "audio_network_adaptor/config.proto",
     ]
-    proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor"
+    proto_out_dir = "modules/audio_coding/audio_network_adaptor"
   }
 }
 
@@ -1461,7 +1461,7 @@ if (rtc_include_tests) {
   }  # insert_packet_with_timing
 
   audio_decoder_unittests_resources =
-      [ "../../../resources/audio_coding/testfile32kHz.pcm" ]
+      [ "../../resources/audio_coding/testfile32kHz.pcm" ]
 
   if (is_ios) {
     bundle_data("audio_decoder_unittests_bundle_data") {
@@ -1519,7 +1519,7 @@ if (rtc_include_tests) {
       sources = [
         "neteq/neteq_unittest.proto",
       ]
-      proto_out_dir = "webrtc/modules/audio_coding/neteq"
+      proto_out_dir = "modules/audio_coding/neteq"
     }
 
     rtc_test("neteq_rtpplay") {
@@ -1940,7 +1940,7 @@ if (rtc_include_tests) {
     ]
 
     data = [
-      "../../../resources/speech_and_misc_wb.pcm",
+      "../../resources/speech_and_misc_wb.pcm",
     ]
 
     if (is_win) {

modules/audio_coding/DEPS

@@ -1,7 +1,7 @@
 include_rules = [
-  "+webrtc/call",
-  "+webrtc/common_audio",
-  "+webrtc/logging/rtc_event_log",
-  "+webrtc/audio_coding/neteq/neteq_unittest.pb.h",  # Different path.
-  "+webrtc/system_wrappers",
+  "+call",
+  "+common_audio",
+  "+logging/rtc_event_log",
+  "+audio_coding/neteq/neteq_unittest.pb.h",  # Different path.
+  "+system_wrappers",
 ]

modules/audio_coding/acm2/acm_codec_database.cc

@@ -15,11 +15,11 @@
 
 // TODO(tlegrand): Change constant input pointers in all functions to constant
 // references, where appropriate.
-#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
+#include "modules/audio_coding/acm2/acm_codec_database.h"
 
 #include <assert.h>
 
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
 
 #if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
 #error iSAC and iSACFX codecs cannot be enabled at the same time

modules/audio_coding/acm2/acm_codec_database.h

@@ -13,12 +13,12 @@
  * codecs.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
 
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "typedefs.h"
 
 namespace webrtc {
 
@@ -79,4 +79,4 @@ class ACMCodecDB {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_

modules/audio_coding/acm2/acm_receive_test.cc

@@ -8,20 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/acm_receive_test.h"
+#include "modules/audio_coding/acm2/acm_receive_test.h"
 
 #include <assert.h>
 #include <stdio.h>
 
 #include <memory>
 
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/test/gtest.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 namespace test {

modules/audio_coding/acm2/acm_receive_test.h

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
 
 #include <memory>
 #include <string>
 
-#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 class AudioCodingModule;
@@ -93,4 +93,4 @@ class AcmReceiveTestToggleOutputFreqOldApi : public AcmReceiveTestOldApi {
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_

modules/audio_coding/acm2/acm_receiver.cc

@@ -8,25 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
 
 #include <stdlib.h>  // malloc
 
 #include <algorithm>  // sort
 #include <vector>
 
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_types.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/format_macros.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 

modules/audio_coding/acm2/acm_receiver.h

@@ -8,25 +8,25 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/api/array_view.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/common_audio/vad/include/webrtc_vad.h"
-#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/typedefs.h"
+#include "api/array_view.h"
+#include "api/optional.h"
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/thread_annotations.h"
+#include "typedefs.h"
 
 namespace webrtc {
 
@@ -291,4 +291,4 @@ class AcmReceiver {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_

modules/audio_coding/acm2/acm_receiver_unittest.cc

@@ -8,20 +8,20 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
 
 #include <algorithm>  // std::min
 #include <memory>
 
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
 
 namespace webrtc {
 

modules/audio_coding/acm2/acm_resampler.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
 
