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/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index d415098..ffa88af 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -871,13 +871,13 @@
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 @@
} # 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 @@
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 @@
]
data = [
- "../../../resources/speech_and_misc_wb.pcm",
+ "../../resources/speech_and_misc_wb.pcm",
]
if (is_win) {
diff --git a/modules/audio_coding/DEPS b/modules/audio_coding/DEPS
index 3a745ff..3dc9624 100644
--- a/modules/audio_coding/DEPS
+++ b/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",
]
diff --git a/modules/audio_coding/acm2/acm_codec_database.cc b/modules/audio_coding/acm2/acm_codec_database.cc
index 7bd8ee7..7b3b1d2 100644
--- a/modules/audio_coding/acm2/acm_codec_database.cc
+++ b/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
diff --git a/modules/audio_coding/acm2/acm_codec_database.h b/modules/audio_coding/acm2/acm_codec_database.h
index d586cb9..74275a3 100644
--- a/modules/audio_coding/acm2/acm_codec_database.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
+#endif // MODULES_AUDIO_CODING_ACM2_ACM_CODEC_DATABASE_H_
diff --git a/modules/audio_coding/acm2/acm_receive_test.cc b/modules/audio_coding/acm2/acm_receive_test.cc
index 1eae859..6cfe464 100644
--- a/modules/audio_coding/acm2/acm_receive_test.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/acm_receive_test.h b/modules/audio_coding/acm2/acm_receive_test.h
index c7faa56..657a124 100644
--- a/modules/audio_coding/acm2/acm_receive_test.h
+++ b/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 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
+#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 8912eaa..f8393ed 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index 93ff89c..b8b017b 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
+#endif // MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
diff --git a/modules/audio_coding/acm2/acm_receiver_unittest.cc b/modules/audio_coding/acm2/acm_receiver_unittest.cc
index 4026c5b..8fbea84 100644
--- a/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/acm_resampler.cc b/modules/audio_coding/acm2/acm_resampler.cc
index 555263a..3cd7caa 100644
--- a/modules/audio_coding/acm2/acm_resampler.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/acm_resampler.h b/modules/audio_coding/acm2/acm_resampler.h
index 268db8b..462201f 100644
--- a/modules/audio_coding/acm2/acm_resampler.h
+++ b/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 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
+#endif // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
diff --git a/modules/audio_coding/acm2/acm_send_test.cc b/modules/audio_coding/acm2/acm_send_test.cc
index a88ba7d..d5f196b 100644
--- a/modules/audio_coding/acm2/acm_send_test.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/acm_send_test.h b/modules/audio_coding/acm2/acm_send_test.h
index b451e99..6aea0f1 100644
--- a/modules/audio_coding/acm2/acm_send_test.h
+++ b/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 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
+#endif // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index 0244882..1f108f5 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/modules/audio_coding/acm2/audio_coding_module_unittest.cc
index 9c7e4cd..80fc4d8 100644
--- a/modules/audio_coding/acm2/audio_coding_module_unittest.cc
+++ b/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;
diff --git a/modules/audio_coding/acm2/call_statistics.cc b/modules/audio_coding/acm2/call_statistics.cc
index 28511e9..a506ead 100644
--- a/modules/audio_coding/acm2/call_statistics.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/call_statistics.h b/modules/audio_coding/acm2/call_statistics.h
index 3993319..462afb5 100644
--- a/modules/audio_coding/acm2/call_statistics.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
+#endif // MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
diff --git a/modules/audio_coding/acm2/call_statistics_unittest.cc b/modules/audio_coding/acm2/call_statistics_unittest.cc
index ce4468a..77c3863 100644
--- a/modules/audio_coding/acm2/call_statistics_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/codec_manager.cc b/modules/audio_coding/acm2/codec_manager.cc
index 7bbafc7..63f0244 100644
--- a/modules/audio_coding/acm2/codec_manager.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/codec_manager.h b/modules/audio_coding/acm2/codec_manager.h
index b5c8d3e..00652cc 100644
--- a/modules/audio_coding/acm2/codec_manager.h
+++ b/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 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#endif // MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
diff --git a/modules/audio_coding/acm2/codec_manager_unittest.cc b/modules/audio_coding/acm2/codec_manager_unittest.cc
index 4dfcbda..e041b5e 100644
--- a/modules/audio_coding/acm2/codec_manager_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/rent_a_codec.cc b/modules/audio_coding/acm2/rent_a_codec.cc
index 3bc1464..82a1a8d 100644
--- a/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/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 {
diff --git a/modules/audio_coding/acm2/rent_a_codec.h b/modules/audio_coding/acm2/rent_a_codec.h
index 6dcf793..7dd2d6b 100644
--- a/modules/audio_coding/acm2/rent_a_codec.h
+++ b/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 @@
} // namespace acm2
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
+#endif // MODULES_AUDIO_CODING_ACM2_RENT_A_CODEC_H_
diff --git a/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/modules/audio_coding/acm2/rent_a_codec_unittest.cc
index dcf6383..9eded20 100644
--- a/modules/audio_coding/acm2/rent_a_codec_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
index e367e10..55b326d 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_config.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index e2786ca..54423e6 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 8e76db2..14000fe 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index a0dc12c..d297d27 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
index b585ce3..587c858 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
index 740edbd..cad6d6a 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
index 9fab781..9726992 100644
--- a/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.cc b/modules/audio_coding/audio_network_adaptor/channel_controller.cc
index 052808d..55a913a 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller.h b/modules/audio_coding/audio_network_adaptor/channel_controller.h
index 4920668..f53ddd6 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
+#endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CHANNEL_CONTROLLER_H_
diff --git a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
index 980292c..73160fe 100644
--- a/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/controller.cc b/modules/audio_coding/audio_network_adaptor/controller.cc
index 549352f..5e2dc85 100644
--- a/modules/audio_coding/audio_network_adaptor/controller.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/controller.h b/modules/audio_coding/audio_network_adaptor/controller.h
index 38c2c21..af2f569 100644
--- a/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
+#endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 425c213..319e752 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/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
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager.h b/modules/audio_coding/audio_network_adaptor/controller_manager.h
index 600fe65..5c63f2f 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
+#endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_MANAGER_H_
diff --git a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
index aa4ff36..846397a 100644
--- a/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
+++ b/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
diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index b3e8776..97de177 100644
--- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/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
diff --git a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
index 0d9206d..e40c832 100644
--- a/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
index 1941a5e..7c7d6ad 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/modules/audio_coding/audio_network_adaptor/dtx_controller.h
index 2945e11..8a2427e 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
+#endif // MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_DTX_CONTROLLER_H_
diff --git a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
index 73527ee..2c58249 100644
--- a/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
index b4fcbfd..a070b77 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer.h b/modules/audio_coding/audio_network_adaptor/event_log_writer.h
index 7f7acef..fca8e53 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
index 443e4d1..3853705 100644
--- a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
index 58644b9..80cc695 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
index 9259f30..c273537 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
index 1ae7b45..41d959d 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
index 0cf6d3b..eb56ea0 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
index ce04dc7..ade55ae 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc b/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
index a7b057c..f20122f 100644
--- a/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
index 4dd6088..9a5f03221 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
index 47a48fc..bbda77e 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
index d2b535c..fa0af7d 100644
--- a/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index a91b33b..7687446 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
index 148dec2..5b2d113 100644
--- a/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
index f58a482..15dc741 100644
--- a/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h b/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
index e856601..df28e9e 100644
--- a/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h b/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h
index 4976fd8..8d410a7 100644
--- a/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h b/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
index fba9ccc..06650ab 100644
--- a/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h b/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h
index 2f816db..0375386 100644
--- a/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h
+++ b/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 @@
} // 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_
diff --git a/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc b/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
index 0897d0d..0375e76 100644
--- a/modules/audio_coding/audio_network_adaptor/util/threshold_curve_unittest.cc
+++ b/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.
diff --git a/modules/audio_coding/codecs/audio_decoder.h b/modules/audio_coding/codecs/audio_decoder.h
index da06282..b7b15cd 100644
--- a/modules/audio_coding/codecs/audio_decoder.h
+++ b/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_
diff --git a/modules/audio_coding/codecs/audio_encoder.h b/modules/audio_coding/codecs/audio_encoder.h
index 942abb6..010ae67 100644
--- a/modules/audio_coding/codecs/audio_encoder.h
+++ b/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_
diff --git a/modules/audio_coding/codecs/audio_format_conversion.cc b/modules/audio_coding/codecs/audio_format_conversion.cc
index 1e53bf3..39b0963 100644
--- a/modules/audio_coding/codecs/audio_format_conversion.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/audio_format_conversion.h b/modules/audio_coding/codecs/audio_format_conversion.h
index 0fa3a50..23b45f6 100644
--- a/modules/audio_coding/codecs/audio_format_conversion.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
+#endif // MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory.h b/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
index 7e6407a..7494ac0 100644
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
+++ b/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_
diff --git a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
index f6b44b6..27b89b6 100644
--- a/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory.h b/modules/audio_coding/codecs/builtin_audio_encoder_factory.h
index fd0ed79..6ec765a 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory.h
+++ b/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_
diff --git a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
index 1446268..3955e4a 100644
--- a/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 52661d3..78148ab 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/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>
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 66305e6..4491289 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
+#endif // MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
diff --git a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 5de1b03..ef3ff31 100644
--- a/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/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::_;
diff --git a/modules/audio_coding/codecs/cng/cng_unittest.cc b/modules/audio_coding/codecs/cng/cng_unittest.cc
index 3835e93..54e5189 100644
--- a/modules/audio_coding/codecs/cng/cng_unittest.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.cc b/modules/audio_coding/codecs/cng/webrtc_cng.cc
index b891d84..e2a3347 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/cng/webrtc_cng.h b/modules/audio_coding/codecs/cng/webrtc_cng.h
index 98231e7..7653fbd 100644
--- a/modules/audio_coding/codecs/cng/webrtc_cng.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
+#endif // MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index fd285a7..a620a3e 100644
--- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index f72f202..29e4fa6 100644
--- a/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
+#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_DECODER_PCM_H_
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
index 711eed7..b4384c5 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index 22a15a1..37b67cf 100644
--- a/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
+#endif // MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_
diff --git a/modules/audio_coding/codecs/g711/g711.c b/modules/audio_coding/codecs/g711/g711.c
index a183757..ac9e44a 100644
--- a/modules/audio_coding/codecs/g711/g711.c
+++ b/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] = {
diff --git a/modules/audio_coding/codecs/g711/g711.h b/modules/audio_coding/codecs/g711/g711.h
index 3b07d8b..f34d266 100644
--- a/modules/audio_coding/codecs/g711/g711.h
+++ b/modules/audio_coding/codecs/g711/g711.h
@@ -49,7 +49,7 @@
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
diff --git a/modules/audio_coding/codecs/g711/g711_interface.c b/modules/audio_coding/codecs/g711/g711_interface.c
index 5b96a9c..52f73fb 100644
--- a/modules/audio_coding/codecs/g711/g711_interface.c
+++ b/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,
diff --git a/modules/audio_coding/codecs/g711/g711_interface.h b/modules/audio_coding/codecs/g711/g711_interface.h
index 00854bb..bfaac3a 100644
--- a/modules/audio_coding/codecs/g711/g711_interface.h
+++ b/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 @@
}
#endif
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
+#endif // MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
diff --git a/modules/audio_coding/codecs/g711/test/testG711.cc b/modules/audio_coding/codecs/g711/test/testG711.cc
index 5675b1f..98f3925 100644
--- a/modules/audio_coding/codecs/g711/test/testG711.cc
+++ b/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>
diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
index 7e597f8..ea4a721 100644
--- a/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index ee2270f..3240448 100644
--- a/modules/audio_coding/codecs/g722/audio_decoder_g722.h
+++ b/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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
+#endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_DECODER_G722_H_
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 4c3e82d..1d0ca93 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index b4d9ef0..1f4b943 100644
--- a/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/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 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
+#endif // MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
diff --git a/modules/audio_coding/codecs/g722/g722_decode.c b/modules/audio_coding/codecs/g722/g722_decode.c
index 71646c5..28740df 100644
--- a/modules/audio_coding/codecs/g722/g722_decode.c
+++ b/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
diff --git a/modules/audio_coding/codecs/g722/g722_enc_dec.h b/modules/audio_coding/codecs/g722/g722_enc_dec.h
index 7db4895..3074aed 100644
--- a/modules/audio_coding/codecs/g722/g722_enc_dec.h
+++ b/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?
diff --git a/modules/audio_coding/codecs/g722/g722_encode.c b/modules/audio_coding/codecs/g722/g722_encode.c
index 2bd784b..1771319 100644
--- a/modules/audio_coding/codecs/g722/g722_encode.c
+++ b/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
diff --git a/modules/audio_coding/codecs/g722/g722_interface.c b/modules/audio_coding/codecs/g722/g722_interface.c
index 4244d5c..3c869d6 100644
--- a/modules/audio_coding/codecs/g722/g722_interface.c
+++ b/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)
{
diff --git a/modules/audio_coding/codecs/g722/g722_interface.h b/modules/audio_coding/codecs/g722/g722_interface.h
index b411ef0..74930cc 100644
--- a/modules/audio_coding/codecs/g722/g722_interface.h
+++ b/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 @@
#endif
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
diff --git a/modules/audio_coding/codecs/g722/test/testG722.cc b/modules/audio_coding/codecs/g722/test/testG722.cc
index c55a2eb..925e8af 100644
--- a/modules/audio_coding/codecs/g722/test/testG722.cc
+++ b/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>
diff --git a/modules/audio_coding/codecs/ilbc/abs_quant.h b/modules/audio_coding/codecs/ilbc/abs_quant.h
index 5154534..090ed73 100644
--- a/modules/audio_coding/codecs/ilbc/abs_quant.h
+++ b/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"
diff --git a/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
index c8bf675..92f6fff 100644
--- a/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
+++ b/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"
diff --git a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
index 61ec7bd..153b742 100644
--- a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
+++ b/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 {
diff --git a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
index f2320fa..edb65d0 100644
--- a/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_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 iLBC_decinst_t_ IlbcDecoderInstance;
@@ -43,4 +43,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_DECODER_ILBC_H_
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
index 2a6dda7..f6d90c8 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_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_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
#include <algorithm>
#include <limits>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 6a80a69..a238689 100644
--- a/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -51,4 +51,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
diff --git a/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
index c5c4088..e8eb069 100644
--- a/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ b/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/bw_expand.h b/modules/audio_coding/codecs/ilbc/bw_expand.h
index b3b16d5..6268baa 100644
--- a/modules/audio_coding/codecs/ilbc/bw_expand.h
+++ b/modules/audio_coding/codecs/ilbc/bw_expand.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/cb_construct.h b/modules/audio_coding/codecs/ilbc/cb_construct.h
index 12df628..aeb00da 100644
--- a/modules/audio_coding/codecs/ilbc/cb_construct.h
+++ b/modules/audio_coding/codecs/ilbc/cb_construct.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
#include <stdbool.h>
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
index 6da2f43..e8e2fe9 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
void WebRtcIlbcfix_CbMemEnergy(
size_t range,
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
index 594ba5f..00eb017 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
void WebRtcIlbcfix_CbMemEnergyAugmentation(
int16_t *interpSamples, /* (i) The interpolated samples */
diff --git a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
index 2991869..af8e658 100644
--- a/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
+++ b/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
void WebRtcIlbcfix_CbMemEnergyCalc(
int32_t energy, /* (i) input start energy */
diff --git a/modules/audio_coding/codecs/ilbc/cb_search.h b/modules/audio_coding/codecs/ilbc/cb_search.h
index ed1580c..c8626c5 100644
--- a/modules/audio_coding/codecs/ilbc/cb_search.h
+++ b/modules/audio_coding/codecs/ilbc/cb_search.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
void WebRtcIlbcfix_CbSearch(
IlbcEncoder *iLBCenc_inst,
diff --git a/modules/audio_coding/codecs/ilbc/cb_search_core.h b/modules/audio_coding/codecs/ilbc/cb_search_core.h
index 9648cf2..1db7f70 100644
--- a/modules/audio_coding/codecs/ilbc/cb_search_core.h
+++ b/modules/audio_coding/codecs/ilbc/cb_search_core.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
index a20fa38..4929b64 100644
--- a/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
+++ b/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/chebyshev.h b/modules/audio_coding/codecs/ilbc/chebyshev.h
index bf10132..f990d0b 100644
--- a/modules/audio_coding/codecs/ilbc/chebyshev.h
+++ b/modules/audio_coding/codecs/ilbc/chebyshev.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/comp_corr.h b/modules/audio_coding/codecs/ilbc/comp_corr.h
index ab78c72..6f6c5ec 100644
--- a/modules/audio_coding/codecs/ilbc/comp_corr.h
+++ b/modules/audio_coding/codecs/ilbc/comp_corr.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/constants.h b/modules/audio_coding/codecs/ilbc/constants.h
index 7c4ad4d..9e12227 100644
--- a/modules/audio_coding/codecs/ilbc/constants.h
+++ b/modules/audio_coding/codecs/ilbc/constants.h
@@ -16,11 +16,11 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
#include "defines.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/* high pass filters */
diff --git a/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
index e05b88b..adc428d 100644
--- a/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
+++ b/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
@@ -18,8 +18,8 @@
#include "defines.h"
#include "constants.h"
-#include "webrtc/rtc_base/sanitizer.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/sanitizer.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
/*----------------------------------------------------------------*
* Recreate a specific codebook vector from the augmented part.
diff --git a/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
index f2582cd..0616e2e 100644
--- a/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
+++ b/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/decode.h b/modules/audio_coding/codecs/ilbc/decode.h
index 2d05182..b35be96 100644
--- a/modules/audio_coding/codecs/ilbc/decode.h
+++ b/modules/audio_coding/codecs/ilbc/decode.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/decode_residual.h b/modules/audio_coding/codecs/ilbc/decode_residual.h
index e3fb7f7..048671e 100644
--- a/modules/audio_coding/codecs/ilbc/decode_residual.h
+++ b/modules/audio_coding/codecs/ilbc/decode_residual.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
#include <stdbool.h>
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
index 37ecf07..d83ad47 100644
--- a/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/defines.h b/modules/audio_coding/codecs/ilbc/defines.h
index 2faaea1..3abac27 100644
--- a/modules/audio_coding/codecs/ilbc/defines.h
+++ b/modules/audio_coding/codecs/ilbc/defines.h
@@ -15,12 +15,12 @@
define.h
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
#include <string.h>
#include "signal_processing_library.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/* general codec settings */
diff --git a/modules/audio_coding/codecs/ilbc/do_plc.h b/modules/audio_coding/codecs/ilbc/do_plc.h
index 38b8fdb..76a598a 100644
--- a/modules/audio_coding/codecs/ilbc/do_plc.h
+++ b/modules/audio_coding/codecs/ilbc/do_plc.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/encode.c b/modules/audio_coding/codecs/ilbc/encode.c
index fa4ffd0..7254355 100644
--- a/modules/audio_coding/codecs/ilbc/encode.c
+++ b/modules/audio_coding/codecs/ilbc/encode.c
@@ -29,7 +29,7 @@
#include "index_conv_enc.h"
#include "pack_bits.h"
#include "hp_input.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
#ifdef SPLIT_10MS
#include "unpack_bits.h"
diff --git a/modules/audio_coding/codecs/ilbc/encode.h b/modules/audio_coding/codecs/ilbc/encode.h
index fe6ae62..0692a34 100644
--- a/modules/audio_coding/codecs/ilbc/encode.h
+++ b/modules/audio_coding/codecs/ilbc/encode.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/energy_inverse.h b/modules/audio_coding/codecs/ilbc/energy_inverse.h
index fe25094..cf5f88b 100644
--- a/modules/audio_coding/codecs/ilbc/energy_inverse.h
+++ b/modules/audio_coding/codecs/ilbc/energy_inverse.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/enh_upsample.h b/modules/audio_coding/codecs/ilbc/enh_upsample.h
index 00bb28b..62f6923 100644
--- a/modules/audio_coding/codecs/ilbc/enh_upsample.h
+++ b/modules/audio_coding/codecs/ilbc/enh_upsample.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/enhancer.h b/modules/audio_coding/codecs/ilbc/enhancer.h
index ed219fb..5b4a30b 100644
--- a/modules/audio_coding/codecs/ilbc/enhancer.h
+++ b/modules/audio_coding/codecs/ilbc/enhancer.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/modules/audio_coding/codecs/ilbc/enhancer_interface.h
index d540533..17a0354 100644
--- a/modules/audio_coding/codecs/ilbc/enhancer_interface.h
+++ b/modules/audio_coding/codecs/ilbc/enhancer_interface.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
index d23b25c..947c462 100644
--- a/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
+++ b/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/frame_classify.h b/modules/audio_coding/codecs/ilbc/frame_classify.h
index 99f7144..60b3249 100644
--- a/modules/audio_coding/codecs/ilbc/frame_classify.h
+++ b/modules/audio_coding/codecs/ilbc/frame_classify.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
size_t WebRtcIlbcfix_FrameClassify(
/* (o) Index to the max-energy sub frame */
diff --git a/modules/audio_coding/codecs/ilbc/gain_dequant.h b/modules/audio_coding/codecs/ilbc/gain_dequant.h
index 6bda066..efa3c7a 100644
--- a/modules/audio_coding/codecs/ilbc/gain_dequant.h
+++ b/modules/audio_coding/codecs/ilbc/gain_dequant.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/gain_quant.h b/modules/audio_coding/codecs/ilbc/gain_quant.h
index 3954364..5780bee 100644
--- a/modules/audio_coding/codecs/ilbc/gain_quant.h
+++ b/modules/audio_coding/codecs/ilbc/gain_quant.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/modules/audio_coding/codecs/ilbc/get_cd_vec.h
index a5abb5e..f16789e 100644
--- a/modules/audio_coding/codecs/ilbc/get_cd_vec.h
+++ b/modules/audio_coding/codecs/ilbc/get_cd_vec.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
#include <stdbool.h>
diff --git a/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
index 46ade48..537b771 100644
--- a/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
+++ b/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/modules/audio_coding/codecs/ilbc/get_sync_seq.h
index 0e3b207..9e97e60 100644
--- a/modules/audio_coding/codecs/ilbc/get_sync_seq.h
+++ b/modules/audio_coding/codecs/ilbc/get_sync_seq.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/hp_input.h b/modules/audio_coding/codecs/ilbc/hp_input.h
index acdfa91..fdc0c37 100644
--- a/modules/audio_coding/codecs/ilbc/hp_input.h
+++ b/modules/audio_coding/codecs/ilbc/hp_input.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/hp_output.h b/modules/audio_coding/codecs/ilbc/hp_output.h
index 1840b68..6f6a56d 100644
--- a/modules/audio_coding/codecs/ilbc/hp_output.h
+++ b/modules/audio_coding/codecs/ilbc/hp_output.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/ilbc.c b/modules/audio_coding/codecs/ilbc/ilbc.c
index 18e0f84..08677d9 100644
--- a/modules/audio_coding/codecs/ilbc/ilbc.c
+++ b/modules/audio_coding/codecs/ilbc/ilbc.c
@@ -22,7 +22,7 @@
#include "encode.h"
#include "init_decode.h"
#include "decode.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
#include <stdlib.h>
int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
diff --git a/modules/audio_coding/codecs/ilbc/ilbc.h b/modules/audio_coding/codecs/ilbc/ilbc.h
index c021f5b..aa007a9 100644
--- a/modules/audio_coding/codecs/ilbc/ilbc.h
+++ b/modules/audio_coding/codecs/ilbc/ilbc.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
#include <stddef.h>
@@ -24,7 +24,7 @@
* Define the fixpoint numeric formats
*/
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/*
* Solution to support multiple instances
@@ -255,4 +255,4 @@
}
#endif
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
diff --git a/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
index 69e2181..b8d3c7c 100644
--- a/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
+++ b/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
@@ -8,10 +8,10 @@
* 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 "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/modules/audio_coding/codecs/ilbc/index_conv_dec.h
index 354c5b8..4b61d31 100644
--- a/modules/audio_coding/codecs/ilbc/index_conv_dec.h
+++ b/modules/audio_coding/codecs/ilbc/index_conv_dec.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/modules/audio_coding/codecs/ilbc/index_conv_enc.h
index d686331..935928a 100644
--- a/modules/audio_coding/codecs/ilbc/index_conv_enc.h
+++ b/modules/audio_coding/codecs/ilbc/index_conv_enc.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/init_decode.h b/modules/audio_coding/codecs/ilbc/init_decode.h
index cdd2192..6f3a17b 100644
--- a/modules/audio_coding/codecs/ilbc/init_decode.h
+++ b/modules/audio_coding/codecs/ilbc/init_decode.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/init_encode.h b/modules/audio_coding/codecs/ilbc/init_encode.h
index 7154661..fc11cc7 100644
--- a/modules/audio_coding/codecs/ilbc/init_encode.h
+++ b/modules/audio_coding/codecs/ilbc/init_encode.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/interpolate.h b/modules/audio_coding/codecs/ilbc/interpolate.h
index 0483232..befe241 100644
--- a/modules/audio_coding/codecs/ilbc/interpolate.h
+++ b/modules/audio_coding/codecs/ilbc/interpolate.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/modules/audio_coding/codecs/ilbc/interpolate_samples.h
index 7549d2c..8a28273 100644
--- a/modules/audio_coding/codecs/ilbc/interpolate_samples.h
+++ b/modules/audio_coding/codecs/ilbc/interpolate_samples.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lpc_encode.h b/modules/audio_coding/codecs/ilbc/lpc_encode.h
index 9f6f504..776b3a1 100644
--- a/modules/audio_coding/codecs/ilbc/lpc_encode.h
+++ b/modules/audio_coding/codecs/ilbc/lpc_encode.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lsf_check.h b/modules/audio_coding/codecs/ilbc/lsf_check.h
index 2f4ac8c..bfcea08 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_check.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_check.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
index 3540c1c..4530528 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
index 799c100..a2d2db6 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
index b2104d7..ace29de 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
index d85f510..782ef9f 100644
--- a/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
+++ b/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
index a2bcaff..1173985 100644
--- a/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/my_corr.h b/modules/audio_coding/codecs/ilbc/my_corr.h
index 2149464..1d0ac85 100644
--- a/modules/audio_coding/codecs/ilbc/my_corr.h
+++ b/modules/audio_coding/codecs/ilbc/my_corr.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
index 7d7fb6f..8ed52bc 100644
--- a/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
+++ b/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/pack_bits.h b/modules/audio_coding/codecs/ilbc/pack_bits.h
index 603ddd4..66df7fc 100644
--- a/modules/audio_coding/codecs/ilbc/pack_bits.h
+++ b/modules/audio_coding/codecs/ilbc/pack_bits.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
index 5a7f7bb..3f2a2a0 100644
--- a/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
index ed20fd9..cc27507 100644
--- a/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
+++ b/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/refiner.h b/modules/audio_coding/codecs/ilbc/refiner.h
index f8a2abc..92b81be 100644
--- a/modules/audio_coding/codecs/ilbc/refiner.h
+++ b/modules/audio_coding/codecs/ilbc/refiner.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
index 6abcc00..ccf66d3 100644
--- a/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
+++ b/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
index 7808da1..eb5b803 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
+++ b/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
index 353edb2..660f8fe 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
+++ b/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
index 94f804b..6dc5719 100644
--- a/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
+++ b/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/smooth.h b/modules/audio_coding/codecs/ilbc/smooth.h
index add0c7b..3515432 100644
--- a/modules/audio_coding/codecs/ilbc/smooth.h
+++ b/modules/audio_coding/codecs/ilbc/smooth.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/modules/audio_coding/codecs/ilbc/smooth_out_data.h
index 8324439..7ed1a05 100644
--- a/modules/audio_coding/codecs/ilbc/smooth_out_data.h