 #include <assert.h>
 #include <string.h>
 
-#include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/rtc_base/logging.h"
+#include "common_audio/resampler/include/resampler.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 namespace acm2 {

modules/audio_coding/acm2/acm_resampler.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
 
-#include "webrtc/common_audio/resampler/include/push_resampler.h"
-#include "webrtc/typedefs.h"
+#include "common_audio/resampler/include/push_resampler.h"
+#include "typedefs.h"
 
 namespace webrtc {
 namespace acm2 {
@@ -36,4 +36,4 @@ class ACMResampler {
 }  // namespace acm2
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_

modules/audio_coding/acm2/acm_send_test.cc

@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
+#include "modules/audio_coding/acm2/acm_send_test.h"
 
 #include <assert.h>
 #include <stdio.h>
 #include <string.h>
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/gtest.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 namespace test {

modules/audio_coding/acm2/acm_send_test.h

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+#define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
 
 #include <memory>
 #include <vector>
 
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/constructormagic.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 class AudioEncoder;
@@ -85,4 +85,4 @@ class AcmSendTestOldApi : public AudioPacketizationCallback,
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_

modules/audio_coding/acm2/audio_coding_module.cc

@@ -8,19 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
 
 #include <algorithm>
 
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
-#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/system_wrappers/include/metrics.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/acm2/acm_receiver.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/acm2/codec_manager.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/safe_conversions.h"
+#include "system_wrappers/include/metrics.h"
 
 namespace webrtc {
 

modules/audio_coding/acm2/audio_coding_module_unittest.cc

@@ -13,38 +13,38 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/acm2/acm_receive_test.h"
-#include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/output_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/output_wav_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/md5digest.h"
-#include "webrtc/rtc_base/platform_thread.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder.h"
-#include "webrtc/test/mock_audio_encoder.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/acm2/acm_receive_test.h"
+#include "modules/audio_coding/acm2/acm_send_test.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "modules/audio_coding/neteq/tools/audio_checksum.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_audio_file.h"
+#include "modules/audio_coding/neteq/tools/output_wav_file.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/md5digest.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "system_wrappers/include/sleep.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_encoder.h"
+#include "test/testsupport/fileutils.h"
 
 using ::testing::AtLeast;
 using ::testing::Invoke;

modules/audio_coding/acm2/call_statistics.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
 
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

modules/audio_coding/acm2/call_statistics.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
 
-#include "webrtc/common_types.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "common_types.h"
+#include "modules/include/module_common_types.h"
 
 //
 // This class is for book keeping of calls to ACM. It is not useful to log API
@@ -61,4 +61,4 @@ class CallStatistics {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_

modules/audio_coding/acm2/call_statistics_unittest.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/call_statistics.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/acm2/call_statistics.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/acm2/codec_manager.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
+#include "modules/audio_coding/acm2/codec_manager.h"
 
-#include "webrtc/rtc_base/checks.h"
-//#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/checks.h"
+//#include "rtc_base/format_macros.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "rtc_base/logging.h"
+#include "typedefs.h"
 
 namespace webrtc {
 namespace acm2 {

modules/audio_coding/acm2/codec_manager.h

@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#define MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
 
 #include <map>
 
-#include "webrtc/api/optional.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/thread_checker.h"
+#include "api/optional.h"
+#include "common_types.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/thread_checker.h"
 
 namespace webrtc {
 
@@ -72,4 +72,4 @@ class CodecManager final {
 
 }  // namespace acm2
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_

modules/audio_coding/acm2/codec_manager_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_encoder.h"
+#include "modules/audio_coding/acm2/codec_manager.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
 