+++ b/modules/audio_coding/codecs/ilbc/smooth_out_data.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/sort_sq.h b/modules/audio_coding/codecs/ilbc/sort_sq.h
index eaf175b..436e118 100644
--- a/modules/audio_coding/codecs/ilbc/sort_sq.h
+++ b/modules/audio_coding/codecs/ilbc/sort_sq.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/split_vq.h b/modules/audio_coding/codecs/ilbc/split_vq.h
index 2ca98cb..3b303bd 100644
--- a/modules/audio_coding/codecs/ilbc/split_vq.h
+++ b/modules/audio_coding/codecs/ilbc/split_vq.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/state_construct.h b/modules/audio_coding/codecs/ilbc/state_construct.h
index 2631919..9339f65 100644
--- a/modules/audio_coding/codecs/ilbc/state_construct.h
+++ b/modules/audio_coding/codecs/ilbc/state_construct.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
/*----------------------------------------------------------------*
* Generate the start state from the quantized indexes
diff --git a/modules/audio_coding/codecs/ilbc/state_search.h b/modules/audio_coding/codecs/ilbc/state_search.h
index 800beac..94a9d09 100644
--- a/modules/audio_coding/codecs/ilbc/state_search.h
+++ b/modules/audio_coding/codecs/ilbc/state_search.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/swap_bytes.h b/modules/audio_coding/codecs/ilbc/swap_bytes.h
index a4484d6..9518299 100644
--- a/modules/audio_coding/codecs/ilbc/swap_bytes.h
+++ b/modules/audio_coding/codecs/ilbc/swap_bytes.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
index b440c7a..4dbc185 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
+++ b/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
@@ -19,7 +19,7 @@
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
/*---------------------------------------------------------------*
* Main program to test iLBC encoding and decoding
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
index 7ffa4a7..132f3bd 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ b/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
@@ -21,7 +21,7 @@
#include <stdio.h>
#include <string.h>
#include <time.h>
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
//#define JUNK_DATA
#ifdef JUNK_DATA
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
index 5454948..a62a42e 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
+++ b/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
@@ -21,13 +21,13 @@
#include <stdio.h>
#include <string.h>
-#include "webrtc/modules/audio_coding/codecs/ilbc/defines.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/nit_encode.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/encode.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/init_decode.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/decode.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/constants.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/defines.h"
+#include "modules/audio_coding/codecs/ilbc/nit_encode.h"
+#include "modules/audio_coding/codecs/ilbc/encode.h"
+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
+#include "modules/audio_coding/codecs/ilbc/decode.h"
+#include "modules/audio_coding/codecs/ilbc/constants.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
diff --git a/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt b/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
index db0e9a0..99c6092 100644
--- a/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
+++ b/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
@@ -5,7 +5,7 @@
# This script can be used to verify the bit exactness of iLBC fixed-point version 1.0.6
#
-INP=../../../../../../../resources/audio_coding
+INP=../../../../../../resources/audio_coding
EXEP=../../../../../../../out/Release
OUTP=./GeneratedFiles
mkdir ./GeneratedFiles
diff --git a/modules/audio_coding/codecs/ilbc/unpack_bits.h b/modules/audio_coding/codecs/ilbc/unpack_bits.h
index 9586a12..01cf398 100644
--- a/modules/audio_coding/codecs/ilbc/unpack_bits.h
+++ b/modules/audio_coding/codecs/ilbc/unpack_bits.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/vq3.h b/modules/audio_coding/codecs/ilbc/vq3.h
index b146ea3..9aab03d 100644
--- a/modules/audio_coding/codecs/ilbc/vq3.h
+++ b/modules/audio_coding/codecs/ilbc/vq3.h
@@ -16,10 +16,10 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/*----------------------------------------------------------------*
* Vector quantization of order 3 (based on MSE)
diff --git a/modules/audio_coding/codecs/ilbc/vq4.h b/modules/audio_coding/codecs/ilbc/vq4.h
index 9263759..cc27b0c 100644
--- a/modules/audio_coding/codecs/ilbc/vq4.h
+++ b/modules/audio_coding/codecs/ilbc/vq4.h
@@ -16,10 +16,10 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/*----------------------------------------------------------------*
* Vector quantization of order 4 (based on MSE)
diff --git a/modules/audio_coding/codecs/ilbc/window32_w32.h b/modules/audio_coding/codecs/ilbc/window32_w32.h
index 27ed1b6..f53592e 100644
--- a/modules/audio_coding/codecs/ilbc/window32_w32.h
+++ b/modules/audio_coding/codecs/ilbc/window32_w32.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/modules/audio_coding/codecs/ilbc/xcorr_coef.h
index 9b81c0f..e2d511a 100644
--- a/modules/audio_coding/codecs/ilbc/xcorr_coef.h
+++ b/modules/audio_coding/codecs/ilbc/xcorr_coef.h
@@ -16,8 +16,8 @@
******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
+#define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
#include "defines.h"
diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
index 66f021a..eda1cfa 100644
--- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
#include <vector>
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/optional.h"
+#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@@ -56,4 +56,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_H_
diff --git a/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
index 1abfe37..90f3e8d 100644
--- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
@@ -106,4 +106,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index c12d734..541b90c 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
#include <vector>
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@@ -95,4 +95,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
diff --git a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index 854f2ee..de41ace 100644
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/checks.h"
+#include "common_types.h"
+#include "rtc_base/checks.h"
namespace webrtc {
@@ -186,4 +186,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
diff --git a/modules/audio_coding/codecs/isac/bandwidth_info.h b/modules/audio_coding/codecs/isac/bandwidth_info.h
index 1e3f4c9..cb565e0 100644
--- a/modules/audio_coding/codecs/isac/bandwidth_info.h
+++ b/modules/audio_coding/codecs/isac/bandwidth_info.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
typedef struct {
int in_use;
@@ -21,4 +21,4 @@
int16_t jitter_info;
} IsacBandwidthInfo;
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
diff --git a/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h b/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h
index 4ddc3bb..0b4eadd 100644
--- a/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h
+++ b/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
-#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
+#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
namespace webrtc {
using AudioDecoderIsacFixImpl = AudioDecoderIsacT<IsacFix>;
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_DECODER_ISACFIX_H_
diff --git a/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h b/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h
index aefad78..f0cc038 100644
--- a/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h
+++ b/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
-#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
+#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
+#include "modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h"
namespace webrtc {
using AudioEncoderIsacFixImpl = AudioEncoderIsacT<IsacFix>;
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_AUDIO_ENCODER_ISACFIX_H_
diff --git a/modules/audio_coding/codecs/isac/fix/include/isacfix.h b/modules/audio_coding/codecs/isac/fix/include/isacfix.h
index 7f277ca..74df442 100644
--- a/modules/audio_coding/codecs/isac/fix/include/isacfix.h
+++ b/modules/audio_coding/codecs/isac/fix/include/isacfix.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_
#include <stddef.h>
-#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "typedefs.h"
typedef struct {
void *dummy;
@@ -634,4 +634,4 @@
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
index c76bf1a..830c52f 100644
--- a/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
+++ b/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
#include "structs.h"
@@ -157,4 +157,4 @@
const uint16_t *initIndex,
const int16_t lenData);
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc b/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
index 45eefb9..21259ee 100644
--- a/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/audio_decoder_isacfix.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
+#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc b/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
index 257a8b5..0190ab9 100644
--- a/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/audio_encoder_isacfix.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
+#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index ab5aa0a..d926f47 100644
--- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@@ -21,7 +21,7 @@
#include "bandwidth_estimator.h"
#include "settings.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
/* array of quantization levels for bottle neck info; Matlab code: */
/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
diff --git a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
index 101ef62..f80a587 100644
--- a/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
+++ b/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
#include "structs.h"
@@ -132,4 +132,4 @@
int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples);
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/codec.h b/modules/audio_coding/codecs/isac/fix/source/codec.h
index 001a04f..88d4055 100644
--- a/modules/audio_coding/codecs/isac/fix/source/codec.h
+++ b/modules/audio_coding/codecs/isac/fix/source/codec.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
#include "structs.h"
@@ -225,4 +225,4 @@
} // extern "C"
#endif
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/encode.c b/modules/audio_coding/codecs/isac/fix/source/encode.c
index ee07660..ef3e320 100644
--- a/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -15,20 +15,20 @@
*
*/
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "rtc_base/checks.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
#include <stdio.h>
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
+#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
+#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
+#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
+#include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h"
+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
int WebRtcIsacfix_EncodeImpl(int16_t *in,
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 0a6e26f..ed816c0 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -26,7 +26,7 @@
#include "lpc_tables.h"
#include "settings.h"
#include "signal_processing_library.h"
-#include "webrtc/rtc_base/sanitizer.h"
+#include "rtc_base/sanitizer.h"
/*
* Eenumerations for arguments to functions WebRtcIsacfix_MatrixProduct1()
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
index 1b87d0e..31f044d 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
#include "structs.h"
@@ -186,4 +186,4 @@
const int matrix0_index_step);
#endif
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c
index 599f8f0..a66a43e 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding_mips.c
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
// MIPS optimization of the function WebRtcIsacfix_MatrixProduct1.
// Bit-exact with the function WebRtcIsacfix_MatrixProduct1C from
diff --git a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c b/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
index 20437e5..2eccdfa 100644
--- a/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
+++ b/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
@@ -20,7 +20,7 @@
#include <stddef.h>
#include "signal_processing_library.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
const int32_t matrix1[],
diff --git a/modules/audio_coding/codecs/isac/fix/source/fft.h b/modules/audio_coding/codecs/isac/fix/source/fft.h
index dc7cea8..83062ea 100644
--- a/modules/audio_coding/codecs/isac/fix/source/fft.h
+++ b/modules/audio_coding/codecs/isac/fix/source/fft.h
@@ -27,8 +27,8 @@
* See the comments in the code for correct usage!
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
#include "structs.h"
@@ -38,4 +38,4 @@
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
index d488339..0a6b4a4 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h b/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h
index c96fb05..66f285a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h
@@ -16,10 +16,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
@@ -49,4 +49,4 @@
}
#endif
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
index bc21617..b4583e9 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
@@ -23,7 +23,7 @@
#include "codec.h"
#include "filterbank_tables.h"
#include "settings.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
// Declare a function pointer.
AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16;
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c
index 4dd70cf..949bca7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks_mips.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
// WebRtcIsacfix_AllpassFilter2FixDec16 function optimized for MIPSDSP platform.
// Bit-exact with WebRtcIsacfix_AllpassFilter2FixDec16C from filterbanks.c.
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c b/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
index cd0bb07..fd29ccb 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
@@ -15,7 +15,7 @@
#include <arm_neon.h>
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
int16_t* data_ch1, // Input and output in channel 1, in Q0
diff --git a/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
index c8e6ac8..4a42b07 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/rtc_base/sanitizer.h"
-#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "rtc_base/sanitizer.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+#include "typedefs.h"
class FilterBanksTest : public testing::Test {
protected:
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters.c b/modules/audio_coding/codecs/isac/fix/source/filters.c
index 31d4e08..85860f7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filters.c
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "rtc_base/checks.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
// Autocorrelation function in fixed point.
// NOTE! Different from SPLIB-version in how it scales the signal.
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters_mips.c b/modules/audio_coding/codecs/isac/fix/source/filters_mips.c
index 056dc27..ded3d03 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filters_mips.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
// MIPS optimized implementation of the Autocorrelation function in fixed point.
// NOTE! Different from SPLIB-version in how it scales the signal.
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters_neon.c b/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
index 9ec2b13..1734a96 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
+++ b/modules/audio_coding/codecs/isac/fix/source/filters_neon.c
@@ -10,8 +10,8 @@
#include <arm_neon.h>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "rtc_base/checks.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
// Autocorrelation function in fixed point.
// NOTE! Different from SPLIB-version in how it scales the signal.
diff --git a/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
index 527841a..6424e38 100644
--- a/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+#include "typedefs.h"
class FiltersTest : public testing::Test {
protected:
diff --git a/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h b/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
index a39444e..5f0f822 100644
--- a/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
+++ b/modules/audio_coding/codecs/isac/fix/source/isac_fix_type.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#include "rtc_base/checks.h"
namespace webrtc {
@@ -120,4 +120,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ISAC_FIX_TYPE_H_
diff --git a/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index 3377bf7..bbe9098 100644
--- a/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -15,18 +15,18 @@
*
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#include <stdlib.h>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
-#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
+#include "rtc_base/checks.h"
+#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
+#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
// Declare function pointers.
FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix;
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice.c b/modules/audio_coding/codecs/isac/fix/source/lattice.c
index 1f8b2c2..8136999 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice.c
@@ -17,7 +17,7 @@
#include "codec.h"
#include "settings.h"
-#include "webrtc/rtc_base/sanitizer.h"
+#include "rtc_base/sanitizer.h"
#define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32) \
((int32_t)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S b/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
index 36411df..e2d4318 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
@@ -25,7 +25,7 @@
@ r12: constant #16384
@ r6, r7, r8, r10, r11: scratch
-#include "webrtc/system_wrappers/include/asm_defines.h"
+#include "system_wrappers/include/asm_defines.h"
#include "settings.h"
GLOBAL_FUNCTION WebRtcIsacfix_FilterArLoop
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_c.c b/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
index 40c3bf8..8f3cc53 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
@@ -16,7 +16,7 @@
#include "settings.h"
#include "signal_processing_library.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/* Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients
* cth_Q15[] and sth_Q15[].
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c b/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
index d488bfc..d5805b7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
@@ -10,8 +10,8 @@
#include <stddef.h>
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "typedefs.h"
// Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients
// cth_Q15[] and sth_Q15[].
diff --git a/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c b/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c
index 9218a3a..8ea9b63 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lattice_neon.c
@@ -10,8 +10,8 @@
#include <arm_neon.h>
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
// Contains a function for the core loop in the normalized lattice MA
// filter routine for iSAC codec, optimized for ARM Neon platform.
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
index aac9275..97d291a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
#ifdef __cplusplus
extern "C" {
@@ -66,4 +66,4 @@
} /* extern "C" */
#endif
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c
index 55602b9..727008d 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_mips.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
+#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
// MIPS DSPR2 optimization for function WebRtcIsacfix_CalculateResidualEnergy
// Bit-exact with WebRtcIsacfix_CalculateResidualEnergyC from file
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
index a81814d..3d85866 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
-#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
+#include "typedefs.h"
class LpcMaskingModelTest : public testing::Test {
protected:
diff --git a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
index 7e8121e..7dfd5c7 100644
--- a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
@@ -15,10 +15,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/* indices of KLT coefficients used */
extern const uint16_t WebRtcIsacfix_kSelIndGain[12];
@@ -94,4 +94,4 @@
extern const int32_t WebRtcIsacfix_kMeansShapeQ17[3][108];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
index ac8511e..78cb93f 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.c
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/rtc_base/compile_assert_c.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/compile_assert_c.h"
/* log2[0.2, 0.5, 0.98] in Q8 */
static const int16_t kLogLagWinQ8[3] = {
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
index 40f15c4..0aa727d 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
#include "structs.h"
@@ -62,4 +62,4 @@
void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8);
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
index 81ec1d6..1214e23 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_c.c
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
#ifdef WEBRTC_HAS_NEON
#include <arm_neon.h>
#endif
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/rtc_base/compile_assert_c.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/compile_assert_c.h"
extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x);
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
index a76ed7d..4ead84c 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_estimator_mips.c
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/rtc_base/compile_assert_c.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/compile_assert_c.h"
extern int32_t WebRtcIsacfix_Log2Q8(uint32_t x);
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
index bcd6038..e565e85 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h"
-#include "webrtc/rtc_base/compile_assert_c.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
+#include "rtc_base/compile_assert_c.h"
// Number of segments in a pitch subframe.
static const int kSegments = 5;
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
index bc18d44..44fcac8 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
@@ -13,7 +13,7 @@
@
@ Output is bit-exact with the reference C code in pitch_filter.c.
-#include "webrtc/system_wrappers/include/asm_defines.h"
+#include "system_wrappers/include/asm_defines.h"
#include "settings.h"
GLOBAL_FUNCTION WebRtcIsacfix_PitchFilterCore
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
index 366eef0..f23d19d 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_c.c
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
/* Filter coefficicients in Q15. */
static const int16_t kDampFilter[PITCH_DAMPORDER] = {
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c
index 0f390b8..785fd94 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_filter_mips.c
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
void WebRtcIsacfix_PitchFilterCore(int loopNumber,
int16_t gain,
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
index 4aab2b6..fbc797c 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
@@ -15,10 +15,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/********************* Pitch Filter Gain Coefficient Tables ************************/
/* cdf for quantized pitch filter gains */
@@ -41,4 +41,4 @@
/* transform matrix */
extern const int16_t WebRtcIsacfix_kTransform[4][4];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
index a5478b2..ffe3cd8 100644
--- a/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
@@ -15,10 +15,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/********************* Pitch Filter Lag Coefficient Tables ************************/
@@ -98,4 +98,4 @@
extern const int16_t WebRtcIsacfix_kMeanLag4Hi[34];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/settings.h b/modules/audio_coding/codecs/isac/fix/source/settings.h
index 82eb51a..34c0efe 100644
--- a/modules/audio_coding/codecs/isac/fix/source/settings.h
+++ b/modules/audio_coding/codecs/isac/fix/source/settings.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
/* sampling frequency (Hz) */
@@ -212,4 +212,4 @@
#define ISAC_INCOMPATIBLE_FORMATS 6810
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
index 5583206..fd8d04a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
@@ -16,11 +16,11 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/********************* AR Coefficient Tables ************************/
/* cdf for quantized reflection coefficient 1 */
@@ -92,4 +92,4 @@
/* Cosine table */
extern const int16_t WebRtcIsacfix_kCos[6][60];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/structs.h b/modules/audio_coding/codecs/isac/fix/source/structs.h
index 278af75..8fdb109 100644
--- a/modules/audio_coding/codecs/isac/fix/source/structs.h
+++ b/modules/audio_coding/codecs/isac/fix/source/structs.h
@@ -15,14 +15,14 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/typedefs.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "typedefs.h"
/* Bitstream struct for decoder */
typedef struct Bitstreamstruct_dec {
@@ -379,4 +379,4 @@
//Bitstr_enc myBitStr;
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform.c b/modules/audio_coding/codecs/isac/fix/source/transform.c
index 362610a..2f1275d 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform.c
+++ b/modules/audio_coding/codecs/isac/fix/source/transform.c
@@ -15,9 +15,9 @@
*
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "modules/audio_coding/codecs/isac/fix/source/fft.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
/* Tables are defined in transform_tables.c file or ARM assembly files. */
/* Cosine table 1 in Q14 */
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_mips.c b/modules/audio_coding/codecs/isac/fix/source/transform_mips.c
index e5d35f2..a87b3b5 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform_mips.c
+++ b/modules/audio_coding/codecs/isac/fix/source/transform_mips.c
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "modules/audio_coding/codecs/isac/fix/source/fft.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
// The tables are defined in transform_tables.c file.
extern const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2];
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_neon.c b/modules/audio_coding/codecs/isac/fix/source/transform_neon.c
index f0cbd5d..79dadc4 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform_neon.c
+++ b/modules/audio_coding/codecs/isac/fix/source/transform_neon.c
@@ -10,9 +10,9 @@
#include <arm_neon.h>
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "modules/audio_coding/codecs/isac/fix/source/fft.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
// Tables are defined in transform_tables.c file.
// Cosine table 1 in Q14.
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_tables.c b/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
index 8f89fb8..49f235a 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
+++ b/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
@@ -13,8 +13,8 @@
* transform functions WebRtcIsacfix_Time2Spec and WebRtcIsacfix_Spec2Time.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "typedefs.h"
/* Cosine table 1 in Q14. */
const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = {
diff --git a/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc b/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
index 35e5800..347b049 100644
--- a/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
+++ b/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h"
-#include "webrtc/system_wrappers/include/cpu_features_wrapper.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
+#include "system_wrappers/include/cpu_features_wrapper.h"
+#include "test/gtest.h"
static const int kSamples = FRAMESAMPLES/2;
static const int32_t spec2time_out_expected_1[kSamples] = {
diff --git a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
index 276eb60..f9f6cc9 100644
--- a/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
+++ b/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h"
-#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
using ::std::string;
diff --git a/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index 79a525b..b11c670 100644
--- a/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -14,9 +14,9 @@
#include <time.h>
#include <ctype.h>
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/perf_test.h"
+#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#include "test/gtest.h"
+#include "test/testsupport/perf_test.h"
// TODO(kma): Clean up the code and change benchmarking the whole codec to
// separate encoder and decoder.
diff --git a/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc b/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
index 5b9439f..80d10ab 100644
--- a/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
+++ b/modules/audio_coding/codecs/isac/locked_bandwidth_info.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/locked_bandwidth_info.h b/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
index a0e74b5..37074f8 100644
--- a/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
+++ b/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
-#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/rtc_base/atomicops.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/thread_annotations.h"
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "rtc_base/atomicops.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/thread_annotations.h"
namespace webrtc {
@@ -53,4 +53,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
diff --git a/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h b/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h
index 06821c0..fae2f3d 100644
--- a/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h
+++ b/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_DECODER_ISAC_H_
-#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
+#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
namespace webrtc {
using AudioDecoderIsacFloatImpl = AudioDecoderIsacT<IsacFloat>;
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
diff --git a/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h b/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h
index 06bef4c..dc32bcd 100644
--- a/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h
+++ b/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
-#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
+#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
+#include "modules/audio_coding/codecs/isac/main/source/isac_float_type.h"
namespace webrtc {
using AudioEncoderIsacFloatImpl = AudioEncoderIsacT<IsacFloat>;
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_AUDIO_ENCODER_ISAC_H_
diff --git a/modules/audio_coding/codecs/isac/main/include/isac.h b/modules/audio_coding/codecs/isac/main/include/isac.h
index 327e7f4..fcdb04a 100644
--- a/modules/audio_coding/codecs/isac/main/include/isac.h
+++ b/modules/audio_coding/codecs/isac/main/include/isac.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_
#include <stddef.h>
-#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "typedefs.h"
typedef struct WebRtcISACStruct ISACStruct;
@@ -721,4 +721,4 @@
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/arith_routines.h b/modules/audio_coding/codecs/isac/main/source/arith_routines.h
index d2fcbfe..73d0c23 100644
--- a/modules/audio_coding/codecs/isac/main/source/arith_routines.h
+++ b/modules/audio_coding/codecs/isac/main/source/arith_routines.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
#include "structs.h"
@@ -60,4 +60,4 @@
const uint16_t *init_index,/* input: vector of initial cdf table search entries */
const int N); /* input: data vector length */
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc b/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
index 8e0603e..b671002 100644
--- a/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
+++ b/modules/audio_coding/codecs/isac/main/source/audio_decoder_isac.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
+#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
index 64b9815..b7f2c0b 100644
--- a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
+++ b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
+#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
index 3fe8c1a..333ab52 100644
--- a/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
+++ b/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
@@ -10,8 +10,8 @@
#include <limits>
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
index c2c166c..4a16932 100644
--- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
+++ b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
@@ -19,7 +19,7 @@
#include "bandwidth_estimator.h"
#include "settings.h"
#include "isac.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
#include <math.h>
#include <string.h>
diff --git a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
index 0704337..ddebdcd 100644
--- a/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
+++ b/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
#include "structs.h"
#include "settings.h"
@@ -181,4 +181,4 @@
#endif
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/codec.h b/modules/audio_coding/codecs/isac/main/source/codec.h
index 7ef64b5..d938725 100644
--- a/modules/audio_coding/codecs/isac/main/source/codec.h
+++ b/modules/audio_coding/codecs/isac/main/source/codec.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
#include "structs.h"
@@ -230,4 +230,4 @@
void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/crc.h b/modules/audio_coding/codecs/isac/main/source/crc.h
index 09583df..dc8942f 100644
--- a/modules/audio_coding/codecs/isac/main/source/crc.h
+++ b/modules/audio_coding/codecs/isac/main/source/crc.h
@@ -15,10 +15,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/****************************************************************************
* WebRtcIsac_GetCrc(...)
@@ -43,4 +43,4 @@
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
index 12a263d..dcb658e 100644
--- a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
+++ b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
@@ -26,7 +26,7 @@
#include "lpc_shape_swb12_tables.h"
#include "lpc_shape_swb16_tables.h"
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/******************************************************************************
* WebRtcIsac_RemoveLarMean()
diff --git a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
index 3dd2311..c3efd88 100644
--- a/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
+++ b/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
@@ -16,12 +16,12 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
#include "settings.h"
#include "structs.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/******************************************************************************
* WebRtcIsac_RemoveLarMean()
@@ -279,4 +279,4 @@
double* lpcGains);
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/entropy_coding.h b/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
index d715d86..2ccd96b 100644
--- a/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
+++ b/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
#include "settings.h"
#include "structs.h"
@@ -340,4 +340,4 @@
int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
int32_t* jitterInfo);
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/fft.h b/modules/audio_coding/codecs/isac/main/source/fft.h
index a42f57b..b583b50 100644
--- a/modules/audio_coding/codecs/isac/main/source/fft.h
+++ b/modules/audio_coding/codecs/isac/main/source/fft.h
@@ -27,8 +27,8 @@
* See the comments in the code for correct usage!