 namespace webrtc {
 namespace acm2 {

modules/audio_coding/acm2/rent_a_codec.cc

@@ -8,39 +8,39 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
 
 #include <memory>
 #include <utility>
 
-#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
-#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
-#include "webrtc/rtc_base/logging.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "rtc_base/logging.h"
 #ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 #endif
 #ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
 #endif
 #ifdef WEBRTC_CODEC_ISACFX
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"  // nogncheck
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"  // nogncheck
+#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"  // nogncheck
+#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"  // nogncheck
 #endif
 #ifdef WEBRTC_CODEC_ISAC
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"  // nogncheck
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"  // nogncheck
+#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"  // nogncheck
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"  // nogncheck
 #endif
 #ifdef WEBRTC_CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 #endif
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
 #ifdef WEBRTC_CODEC_RED
-#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
 #endif
-#include "webrtc/modules/audio_coding/acm2/acm_codec_database.h"
+#include "modules/audio_coding/acm2/acm_codec_database.h"
 
 #if defined(WEBRTC_CODEC_ISACFX) || defined(WEBRTC_CODEC_ISAC)
-#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 #endif
 
 namespace webrtc {

modules/audio_coding/acm2/rent_a_codec.h

@@ -8,22 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#ifndef MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#define MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
 
 #include <stddef.h>
 #include <map>
 #include <memory>
 
-#include "webrtc/api/array_view.h"
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/optional.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"
 
 namespace webrtc {
 
@@ -204,4 +204,4 @@ class RentACodec {
 }  // namespace acm2
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#endif  // MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_

modules/audio_coding/acm2/rent_a_codec_unittest.cc

@@ -10,10 +10,10 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_encoder.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "rtc_base/arraysize.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
 
 namespace webrtc {
 namespace acm2 {

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 
 #include <utility>
 
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/timeutils.h"
+#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h

@@ -8,17 +8,17 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -90,4 +90,4 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_AUDIO_NETWORK_ADAPTOR_IMPL_H_

modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc

@@ -11,14 +11,14 @@
 #include <utility>
 #include <vector>
 
-#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
-#include "webrtc/rtc_base/fakeclock.h"
-#include "webrtc/test/field_trial.h"
-#include "webrtc/test/gtest.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "rtc_base/fakeclock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/bitrate_controller.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 
 #include <algorithm>
 
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {

modules/audio_coding/audio_network_adaptor/bitrate_controller.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {
@@ -46,4 +46,4 @@ class BitrateController final : public Controller {
 }  // namespace audio_network_adaptor
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_BITRATE_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-#include "webrtc/test/field_trial.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 namespace audio_network_adaptor {

modules/audio_coding/audio_network_adaptor/channel_controller.cc

@@ -10,8 +10,8 @@
 
 #include <algorithm>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/channel_controller.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -50,4 +50,4 @@ class ChannelController final : public Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/controller.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/controller.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
 
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "api/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 
 namespace webrtc {
 
@@ -40,4 +40,4 @@ class Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/controller_manager.cc

@@ -8,28 +8,28 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
 
 #include <cmath>
 #include <utility>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/rtc_base/timeutils.h"
+#include "modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/channel_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/timeutils.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 #endif

modules/audio_coding/audio_network_adaptor/controller_manager.h

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
 
 #include <map>
 #include <memory>
 #include <vector>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/protobuf_utils.h"
 
 namespace webrtc {
 
@@ -120,4 +120,4 @@ class ControllerManagerImpl final : public ControllerManager {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_

modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc

@@ -10,20 +10,20 @@
 
 #include <utility>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
-#include "webrtc/rtc_base/fakeclock.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
+#include "rtc_base/fakeclock.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/protobuf_utils.h"
+#include "test/gtest.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 #endif

modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc

@@ -8,18 +8,18 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
 
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/protobuf_utils.h"
 