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
#include "structs.h"
@@ -42,4 +42,4 @@
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h b/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h
index e8fda5e..714e227 100644
--- a/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
#include "structs.h"
@@ -43,4 +43,4 @@
/* The matrix for transforming the backward composite state to lower channel state */
extern const float WebRtcIsac_kTransform2Float[8];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/isac.c b/modules/audio_coding/codecs/isac/main/source/isac.c
index f83d4f8..79dc7e2 100644
--- a/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -15,22 +15,22 @@
*
*/
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/codec.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/crc.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/structs.h"
+#include "rtc_base/checks.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
+#include "modules/audio_coding/codecs/isac/main/source/crc.h"
+#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
#define BIT_MASK_DEC_INIT 0x0001
#define BIT_MASK_ENC_INIT 0x0002
diff --git a/modules/audio_coding/codecs/isac/main/source/isac_float_type.h b/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
index e150d39..59a8805 100644
--- a/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
+++ b/modules/audio_coding/codecs/isac/main/source/isac_float_type.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
namespace webrtc {
@@ -114,4 +114,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_FLOAT_TYPE_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
index 3f7170c..727f0f6 100644
--- a/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
+++ b/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
@@ -9,9 +9,9 @@
*/
#include <string>
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
struct WebRtcISACStruct;
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
index 8dfe383..1566ab0 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
#include "settings.h"
#include "structs.h"
@@ -47,4 +47,4 @@
double* varscale,
int16_t bandwidth);
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
index 5cc6c11..8ce004b 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
@@ -18,7 +18,7 @@
#include "lpc_gain_swb_tables.h"
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
const double WebRtcIsac_kQSizeLpcGain = 0.100000;
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
index c163f4a..8cbbbfb 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
@@ -16,11 +16,11 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
extern const double WebRtcIsac_kQSizeLpcGain;
@@ -46,4 +46,4 @@
extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES];
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
index 599b89d..2c5698f 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
@@ -18,7 +18,7 @@
#include "lpc_shape_swb12_tables.h"
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/*
* Mean value of LAR
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
index 256f1d4..b49fd98 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
@@ -16,11 +16,11 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
extern const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER];
@@ -62,4 +62,4 @@
extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb12
[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
index 6176d2c..0f567ed 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
@@ -18,7 +18,7 @@
#include "lpc_shape_swb16_tables.h"
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/*
* Mean value of LAR
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
index 3e1bdf7..737f363 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
@@ -16,11 +16,11 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
#include "settings.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
extern const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER];
@@ -75,4 +75,4 @@
extern const double WebRtcIsac_kLpcShapeQStepSizeUb16;
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
index 51f6316..9412f2e 100644
--- a/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
#include "structs.h"
@@ -97,4 +97,4 @@
extern const double WebRtcIsac_kLpcMeansShape[108];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h b/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
index 2b446e9..dad3264 100644
--- a/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
+++ b/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
@@ -9,11 +9,11 @@
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
#include <math.h>
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
#if defined(WEBRTC_POSIX)
#define WebRtcIsac_lrint lrint
@@ -38,4 +38,4 @@
#endif
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
+#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h b/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
index 6fb02b3..96e1e17 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
#include "structs.h"
@@ -72,4 +72,4 @@
size_t N, /* number of input samples */
double *out); /* array of size N/2 */
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
index 3010fbc..24ec63a 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
@@ -16,7 +16,7 @@
#include "os_specific_inline.h"
-#include "webrtc/rtc_base/compile_assert_c.h"
+#include "rtc_base/compile_assert_c.h"
/*
* We are implementing the following filters;
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
index 8d708ce..16bfb06 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
@@ -15,10 +15,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/* header file for coding tables for the pitch filter side-info in the entropy coder */
/********************* Pitch Filter Gain Coefficient Tables ************************/
@@ -42,4 +42,4 @@
/* size of cdf table */
extern const uint16_t WebRtcIsac_kQCdfTableSizeGain[1];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
index 01989f0..799c45a 100644
--- a/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
@@ -15,10 +15,10 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
/* header file for coding tables for the pitch filter side-info in the entropy coder */
/********************* Pitch Filter Lag Coefficient Tables ************************/
@@ -111,4 +111,4 @@
/* transpose transform matrix */
extern const double WebRtcIsac_kTransformTranspose[4][4];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/settings.h b/modules/audio_coding/codecs/isac/main/source/settings.h
index 31a8065..c08d72f 100644
--- a/modules/audio_coding/codecs/isac/main/source/settings.h
+++ b/modules/audio_coding/codecs/isac/main/source/settings.h
@@ -15,8 +15,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
/* sampling frequency (Hz) */
#define FS 16000
@@ -202,4 +202,4 @@
/* 6800 Call setup formats */
#define ISAC_INCOMPATIBLE_FORMATS 6810
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
index 989cb36..82ab363 100644
--- a/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
+++ b/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
@@ -16,8 +16,8 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
#include "structs.h"
@@ -75,4 +75,4 @@
/* Cosine table */
extern const int16_t WebRtcIsac_kCos[6][60];
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/source/structs.h b/modules/audio_coding/codecs/isac/main/source/structs.h
index a2cdca2..a33ef69 100644
--- a/modules/audio_coding/codecs/isac/main/source/structs.h
+++ b/modules/audio_coding/codecs/isac/main/source/structs.h
@@ -15,13 +15,13 @@
*
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
-#include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/source/settings.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/isac/bandwidth_info.h"
+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
+#include "typedefs.h"
typedef struct Bitstreamstruct {
@@ -492,4 +492,4 @@
TransformTables transform_tables;
} ISACMainStruct;
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
diff --git a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index 9c11d23..33d9efc 100644
--- a/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -21,7 +21,7 @@
/* include API */
#include "isac.h"
#include "utility.h"
-#include "webrtc/rtc_base/format_macros.h"
+#include "rtc_base/format_macros.h"
/* Defines */
#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
diff --git a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 646cd48..0d2d2cb 100644
--- a/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -28,7 +28,7 @@
/* include API */
#include "isac.h"
#include "utility.h"
-#include "webrtc/rtc_base/format_macros.h"
+#include "rtc_base/format_macros.h"
/* max number of samples per frame (= 60 ms frame) */
#define MAX_FRAMESAMPLES_SWB 1920
diff --git a/modules/audio_coding/codecs/isac/main/util/utility.h b/modules/audio_coding/codecs/isac/main/util/utility.h
index 1bb6d29..b5882a5 100644
--- a/modules/audio_coding/codecs/isac/main/util/utility.h
+++ b/modules/audio_coding/codecs/isac/main/util/utility.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
#include <stdlib.h>
#include <stdio.h>
diff --git a/modules/audio_coding/codecs/isac/unittest.cc b/modules/audio_coding/codecs/isac/unittest.cc
index 1f4ae80..7a811cf 100644
--- a/modules/audio_coding/codecs/isac/unittest.cc
+++ b/modules/audio_coding/codecs/isac/unittest.cc
@@ -13,12 +13,12 @@
#include <sstream>
#include <vector>
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/buffer.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
index e0f1faf..8c35b64 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
#include <algorithm>
#include <memory>
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
index e8ea029..275576e 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+#define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
#include <vector>
-#include "webrtc/api/array_view.h"
-#include "webrtc/api/audio_codecs/audio_decoder.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
namespace webrtc {
@@ -45,4 +45,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
+#endif // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_
diff --git a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
index fe049ab..9fd6044 100644
--- a/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
+++ b/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/acm2/rent_a_codec.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index 2270ff4..a48c7db 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#include <utility>
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index d8512ec..70aa40b 100644
--- a/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -52,4 +52,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
+#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_DECODER_OPUS_H_
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index 3b4473a..3da6ea2 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -8,27 +8,27 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
#include <algorithm>
#include <iterator>
#include <utility>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/numerics/exp_filter.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/safe_minmax.h"
-#include "webrtc/rtc_base/string_to_number.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
+#include "common_types.h"
+#include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
+#include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/exp_filter.h"
+#include "rtc_base/protobuf_utils.h"
+#include "rtc_base/ptr_util.h"
+#include "rtc_base/safe_conversions.h"
+#include "rtc_base/safe_minmax.h"
+#include "rtc_base/string_to_number.h"
+#include "rtc_base/timeutils.h"
+#include "system_wrappers/include/field_trial.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index f966dbe..f2a87c9 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -8,23 +8,23 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
#include <functional>
#include <memory>
#include <string>
#include <vector>
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/common_audio/smoothing_filter.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/audio_codecs/opus/audio_encoder_opus_config.h"
+#include "api/optional.h"
+#include "common_audio/smoothing_filter.h"
+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/protobuf_utils.h"
namespace webrtc {
@@ -179,4 +179,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
+#endif // MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 08e168b..5720e86 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -12,17 +12,17 @@
#include <memory>
#include <utility>
-#include "webrtc/common_audio/mocks/mock_smoothing_filter.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/fakeclock.h"
-#include "webrtc/test/field_trial.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_audio/mocks/mock_smoothing_filter.h"
+#include "common_types.h"
+#include "modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/fakeclock.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
using ::testing::NiceMock;
diff --git a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
index 28d5175..2fe6135 100644
--- a/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_complexity_unittest.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/perf_test.h"
+#include "modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/format_macros.h"
+#include "rtc_base/timeutils.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "test/testsupport/perf_test.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/opus/opus_fec_test.cc b/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 8f42bf4..e429f1c 100644
--- a/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -10,10 +10,10 @@
#include <memory>
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "rtc_base/format_macros.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
using ::std::string;
using ::std::tr1::tuple;
diff --git a/modules/audio_coding/codecs/opus/opus_inst.h b/modules/audio_coding/codecs/opus/opus_inst.h
index 976f09e..066fa22 100644
--- a/modules/audio_coding/codecs/opus/opus_inst.h
+++ b/modules/audio_coding/codecs/opus/opus_inst.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
#include <stddef.h>
-#include "webrtc/rtc_base/ignore_wundef.h"
+#include "rtc_base/ignore_wundef.h"
RTC_PUSH_IGNORING_WUNDEF()
#include "opus.h"
@@ -33,4 +33,4 @@
};
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
+#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
diff --git a/modules/audio_coding/codecs/opus/opus_interface.c b/modules/audio_coding/codecs/opus/opus_interface.c
index 6427f0f..5166f4c 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.c
+++ b/modules/audio_coding/codecs/opus/opus_interface.c
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
+#include "rtc_base/checks.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
#include <stdlib.h>
#include <string.h>
diff --git a/modules/audio_coding/codecs/opus/opus_interface.h b/modules/audio_coding/codecs/opus/opus_interface.h
index 2985151..0909ef8 100644
--- a/modules/audio_coding/codecs/opus/opus_interface.h
+++ b/modules/audio_coding/codecs/opus/opus_interface.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
#include <stddef.h>
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
#ifdef __cplusplus
extern "C" {
@@ -394,4 +394,4 @@
} // extern "C"
#endif
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
+#endif // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
diff --git a/modules/audio_coding/codecs/opus/opus_speed_test.cc b/modules/audio_coding/codecs/opus/opus_speed_test.cc
index 7165d29..7a97d22 100644
--- a/modules/audio_coding/codecs/opus/opus_speed_test.cc
+++ b/modules/audio_coding/codecs/opus/opus_speed_test.cc
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
using ::std::string;
diff --git a/modules/audio_coding/codecs/opus/opus_unittest.cc b/modules/audio_coding/codecs/opus/opus_unittest.cc
index 9cff920..9f2c1bb 100644
--- a/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -11,12 +11,12 @@
#include <memory>
#include <string>
-#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 57724a3..b07624d 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
index 46dfbb2..7d23422 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_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"
namespace webrtc {
@@ -41,4 +41,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
+#endif // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_DECODER_PCM16B_H_
diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index 7b4a919..d179ffb 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
#include <algorithm>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 25d548c..d6fd6e1 100644
--- a/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
-#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -47,4 +47,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
+#endif // MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b.c b/modules/audio_coding/codecs/pcm16b/pcm16b.c
index 120c790..7411d74 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b.c
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b.c
@@ -10,7 +10,7 @@
#include "pcm16b.h"
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
size_t WebRtcPcm16b_Encode(const int16_t* speech,
size_t len,
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b.h b/modules/audio_coding/codecs/pcm16b/pcm16b.h
index f96e741..f32eb98 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b.h
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_
/*
* Define the fixpoint numeric formats
*/
#include <stddef.h>
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
#ifdef __cplusplus
extern "C" {
@@ -65,4 +65,4 @@
}
#endif
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */
+#endif /* MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_H_ */
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc b/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
index 36015e7..6d0fc2d 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b_common.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/pcm16b/pcm16b_common.h b/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
index 5d764c9..980a996 100644
--- a/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
+++ b/modules/audio_coding/codecs/pcm16b/pcm16b_common.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+#define MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
#include <vector>
-#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
namespace webrtc {
void Pcm16BAppendSupportedCodecSpecs(std::vector<AudioCodecSpec>* specs);
}
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
+#endif // MODULES_AUDIO_CODING_CODECS_PCM16B_PCM16B_COMMON_H_
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 3680ff0..4b9df6e 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include <string.h>
#include <utility>
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 664c13e..e625c50 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+#define MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
#include <memory>
#include <vector>
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -74,4 +74,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
+#endif // MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_
diff --git a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index a79bb1c..06ae2a7 100644
--- a/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -11,10 +11,10 @@
#include <memory>
#include <vector>
-#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_encoder.h"
+#include "modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/mock_audio_encoder.h"
using ::testing::Return;
using ::testing::_;
diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index b9332e0..3f6a443 100644
--- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
+#include "modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "rtc_base/format_macros.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
using ::std::tr1::get;
diff --git a/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index b79b1ef..73e0d22 100644
--- a/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#ifndef MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#define MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
#include <memory>
#include <string>
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
@@ -89,4 +89,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
+#endif // MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
diff --git a/modules/audio_coding/include/audio_coding_module.h b/modules/audio_coding/include/audio_coding_module.h
index eb59404..07a238a 100644
--- a/modules/audio_coding/include/audio_coding_module.h
+++ b/modules/audio_coding/include/audio_coding_module.h
@@ -8,24 +8,24 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
#include <memory>
#include <string>
#include <vector>
-#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/include/module.h"
-#include "webrtc/rtc_base/deprecation.h"
-#include "webrtc/rtc_base/function_view.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/optional.h"
+#include "common_types.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/include/module.h"
+#include "rtc_base/deprecation.h"
+#include "rtc_base/function_view.h"
+#include "system_wrappers/include/clock.h"
+#include "typedefs.h"
namespace webrtc {
@@ -829,4 +829,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
+#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index 280d6bf..69a60e7 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
#include <map>
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
+#include "modules/include/module_common_types.h"
+#include "typedefs.h"
namespace webrtc {
@@ -48,4 +48,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
+#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/modules/audio_coding/neteq/accelerate.cc b/modules/audio_coding/neteq/accelerate.cc
index 1c36fa8..183ad7b 100644
--- a/modules/audio_coding/neteq/accelerate.cc
+++ b/modules/audio_coding/neteq/accelerate.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/accelerate.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/accelerate.h b/modules/audio_coding/neteq/accelerate.h
index 713344e..0ec33c2 100644
--- a/modules/audio_coding/neteq/accelerate.h
+++ b/modules/audio_coding/neteq/accelerate.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
+#define MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
#include <assert.h>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/time_stretch.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -78,4 +78,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.cc b/modules/audio_coding/neteq/audio_decoder_impl.cc
index d60f4bf..01e934b 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.cc
+++ b/modules/audio_coding/neteq/audio_decoder_impl.cc
@@ -8,30 +8,30 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
#include <assert.h>
-#include "webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
+#include "rtc_base/checks.h"
#ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
#endif
#ifdef WEBRTC_CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_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_decoder_opus.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
#endif
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/audio_decoder_impl.h b/modules/audio_coding/neteq/audio_decoder_impl.h
index 8dc746c..c724e72 100644
--- a/modules/audio_coding/neteq/audio_decoder_impl.h
+++ b/modules/audio_coding/neteq/audio_decoder_impl.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
#include <assert.h>
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
#ifdef WEBRTC_CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
#endif
namespace webrtc {
@@ -28,4 +28,4 @@
bool CodecSupported(NetEqDecoder codec_type);
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_AUDIO_DECODER_IMPL_H_
diff --git a/modules/audio_coding/neteq/audio_decoder_unittest.cc b/modules/audio_coding/neteq/audio_decoder_unittest.cc
index f6ae2dd..203b856 100644
--- a/modules/audio_coding/neteq/audio_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/audio_decoder_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
#include <assert.h>
#include <stdlib.h>
@@ -17,23 +17,23 @@
#include <string>
#include <vector>
-#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.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/g722/audio_decoder_g722.h"
-#include "webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
-#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "api/audio_codecs/opus/audio_encoder_opus.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/g722/audio_decoder_g722.h"
+#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
+#include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
+#include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
+#include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/audio_multi_vector.cc b/modules/audio_coding/neteq/audio_multi_vector.cc
index 8263752..be67d68 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include <assert.h>
#include <algorithm>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/checks.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/audio_multi_vector.h b/modules/audio_coding/neteq/audio_multi_vector.h
index b6f1d39..6a51d56 100644
--- a/modules/audio_coding/neteq/audio_multi_vector.h
+++ b/modules/audio_coding/neteq/audio_multi_vector.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
#include <string.h> // Access to size_t.