 #if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
 #else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 #endif

modules/audio_coding/audio_network_adaptor/debug_dump_writer.h

@@ -8,22 +8,22 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/system_wrappers/include/file_wrapper.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/ignore_wundef.h"
+#include "system_wrappers/include/file_wrapper.h"
 #if WEBRTC_ENABLE_PROTOBUF
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #else
-#include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
+#include "modules/audio_coding/audio_network_adaptor/config.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 #endif
@@ -52,4 +52,4 @@ class DebugDumpWriter {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP_WRITER_H_

modules/audio_coding/audio_network_adaptor/dtx_controller.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/dtx_controller.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -46,4 +46,4 @@ class DtxController final : public Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/dtx_controller.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/event_log_writer.cc

@@ -11,8 +11,8 @@
 #include <math.h>
 #include <algorithm>
 
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/event_log_writer.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 class RtcEventLog;
@@ -39,4 +39,4 @@ class EventLogWriter final {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_EVENT_LOG_WRITER_H_

modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc

@@ -10,9 +10,9 @@
 
 #include <memory>
 
-#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
-#include "webrtc/test/gtest.h"
+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
+#include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
 
 #include <limits>
 #include <utility>
 
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h

@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
 
 #include <memory>
 
-#include "webrtc/common_audio/smoothing_filter.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -69,4 +69,4 @@ class FecControllerPlrBased final : public Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_PLR_BASED_H_

modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc

@@ -10,9 +10,9 @@
 
 #include <utility>
 
-#include "webrtc/common_audio/mocks/mock_smoothing_filter.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
-#include "webrtc/test/gtest.h"
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
 
 #include <limits>
 #include <utility>
 
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -63,4 +63,4 @@ class FecControllerRplrBased final : public Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FEC_CONTROLLER_RPLR_BASED_H_

modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc

@@ -11,8 +11,8 @@
 #include <random>
 #include <utility>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/frame_length_controller.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
 
 #include <utility>
 
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/frame_length_controller.h

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
 
 #include <map>
 #include <vector>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -78,4 +78,4 @@ class FrameLengthController final : public Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_FRAME_LENGTH_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc

@@ -11,8 +11,8 @@
 #include <memory>
 #include <utility>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/optional.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
 
 namespace webrtc {
 
@@ -49,4 +49,4 @@ class AudioNetworkAdaptor {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_

modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
 
-#include "webrtc/api/optional.h"
+#include "api/optional.h"
 
 namespace webrtc {
 
@@ -36,4 +36,4 @@ struct AudioEncoderRuntimeConfig {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_

modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/test/gmock.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "test/gmock.h"
 
 namespace webrtc {
 
@@ -46,4 +46,4 @@ class MockAudioNetworkAdaptor : public AudioNetworkAdaptor {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_AUDIO_NETWORK_ADAPTOR_H_

modules/audio_coding/audio_network_adaptor/mock/mock_controller.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller.h"
-#include "webrtc/test/gmock.h"
+#include "modules/audio_coding/audio_network_adaptor/controller.h"
+#include "test/gmock.h"
 
 namespace webrtc {
 
@@ -27,4 +27,4 @@ class MockController : public Controller {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_H_

modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
 
 #include <vector>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
-#include "webrtc/test/gmock.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "test/gmock.h"
 
 namespace webrtc {
 
@@ -30,4 +30,4 @@ class MockControllerManager : public ControllerManager {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_CONTROLLER_MANAGER_H_

modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
-#include "webrtc/test/gmock.h"
+#include "modules/audio_coding/audio_network_adaptor/debug_dump_writer.h"
+#include "test/gmock.h"
 
 namespace webrtc {
 
@@ -37,4 +37,4 @@ class MockDebugDumpWriter : public DebugDumpWriter {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_MOCK_MOCK_DEBUG_DUMP_WRITER_H_

modules/audio_coding/audio_network_adaptor/util/threshold_curve.h

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+#ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+#define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
 
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 
@@ -115,4 +115,4 @@ class ThresholdCurve {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_
+#endif  // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_UTIL_THRESHOLD_CURVE_H_

modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
+#include "test/gtest.h"
 