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/audio_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -136,4 +136,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_AUDIO_MULTI_VECTOR_H_
diff --git a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
index af8d5f1..18980f2 100644
--- a/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include <assert.h>
#include <stdlib.h>
#include <string>
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/audio_vector.cc b/modules/audio_coding/neteq/audio_vector.cc
index c819daa..c66682b 100644
--- a/modules/audio_coding/neteq/audio_vector.cc
+++ b/modules/audio_coding/neteq/audio_vector.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/audio_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
#include <assert.h>
#include <algorithm>
#include <memory>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/checks.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/audio_vector.h b/modules/audio_coding/neteq/audio_vector.h
index f3dfdd6..9275cb3 100644
--- a/modules/audio_coding/neteq/audio_vector.h
+++ b/modules/audio_coding/neteq/audio_vector.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
#include <string.h> // Access to size_t.
#include <memory>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -165,4 +165,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_AUDIO_VECTOR_H_
diff --git a/modules/audio_coding/neteq/audio_vector_unittest.cc b/modules/audio_coding/neteq/audio_vector_unittest.cc
index e9ef93f..ef024d6 100644
--- a/modules/audio_coding/neteq/audio_vector_unittest.cc
+++ b/modules/audio_coding/neteq/audio_vector_unittest.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/audio_vector.h"
+#include "modules/audio_coding/neteq/audio_vector.h"
#include <assert.h>
#include <stdlib.h>
#include <string>
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/background_noise.cc b/modules/audio_coding/neteq/background_noise.cc
index cf77ec5..eda5c75 100644
--- a/modules/audio_coding/neteq/background_noise.cc
+++ b/modules/audio_coding/neteq/background_noise.cc
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/background_noise.h"
#include <assert.h>
#include <string.h> // memcpy
#include <algorithm> // min, max
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
-#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/background_noise.h b/modules/audio_coding/neteq/background_noise.h
index 0c662bd..c6d51ad 100644
--- a/modules/audio_coding/neteq/background_noise.h
+++ b/modules/audio_coding/neteq/background_noise.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
+#define MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
#include <string.h> // size_t
#include <memory>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -134,4 +134,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_BACKGROUND_NOISE_H_
diff --git a/modules/audio_coding/neteq/background_noise_unittest.cc b/modules/audio_coding/neteq/background_noise_unittest.cc
index 3305b2c..e32492f 100644
--- a/modules/audio_coding/neteq/background_noise_unittest.cc
+++ b/modules/audio_coding/neteq/background_noise_unittest.cc
@@ -10,9 +10,9 @@
// Unit tests for BackgroundNoise class.
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/buffer_level_filter.cc b/modules/audio_coding/neteq/buffer_level_filter.cc
index 9054791..5df59c7 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include <algorithm> // Provide access to std::max.
diff --git a/modules/audio_coding/neteq/buffer_level_filter.h b/modules/audio_coding/neteq/buffer_level_filter.h
index bc9a10e..7a48c72 100644
--- a/modules/audio_coding/neteq/buffer_level_filter.h
+++ b/modules/audio_coding/neteq/buffer_level_filter.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
#include <stddef.h>
-#include "webrtc/rtc_base/constructormagic.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -46,4 +46,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_BUFFER_LEVEL_FILTER_H_
diff --git a/modules/audio_coding/neteq/buffer_level_filter_unittest.cc b/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
index cfa6e3d..72c8727 100644
--- a/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
+++ b/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
@@ -10,11 +10,11 @@
// Unit tests for BufferLevelFilter class.
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
#include <math.h> // Access to pow function.
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/comfort_noise.cc b/modules/audio_coding/neteq/comfort_noise.cc
index 8482e8d..d2635f3 100644
--- a/modules/audio_coding/neteq/comfort_noise.cc
+++ b/modules/audio_coding/neteq/comfort_noise.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
+#include "modules/audio_coding/neteq/comfort_noise.h"
#include <assert.h>
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/rtc_base/logging.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/logging.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/comfort_noise.h b/modules/audio_coding/neteq/comfort_noise.h
index fe663cc..e13abb8 100644
--- a/modules/audio_coding/neteq/comfort_noise.h
+++ b/modules/audio_coding/neteq/comfort_noise.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
+#define MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -68,4 +68,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_COMFORT_NOISE_H_
diff --git a/modules/audio_coding/neteq/comfort_noise_unittest.cc b/modules/audio_coding/neteq/comfort_noise_unittest.cc
index 2de6886..b3fbb4e 100644
--- a/modules/audio_coding/neteq/comfort_noise_unittest.cc
+++ b/modules/audio_coding/neteq/comfort_noise_unittest.cc
@@ -10,11 +10,11 @@
// Unit tests for ComfortNoise class.
-#include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
+#include "modules/audio_coding/neteq/comfort_noise.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/cross_correlation.cc b/modules/audio_coding/neteq/cross_correlation.cc
index ad89ab8..da9c913 100644
--- a/modules/audio_coding/neteq/cross_correlation.cc
+++ b/modules/audio_coding/neteq/cross_correlation.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
#include <cstdlib>
#include <limits>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/cross_correlation.h b/modules/audio_coding/neteq/cross_correlation.h
index db14141..ab18c7e 100644
--- a/modules/audio_coding/neteq/cross_correlation.h
+++ b/modules/audio_coding/neteq/cross_correlation.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+#define MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
-#include "webrtc/common_types.h"
+#include "common_types.h"
namespace webrtc {
@@ -47,4 +47,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_CROSS_CORRELATION_H_
diff --git a/modules/audio_coding/neteq/decision_logic.cc b/modules/audio_coding/neteq/decision_logic.cc
index 94c542d..966d5c3 100644
--- a/modules/audio_coding/neteq/decision_logic.cc
+++ b/modules/audio_coding/neteq/decision_logic.cc
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
#include <algorithm>
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h"
-#include "webrtc/modules/audio_coding/neteq/decision_logic_normal.h"
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/decision_logic_fax.h"
+#include "modules/audio_coding/neteq/decision_logic_normal.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/decision_logic.h b/modules/audio_coding/neteq/decision_logic.h
index 7d9ba60..b1ea84a 100644
--- a/modules/audio_coding/neteq/decision_logic.h
+++ b/modules/audio_coding/neteq/decision_logic.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
-#include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/defines.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -165,4 +165,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_
diff --git a/modules/audio_coding/neteq/decision_logic_fax.cc b/modules/audio_coding/neteq/decision_logic_fax.cc
index 7db8f30..cc21ee9 100644
--- a/modules/audio_coding/neteq/decision_logic_fax.cc
+++ b/modules/audio_coding/neteq/decision_logic_fax.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/decision_logic_fax.h"
+#include "modules/audio_coding/neteq/decision_logic_fax.h"
#include <assert.h>
#include <algorithm>
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/decision_logic_fax.h b/modules/audio_coding/neteq/decision_logic_fax.h
index 4efb874..1eb5b76 100644
--- a/modules/audio_coding/neteq/decision_logic_fax.h
+++ b/modules/audio_coding/neteq/decision_logic_fax.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
-#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -54,4 +54,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
diff --git a/modules/audio_coding/neteq/decision_logic_normal.cc b/modules/audio_coding/neteq/decision_logic_normal.cc
index c5f2570..c78063c 100644
--- a/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/decision_logic_normal.h"
+#include "modules/audio_coding/neteq/decision_logic_normal.h"
#include <assert.h>
#include <algorithm>
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/include/module_common_types.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/decision_logic_normal.h b/modules/audio_coding/neteq/decision_logic_normal.h
index b676bd4..f011ae9 100644
--- a/modules/audio_coding/neteq/decision_logic_normal.h
+++ b/modules/audio_coding/neteq/decision_logic_normal.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
-#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -104,4 +104,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc
index a5c5778..1a7bab9 100644
--- a/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -10,15 +10,15 @@
// Unit tests for DecisionLogic class and derived classes.
-#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/delay_peak_detector.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/decoder_database.cc b/modules/audio_coding/neteq/decoder_database.cc
index 526d159..feb0203 100644
--- a/modules/audio_coding/neteq/decoder_database.cc
+++ b/modules/audio_coding/neteq/decoder_database.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
#include <utility> // pair
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/decoder_database.h b/modules/audio_coding/neteq/decoder_database.h
index 5feadd1..dff6464 100644
--- a/modules/audio_coding/neteq/decoder_database.h
+++ b/modules/audio_coding/neteq/decoder_database.h
@@ -8,22 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
+#define MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
#include <map>
#include <memory>
#include <string>
-#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/common_types.h" // NULL
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "common_types.h" // NULL
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"
namespace webrtc {
@@ -243,4 +243,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DECODER_DATABASE_H_
diff --git a/modules/audio_coding/neteq/decoder_database_unittest.cc b/modules/audio_coding/neteq/decoder_database_unittest.cc
index 3e65485..be3c0b7 100644
--- a/modules/audio_coding/neteq/decoder_database_unittest.cc
+++ b/modules/audio_coding/neteq/decoder_database_unittest.cc
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
#include <assert.h>
#include <stdlib.h>
#include <string>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder.h"
-#include "webrtc/test/mock_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
using testing::_;
using testing::Invoke;
diff --git a/modules/audio_coding/neteq/defines.h b/modules/audio_coding/neteq/defines.h
index 3ed6b61..496a36d 100644
--- a/modules/audio_coding/neteq/defines.h
+++ b/modules/audio_coding/neteq/defines.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
+#define MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
namespace webrtc {
@@ -49,4 +49,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DEFINES_H_
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index bf92588..ab98a06 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
#include <assert.h>
#include <math.h>
#include <algorithm> // max, min
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/delay_peak_detector.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index ac1de25..6339096 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
#include <string.h> // Provide access to size_t.
#include <memory>
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -171,4 +171,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DELAY_MANAGER_H_
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index a6ee58d..6bdbc38 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -10,13 +10,13 @@
// Unit tests for DelayManager class.
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
#include <math.h>
-#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/delay_peak_detector.cc b/modules/audio_coding/neteq/delay_peak_detector.cc
index b3fe8a7..16f41d3 100644
--- a/modules/audio_coding/neteq/delay_peak_detector.cc
+++ b/modules/audio_coding/neteq/delay_peak_detector.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
+#include "modules/audio_coding/neteq/delay_peak_detector.h"
#include <algorithm> // max
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/delay_peak_detector.h b/modules/audio_coding/neteq/delay_peak_detector.h
index 455d4e9..2236ef2 100644
--- a/modules/audio_coding/neteq/delay_peak_detector.h
+++ b/modules/audio_coding/neteq/delay_peak_detector.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
#include <string.h> // size_t
#include <list>
#include <memory>
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -71,4 +71,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DELAY_PEAK_DETECTOR_H_
diff --git a/modules/audio_coding/neteq/delay_peak_detector_unittest.cc b/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
index e927593..058ba66 100644
--- a/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
+++ b/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
@@ -10,9 +10,9 @@
// Unit tests for DelayPeakDetector class.
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
+#include "modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/dsp_helper.cc b/modules/audio_coding/neteq/dsp_helper.cc
index 3275665..2a1d81b 100644
--- a/modules/audio_coding/neteq/dsp_helper.cc
+++ b/modules/audio_coding/neteq/dsp_helper.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
#include <assert.h>
#include <string.h> // Access to memset.
#include <algorithm> // Access to min, max.
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/dsp_helper.h b/modules/audio_coding/neteq/dsp_helper.h
index 599110a..398fc00 100644
--- a/modules/audio_coding/neteq/dsp_helper.h
+++ b/modules/audio_coding/neteq/dsp_helper.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
#include <string.h> // Access to size_t.
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -141,4 +141,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DSP_HELPER_H_
diff --git a/modules/audio_coding/neteq/dsp_helper_unittest.cc b/modules/audio_coding/neteq/dsp_helper_unittest.cc
index fbdc047..1de16c9 100644
--- a/modules/audio_coding/neteq/dsp_helper_unittest.cc
+++ b/modules/audio_coding/neteq/dsp_helper_unittest.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/dtmf_buffer.cc b/modules/audio_coding/neteq/dtmf_buffer.cc
index c5e5adf..b06de5e 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.cc
+++ b/modules/audio_coding/neteq/dtmf_buffer.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
#include <assert.h>
#include <algorithm> // max
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
// Modify the code to obtain backwards bit-exactness. Once bit-exactness is no
// longer required, this #define should be removed (and the code that it
diff --git a/modules/audio_coding/neteq/dtmf_buffer.h b/modules/audio_coding/neteq/dtmf_buffer.h
index 068ad64..341942f 100644
--- a/modules/audio_coding/neteq/dtmf_buffer.h
+++ b/modules/audio_coding/neteq/dtmf_buffer.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
#include <list>
#include <string> // size_t
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -111,4 +111,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DTMF_BUFFER_H_
diff --git a/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
index d906d1c..7bcf1e0 100644
--- a/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
#ifdef WIN32
#include <winsock2.h> // ntohl()
@@ -18,7 +18,7 @@
#include <iostream>
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
// Modify the tests so that they pass with the modifications done to DtmfBuffer
// for backwards bit-exactness. Once bit-exactness is no longer required, this
diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.cc b/modules/audio_coding/neteq/dtmf_tone_generator.cc
index 416835b..b848c60 100644
--- a/modules/audio_coding/neteq/dtmf_tone_generator.cc
+++ b/modules/audio_coding/neteq/dtmf_tone_generator.cc
@@ -28,10 +28,10 @@
// 852 Hz 7 8 9 14
// 941 Hz 10 0 11 15
-#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/dtmf_tone_generator.h b/modules/audio_coding/neteq/dtmf_tone_generator.h
index 713987d..f3f77cc 100644
--- a/modules/audio_coding/neteq/dtmf_tone_generator.h
+++ b/modules/audio_coding/neteq/dtmf_tone_generator.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -51,4 +51,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_DTMF_TONE_GENERATOR_H_
diff --git a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
index 9eea103..437af0d 100644
--- a/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
+++ b/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
@@ -10,12 +10,12 @@
// Unit tests for DtmfToneGenerator class.
-#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
#include <math.h>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/expand.cc b/modules/audio_coding/neteq/expand.cc
index f8ae526..7223f6c 100644
--- a/modules/audio_coding/neteq/expand.cc
+++ b/modules/audio_coding/neteq/expand.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/expand.h"
#include <assert.h>
#include <string.h> // memset
@@ -16,14 +16,14 @@
#include <algorithm> // min, max
#include <limits> // numeric_limits<T>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
-#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/expand.h b/modules/audio_coding/neteq/expand.h
index af15432..0044ea0 100644
--- a/modules/audio_coding/neteq/expand.h
+++ b/modules/audio_coding/neteq/expand.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
#include <assert.h>
#include <memory>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -158,4 +158,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_EXPAND_H_
diff --git a/modules/audio_coding/neteq/expand_unittest.cc b/modules/audio_coding/neteq/expand_unittest.cc
index 40cb2eb..b52d626 100644
--- a/modules/audio_coding/neteq/expand_unittest.cc
+++ b/modules/audio_coding/neteq/expand_unittest.cc
@@ -10,17 +10,17 @@
// Unit tests for Expand class.
-#include "webrtc/modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/expand.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "rtc_base/safe_conversions.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index 9f5af7c..22bff2e 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -8,20 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
+#define MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
#include <string.h> // Provide access to size_t.
#include <string>
#include <vector>
-#include "webrtc/api/optional.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
+#include "api/optional.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"
namespace webrtc {
@@ -311,4 +311,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
diff --git a/modules/audio_coding/neteq/merge.cc b/modules/audio_coding/neteq/merge.cc
index df4f792..71e0564 100644
--- a/modules/audio_coding/neteq/merge.cc
+++ b/modules/audio_coding/neteq/merge.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/merge.h"
+#include "modules/audio_coding/neteq/merge.h"
#include <assert.h>
#include <string.h> // memmove, memcpy, memset, size_t
@@ -16,14 +16,14 @@
#include <algorithm> // min, max
#include <memory>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
-#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/safe_minmax.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/safe_conversions.h"
+#include "rtc_base/safe_minmax.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/merge.h b/modules/audio_coding/neteq/merge.h
index 34e6639..6bb1bba 100644
--- a/modules/audio_coding/neteq/merge.h
+++ b/modules/audio_coding/neteq/merge.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+#define MODULES_AUDIO_CODING_NETEQ_MERGE_H_
#include <assert.h>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -99,4 +99,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MERGE_H_
diff --git a/modules/audio_coding/neteq/merge_unittest.cc b/modules/audio_coding/neteq/merge_unittest.cc
index bfabb22..7ff3b8c 100644
--- a/modules/audio_coding/neteq/merge_unittest.cc
+++ b/modules/audio_coding/neteq/merge_unittest.cc
@@ -10,16 +10,16 @@
// Unit tests for Merge class.
-#include "webrtc/modules/audio_coding/neteq/merge.h"
+#include "modules/audio_coding/neteq/merge.h"
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
index da22845..f662fb6 100644
--- a/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
+++ b/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -34,4 +34,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_BUFFER_LEVEL_FILTER_H_
diff --git a/modules/audio_coding/neteq/mock/mock_decoder_database.h b/modules/audio_coding/neteq/mock/mock_decoder_database.h
index 4018879..049b693 100644
--- a/modules/audio_coding/neteq/mock/mock_decoder_database.h
+++ b/modules/audio_coding/neteq/mock/mock_decoder_database.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
#include <string>
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -58,4 +58,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
diff --git a/modules/audio_coding/neteq/mock/mock_delay_manager.h b/modules/audio_coding/neteq/mock/mock_delay_manager.h
index 85ed71d..61f209d 100644
--- a/modules/audio_coding/neteq/mock/mock_delay_manager.h
+++ b/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -59,4 +59,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_MANAGER_H_
diff --git a/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h b/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
index 3a80cb4..f6cdea0 100644
--- a/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
+++ b/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
+#include "modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -32,4 +32,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DELAY_PEAK_DETECTOR_H_
diff --git a/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h b/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
index 0aac22c..153a4d7 100644
--- a/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
-#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -35,4 +35,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_BUFFER_H_
diff --git a/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
index bb0a535..2cb5980 100644
--- a/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
+++ b/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
-#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -32,4 +32,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DTMF_TONE_GENERATOR_H_
diff --git a/modules/audio_coding/neteq/mock/mock_expand.h b/modules/audio_coding/neteq/mock/mock_expand.h
index 9465f6f..05fdaec 100644
--- a/modules/audio_coding/neteq/mock/mock_expand.h
+++ b/modules/audio_coding/neteq/mock/mock_expand.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
-#include "webrtc/modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/expand.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -61,4 +61,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXPAND_H_
diff --git a/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h b/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
index 3848680..2a9019b 100644
--- a/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
+++ b/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "rtc_base/constructormagic.h"
+#include "test/gmock.h"
+#include "typedefs.h"
namespace webrtc {
@@ -95,4 +95,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_EXTERNAL_DECODER_PCM16B_H_
diff --git a/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
index 1530329..ac7d9b7 100644
--- a/modules/audio_coding/neteq/mock/mock_packet_buffer.h
+++ b/modules/audio_coding/neteq/mock/mock_packet_buffer.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -65,4 +65,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_BUFFER_H_
diff --git a/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h b/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
index 8c6e939..27a2276 100644
--- a/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
+++ b/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
-#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -26,4 +26,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_RED_PAYLOAD_SPLITTER_H_
diff --git a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
index c854b24..85f2620 100644
--- a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
+++ b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -24,4 +24,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_STATISTICS_CALCULATOR_H_
diff --git a/modules/audio_coding/neteq/nack_tracker.cc b/modules/audio_coding/neteq/nack_tracker.cc
index 9829cf9..d187883 100644
--- a/modules/audio_coding/neteq/nack_tracker.cc
+++ b/modules/audio_coding/neteq/nack_tracker.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/nack_tracker.h"
+#include "modules/audio_coding/neteq/nack_tracker.h"
#include <assert.h> // For assert.
#include <algorithm> // For std::max.
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace {
diff --git a/modules/audio_coding/neteq/nack_tracker.h b/modules/audio_coding/neteq/nack_tracker.h
index 4f98c82..4f88d91 100644
--- a/modules/audio_coding/neteq/nack_tracker.h
+++ b/modules/audio_coding/neteq/nack_tracker.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+#define MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
#include <vector>
#include <map>
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/rtc_base/gtest_prod_util.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "rtc_base/gtest_prod_util.h"
//
// The NackTracker class keeps track of the lost packets, an estimate of
@@ -205,4 +205,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
diff --git a/modules/audio_coding/neteq/nack_tracker_unittest.cc b/modules/audio_coding/neteq/nack_tracker_unittest.cc
index f27b512..a3d1800 100644
--- a/modules/audio_coding/neteq/nack_tracker_unittest.cc
+++ b/modules/audio_coding/neteq/nack_tracker_unittest.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/nack_tracker.h"
+#include "modules/audio_coding/neteq/nack_tracker.h"
#include <stdint.h>
#include <algorithm>
#include <memory>
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
namespace {
diff --git a/modules/audio_coding/neteq/neteq.cc b/modules/audio_coding/neteq/neteq.cc
index 73233c4..8b74973 100644
--- a/modules/audio_coding/neteq/neteq.cc
+++ b/modules/audio_coding/neteq/neteq.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
#include <memory>
#include <sstream>
-#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
+#include "modules/audio_coding/neteq/neteq_impl.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/neteq_decoder_enum.cc b/modules/audio_coding/neteq/neteq_decoder_enum.cc
index e9abf20..2c61b0a 100644
--- a/modules/audio_coding/neteq/neteq_decoder_enum.cc
+++ b/modules/audio_coding/neteq/neteq_decoder_enum.cc
@@ -11,7 +11,7 @@
#include <map>
#include <string>
-#include "webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h"
+#include "modules/audio_coding/neteq/neteq_decoder_enum.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/neteq_decoder_enum.h b/modules/audio_coding/neteq/neteq_decoder_enum.h
index e108ccf..024f03c 100644
--- a/modules/audio_coding/neteq/neteq_decoder_enum.h
+++ b/modules/audio_coding/neteq/neteq_decoder_enum.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
+#define MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
-#include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/api/optional.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/optional.h"
namespace webrtc {
@@ -53,4 +53,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
diff --git a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
index ecdcafa..32af0ff 100644
--- a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
@@ -12,15 +12,15 @@
#include <memory>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "test/gmock.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 7858e84..680250f 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
+#include "modules/audio_coding/neteq/neteq_impl.h"
#include <assert.h>
@@ -16,37 +16,37 @@
#include <utility>
#include <vector>
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/accelerate.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/comfort_noise.h"
-#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
-#include "webrtc/modules/audio_coding/neteq/dtmf_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/merge.h"
-#include "webrtc/modules/audio_coding/neteq/nack_tracker.h"
-#include "webrtc/modules/audio_coding/neteq/normal.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
-#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
-#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/rtc_base/sanitizer.h"
-#include "webrtc/rtc_base/trace_event.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/buffer_level_filter.h"
+#include "modules/audio_coding/neteq/comfort_noise.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/defines.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "modules/audio_coding/neteq/delay_peak_detector.h"
+#include "modules/audio_coding/neteq/dtmf_buffer.h"
+#include "modules/audio_coding/neteq/dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/merge.h"
+#include "modules/audio_coding/neteq/nack_tracker.h"
+#include "modules/audio_coding/neteq/normal.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/safe_conversions.h"
+#include "rtc_base/sanitizer.h"
+#include "rtc_base/trace_event.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index ea10558..72d909a 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -8,26 +8,26 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
#include <memory>
#include <string>
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList.
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
-#include "webrtc/modules/audio_coding/neteq/rtcp.h"
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/typedefs.h"
+#include "api/optional.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/defines.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/packet.h" // Declare PacketList.
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/rtcp.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/criticalsection.h"
+#include "rtc_base/thread_annotations.h"
+#include "typedefs.h"
namespace webrtc {
@@ -446,4 +446,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index fafc2df..c7717c8 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -10,29 +10,29 @@
#include <memory>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/accelerate.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
-#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
-#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder.h"
-#include "webrtc/test/mock_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_delay_manager.h"
+#include "modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
+#include "modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
+#include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
+#include "modules/audio_coding/neteq/mock/mock_packet_buffer.h"
+#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
+#include "modules/audio_coding/neteq/neteq_impl.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/safe_conversions.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder.h"
+#include "test/mock_audio_decoder_factory.h"
using ::testing::AtLeast;
using ::testing::Return;
diff --git a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
index bfa9362..de2d9e8 100644
--- a/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
@@ -10,11 +10,11 @@
#include <memory>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/test/gmock.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "test/gmock.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
index 7c25dd4..ca3dde2 100644
--- a/modules/audio_coding/neteq/neteq_stereo_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_stereo_unittest.cc
@@ -15,15 +15,15 @@
#include <string>
#include <list>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 303973c..010fe55 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
#include <math.h>
#include <stdlib.h>
@@ -20,27 +20,27 @@
#include <string>
#include <vector>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/rtc_base/ignore_wundef.h"
-#include "webrtc/rtc_base/protobuf_utils.h"
-#include "webrtc/rtc_base/sha1digest.h"
-#include "webrtc/rtc_base/stringencode.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/flags.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/protobuf_utils.h"
+#include "rtc_base/sha1digest.h"
+#include "rtc_base/stringencode.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
#else
-#include "webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
#endif
diff --git a/modules/audio_coding/neteq/normal.cc b/modules/audio_coding/neteq/normal.cc
index 551fd9b..48d723a 100644
--- a/modules/audio_coding/neteq/normal.cc
+++ b/modules/audio_coding/neteq/normal.cc
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/normal.h"
+#include "modules/audio_coding/neteq/normal.h"
#include <string.h> // memset, memcpy
#include <algorithm> // min
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/rtc_base/checks.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/normal.h b/modules/audio_coding/neteq/normal.h
index c764155..13cd333 100644
--- a/modules/audio_coding/neteq/normal.h
+++ b/modules/audio_coding/neteq/normal.h
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+#define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
#include <string.h> // Access to size_t.
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/defines.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/safe_conversions.h"
+#include "typedefs.h"
namespace webrtc {
@@ -72,4 +72,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
diff --git a/modules/audio_coding/neteq/normal_unittest.cc b/modules/audio_coding/neteq/normal_unittest.cc
index 1044c4b..b0655d9 100644
--- a/modules/audio_coding/neteq/normal_unittest.cc
+++ b/modules/audio_coding/neteq/normal_unittest.cc
@@ -10,21 +10,21 @@
// Unit tests for Normal class.
-#include "webrtc/modules/audio_coding/neteq/normal.h"
+#include "modules/audio_coding/neteq/normal.h"
#include <memory>
#include <vector>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/expand.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_expand.h"
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/test/gtest.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/expand.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_expand.h"
+#include "modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "test/gtest.h"
using ::testing::_;
using ::testing::Invoke;
diff --git a/modules/audio_coding/neteq/packet.cc b/modules/audio_coding/neteq/packet.cc
index f25f81a..3cec310 100644
--- a/modules/audio_coding/neteq/packet.cc
+++ b/modules/audio_coding/neteq/packet.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/packet.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/packet.h b/modules/audio_coding/neteq/packet.h
index 52a9a13..3772f5c 100644
--- a/modules/audio_coding/neteq/packet.h
+++ b/modules/audio_coding/neteq/packet.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_H_
#include <list>
#include <memory>
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "rtc_base/buffer.h"
+#include "typedefs.h"
namespace webrtc {
@@ -121,4 +121,4 @@
typedef std::list<Packet> PacketList;
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_H_
diff --git a/modules/audio_coding/neteq/packet_buffer.cc b/modules/audio_coding/neteq/packet_buffer.cc
index e0e2e9a..b670c69 100644
--- a/modules/audio_coding/neteq/packet_buffer.cc
+++ b/modules/audio_coding/neteq/packet_buffer.cc
@@ -12,15 +12,15 @@
// an STL list. The list is kept sorted at all times so that the next packet to
// decode is at the beginning of the list.
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
#include <algorithm> // find_if()
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/rtc_base/logging.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "rtc_base/logging.h"
namespace webrtc {
namespace {
diff --git a/modules/audio_coding/neteq/packet_buffer.h b/modules/audio_coding/neteq/packet_buffer.h
index 5c2499c..a7e54f7 100644
--- a/modules/audio_coding/neteq/packet_buffer.h
+++ b/modules/audio_coding/neteq/packet_buffer.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "api/optional.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -145,4 +145,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
diff --git a/modules/audio_coding/neteq/packet_buffer_unittest.cc b/modules/audio_coding/neteq/packet_buffer_unittest.cc
index 7dcf6c4..4beead6 100644
--- a/modules/audio_coding/neteq/packet_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/packet_buffer_unittest.cc
@@ -10,14 +10,14 @@
// Unit tests for PacketBuffer class.
-#include "webrtc/modules/audio_coding/neteq/packet_buffer.h"
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
using ::testing::Return;
using ::testing::StrictMock;
diff --git a/modules/audio_coding/neteq/post_decode_vad.cc b/modules/audio_coding/neteq/post_decode_vad.cc
index 714073a..a09d18f 100644
--- a/modules/audio_coding/neteq/post_decode_vad.cc
+++ b/modules/audio_coding/neteq/post_decode_vad.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/post_decode_vad.h b/modules/audio_coding/neteq/post_decode_vad.h
index a98992f..649d20f 100644
--- a/modules/audio_coding/neteq/post_decode_vad.h
+++ b/modules/audio_coding/neteq/post_decode_vad.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+#define MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
#include <string> // size_t
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/common_audio/vad/include/webrtc_vad.h"
-#include "webrtc/common_types.h" // NULL
-#include "webrtc/modules/audio_coding/neteq/defines.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_audio/vad/include/webrtc_vad.h"
+#include "common_types.h" // NULL
+#include "modules/audio_coding/neteq/defines.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -69,4 +69,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
diff --git a/modules/audio_coding/neteq/post_decode_vad_unittest.cc b/modules/audio_coding/neteq/post_decode_vad_unittest.cc
index 231d7f1..da3e4e8 100644
--- a/modules/audio_coding/neteq/post_decode_vad_unittest.cc
+++ b/modules/audio_coding/neteq/post_decode_vad_unittest.cc
@@ -10,9 +10,9 @@
// Unit tests for PostDecodeVad class.
-#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
+#include "modules/audio_coding/neteq/post_decode_vad.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/preemptive_expand.cc b/modules/audio_coding/neteq/preemptive_expand.cc
index f51a5bd..bc75389 100644
--- a/modules/audio_coding/neteq/preemptive_expand.cc
+++ b/modules/audio_coding/neteq/preemptive_expand.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
#include <algorithm> // min, max
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/preemptive_expand.h b/modules/audio_coding/neteq/preemptive_expand.h
index 1275a37..5af0078 100644
--- a/modules/audio_coding/neteq/preemptive_expand.h
+++ b/modules/audio_coding/neteq/preemptive_expand.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
+#define MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
#include <assert.h>
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/audio_coding/neteq/time_stretch.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -85,4 +85,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_PREEMPTIVE_EXPAND_H_
diff --git a/modules/audio_coding/neteq/random_vector.cc b/modules/audio_coding/neteq/random_vector.cc
index b12f217..c2df8cf 100644
--- a/modules/audio_coding/neteq/random_vector.cc
+++ b/modules/audio_coding/neteq/random_vector.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/random_vector.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/random_vector.h b/modules/audio_coding/neteq/random_vector.h
index 394e940..1434493 100644
--- a/modules/audio_coding/neteq/random_vector.h
+++ b/modules/audio_coding/neteq/random_vector.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
#include <string.h> // size_t
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -47,4 +47,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
diff --git a/modules/audio_coding/neteq/random_vector_unittest.cc b/modules/audio_coding/neteq/random_vector_unittest.cc
index ca0a378..44479a6 100644
--- a/modules/audio_coding/neteq/random_vector_unittest.cc
+++ b/modules/audio_coding/neteq/random_vector_unittest.cc
@@ -10,9 +10,9 @@
// Unit tests for RandomVector class.
-#include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "modules/audio_coding/neteq/random_vector.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/red_payload_splitter.cc b/modules/audio_coding/neteq/red_payload_splitter.cc
index daf24e7..74822dd 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
#include <assert.h>
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/red_payload_splitter.h b/modules/audio_coding/neteq/red_payload_splitter.h
index 8134bfc..1475b1b 100644
--- a/modules/audio_coding/neteq/red_payload_splitter.h
+++ b/modules/audio_coding/neteq/red_payload_splitter.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
+#define MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -48,4 +48,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_RED_PAYLOAD_SPLITTER_H_
diff --git a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
index 4a1fb9e..7d97210 100644
--- a/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
+++ b/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
@@ -10,18 +10,18 @@
// Unit tests for RedPayloadSplitter class.
-#include "webrtc/modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/red_payload_splitter.h"
#include <assert.h>
#include <memory>
#include <utility> // pair
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
using ::testing::Return;
using ::testing::ReturnNull;
diff --git a/modules/audio_coding/neteq/rtcp.cc b/modules/audio_coding/neteq/rtcp.cc
index 3f8ef0e..2885398 100644
--- a/modules/audio_coding/neteq/rtcp.cc
+++ b/modules/audio_coding/neteq/rtcp.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/rtcp.h"
+#include "modules/audio_coding/neteq/rtcp.h"
#include <stdlib.h>
#include <string.h>
#include <algorithm>
-#include "webrtc/modules/include/module_common_types.h"
+#include "modules/include/module_common_types.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/rtcp.h b/modules/audio_coding/neteq/rtcp.h
index ae0ed8d..92b2e2c 100644
--- a/modules/audio_coding/neteq/rtcp.h
+++ b/modules/audio_coding/neteq/rtcp.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_RTCP_H_
+#define MODULES_AUDIO_CODING_NETEQ_RTCP_H_
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -55,4 +55,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_RTCP_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_RTCP_H_
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index e47f92f..163cfff 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include <assert.h>
#include <string.h> // memset
#include <algorithm>
-#include "webrtc/modules/audio_coding/neteq/decision_logic.h"
-#include "webrtc/modules/audio_coding/neteq/delay_manager.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/system_wrappers/include/metrics.h"
+#include "modules/audio_coding/neteq/decision_logic.h"
+#include "modules/audio_coding/neteq/delay_manager.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
+#include "system_wrappers/include/metrics.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index f261a66..a3f0715 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
#include <deque>
#include <string>
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -182,4 +182,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_STATISTICS_CALCULATOR_H_
diff --git a/modules/audio_coding/neteq/sync_buffer.cc b/modules/audio_coding/neteq/sync_buffer.cc
index 49f8013..28d7649 100644
--- a/modules/audio_coding/neteq/sync_buffer.cc
+++ b/modules/audio_coding/neteq/sync_buffer.cc
@@ -10,8 +10,8 @@
#include <algorithm> // Access to min.
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/sync_buffer.h b/modules/audio_coding/neteq/sync_buffer.h
index a3a78f5..98a606b 100644
--- a/modules/audio_coding/neteq/sync_buffer.h
+++ b/modules/audio_coding/neteq/sync_buffer.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -98,4 +98,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
diff --git a/modules/audio_coding/neteq/sync_buffer_unittest.cc b/modules/audio_coding/neteq/sync_buffer_unittest.cc
index cbf26e0..f3f7895 100644
--- a/modules/audio_coding/neteq/sync_buffer_unittest.cc
+++ b/modules/audio_coding/neteq/sync_buffer_unittest.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
+#include "modules/audio_coding/neteq/sync_buffer.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
index 56ed72f..0526546 100644
--- a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
@@ -13,8 +13,8 @@
#include <map>
#include <stdio.h>
-#include "webrtc/typedefs.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "typedefs.h"
+#include "modules/include/module_common_types.h"
enum stereoModes {
stereoModeMono,
diff --git a/modules/audio_coding/neteq/test/RTPchange.cc b/modules/audio_coding/neteq/test/RTPchange.cc
index 54395c0..3e90002 100644
--- a/modules/audio_coding/neteq/test/RTPchange.cc
+++ b/modules/audio_coding/neteq/test/RTPchange.cc
@@ -13,8 +13,8 @@
#include <algorithm>
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h"
-#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#define FIRSTLINELEN 40
//#define WEBRTC_DUMMY_RTP
diff --git a/modules/audio_coding/neteq/test/RTPencode.cc b/modules/audio_coding/neteq/test/RTPencode.cc
index 8ad5090..249fbaa 100644
--- a/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/modules/audio_coding/neteq/test/RTPencode.cc
@@ -25,12 +25,12 @@
#include <algorithm>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/checks.h"
+#include "typedefs.h"
// needed for NetEqDecoder
-#include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/audio_decoder_impl.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
/************************/
/* Define payload types */
@@ -132,10 +132,10 @@
#include "webrtc_vad.h"
#if ((defined CODEC_PCM16B) || (defined NETEQ_ARBITRARY_CODEC))
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#endif
#ifdef CODEC_G711
-#include "webrtc/modules/audio_coding/codecs/g711/g711_interface.h"
+#include "modules/audio_coding/codecs/g711/g711_interface.h"
#endif
#ifdef CODEC_G729
#include "G729Interface.h"
@@ -152,19 +152,19 @@
#include "AMRWBCreation.h"
#endif
#ifdef CODEC_ILBC
-#include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
#endif
#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
-#include "webrtc/modules/audio_coding/codecs/isac/main/include/isac.h"
+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
#endif
#ifdef NETEQ_ISACFIX_CODEC
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
#ifdef CODEC_ISAC
#error Cannot have both ISAC and ISACfix defined. Please de-select one.
#endif
#endif
#ifdef CODEC_G722
-#include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
+#include "modules/audio_coding/codecs/g722/g722_interface.h"
#endif
#ifdef CODEC_G722_1_24
#include "G722_1Interface.h"
@@ -194,10 +194,10 @@
#endif
#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