 // A threshold curve divides 2D space into three domains - below, on and above
 // the threshold curve.

modules/audio_coding/codecs/audio_decoder.h

@@ -12,9 +12,9 @@
 // webrtc/api/audio_codecs/audio_decoder.h instead!
 // TODO(kwiberg): Remove it.
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
 
-#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "api/audio_codecs/audio_decoder.h"
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_DECODER_H_

modules/audio_coding/codecs/audio_encoder.h

@@ -12,9 +12,9 @@
 // webrtc/api/audio_codecs/audio_encoder.h instead!
 // TODO(ossu): Remove it.
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder.h"
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_

modules/audio_coding/codecs/audio_format_conversion.cc

@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
 
 #include <string.h>
 
-#include "webrtc/api/array_view.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/sanitizer.h"
+#include "api/array_view.h"
+#include "api/optional.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
+#include "rtc_base/sanitizer.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/audio_format_conversion.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
+#define MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
 
-#include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/common_types.h"
+#include "api/audio_codecs/audio_format.h"
+#include "common_types.h"
 
 namespace webrtc {
 
@@ -21,4 +21,4 @@ CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format);
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_

modules/audio_coding/codecs/builtin_audio_decoder_factory.h

@@ -12,9 +12,9 @@
 // webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead!
 // TODO(kwiberg): Remove it.
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#define MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
 
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_

modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc

@@ -10,8 +10,8 @@
 
 #include <memory>
 
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/test/gtest.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/builtin_audio_encoder_factory.h

@@ -12,9 +12,9 @@
 // webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead!
 // TODO(ossu): Remove it.
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+#define MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
 
-#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_

modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc

@@ -12,9 +12,9 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/cng/audio_encoder_cng.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 
 #include <algorithm>
 #include <memory>

modules/audio_coding/codecs/cng/audio_encoder_cng.h

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
 
 #include <memory>
 #include <vector>
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/common_audio/vad/include/vad.h"
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "common_audio/vad/include/vad.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -93,4 +93,4 @@ class AudioEncoderCng final : public AudioEncoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_

modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc

@@ -11,11 +11,11 @@
 #include <memory>
 #include <vector>
 
-#include "webrtc/common_audio/vad/mock/mock_vad.h"
-#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_encoder.h"
+#include "common_audio/vad/mock/mock_vad.h"
+#include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
+#include "rtc_base/constructormagic.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
 
 using ::testing::Return;
 using ::testing::_;

modules/audio_coding/codecs/cng/cng_unittest.cc

@@ -10,9 +10,9 @@
 #include <memory>
 #include <string>
 
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/cng/webrtc_cng.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
 
 #include <algorithm>
 
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/safe_conversions.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/cng/webrtc_cng.h

@@ -9,14 +9,14 @@
  */
 
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
 
 #include <cstddef>
 
-#include "webrtc/api/array_view.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/typedefs.h"
+#include "api/array_view.h"
+#include "rtc_base/buffer.h"
+#include "typedefs.h"
 
 #define WEBRTC_CNG_MAX_LPC_ORDER 12
 
@@ -96,4 +96,4 @@ class ComfortNoiseEncoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_

modules/audio_coding/codecs/g711/audio_decoder_pcm.cc

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
 
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/g711/audio_decoder_pcm.h

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
 
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -67,4 +67,4 @@ class AudioDecoderPcmA final : public AudioDecoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_

modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 #include <algorithm>
 #include <limits>
 
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
-#include "webrtc/rtc_base/checks.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/g711/audio_encoder_pcm.h

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
 
 #include <vector>
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -121,4 +121,4 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_

modules/audio_coding/codecs/g711/g711.c

@@ -21,7 +21,7 @@
  */
 
 #include "g711.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 /* Copied from the CCITT G.711 specification */
 static const uint8_t ulaw_to_alaw_table[256] = {

modules/audio_coding/codecs/g711/g711.h

@@ -49,7 +49,7 @@ specification by other means.
 extern "C" {
 #endif
 
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 #if defined(__i386__)
 /*! \brief Find the bit position of the highest set bit in a word

modules/audio_coding/codecs/g711/g711_interface.c

@@ -10,7 +10,7 @@
 #include <string.h>
 #include "g711.h"
 #include "g711_interface.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 size_t WebRtcG711_EncodeA(const int16_t* speechIn,
                           size_t len,

modules/audio_coding/codecs/g711/g711_interface.h

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
 
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 // Comfort noise constants
 #define G711_WEBRTC_SPEECH 1
@@ -132,4 +132,4 @@ int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
 }
 #endif
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_

modules/audio_coding/codecs/g711/test/testG711.cc

@@ -17,7 +17,7 @@
 #include <string.h>
 
 /* include API */
-#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
 
 /* Runtime statistics */
 #include <time.h>

modules/audio_coding/codecs/g722/audio_decoder_g722.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
 
 #include <string.h>
 
-#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/g722/audio_decoder_g722.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
 
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/constructormagic.h"
 
 typedef struct WebRtcG722DecInst G722DecInst;
 
@@ -76,4 +76,4 @@ class AudioDecoderG722StereoImpl final : public AudioDecoder {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_

modules/audio_coding/codecs/g722/audio_encoder_g722.cc

@@ -8,15 +8,15 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
 
 #include <algorithm>
 
 #include <limits>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
 
 namespace webrtc {
 

modules/audio_coding/codecs/g722/audio_encoder_g722.h

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
 #include <memory>
 
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/g722/audio_encoder_g722_config.h"
-#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/g722/audio_encoder_g722_config.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/constructormagic.h"
 
 namespace webrtc {
 
@@ -65,4 +65,4 @@ class AudioEncoderG722Impl final : public AudioEncoder {
 };
 
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#endif  // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_

modules/audio_coding/codecs/g722/g722_decode.c

@@ -35,7 +35,7 @@
 #include <stdlib.h>
 
 #include "g722_enc_dec.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 #if !defined(FALSE)
 #define FALSE 0

modules/audio_coding/codecs/g722/g722_enc_dec.h

@@ -31,7 +31,7 @@
 #if !defined(_G722_ENC_DEC_H_)
 #define _G722_ENC_DEC_H_
 
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 /*! \page g722_page G.722 encoding and decoding
 \section g722_page_sec_1 What does it do?

modules/audio_coding/codecs/g722/g722_encode.c

@@ -35,7 +35,7 @@
 #include <stdlib.h>
 
 #include "g722_enc_dec.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 #if !defined(FALSE)
 #define FALSE 0

modules/audio_coding/codecs/g722/g722_interface.c

@@ -14,7 +14,7 @@
 #include <string.h>
 #include "g722_enc_dec.h"
 #include "g722_interface.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
 {

modules/audio_coding/codecs/g722/g722_interface.h

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
 
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 /*
  * Solution to support multiple instances
@@ -179,4 +179,4 @@ int16_t WebRtcG722_Version(char *versionStr, short len);
 #endif
 
 
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */

modules/audio_coding/codecs/g722/test/testG722.cc

@@ -15,10 +15,10 @@
 #include <stdio.h>
 #include <stdlib.h>
 #include <string.h>
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
 
 /* include API */
-#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
 
 /* Runtime statistics */
 #include <time.h>

modules/audio_coding/codecs/ilbc/abs_quant.h

@@ -16,8 +16,8 @@
 
 ******************************************************************/
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
 
 #include "defines.h"
 

modules/audio_coding/codecs/ilbc/abs_quant_loop.h

@@ -16,8 +16,8 @@
 
 ******************************************************************/
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
 
 #include "defines.h"
 

modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
 
 #include <utility>
 
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {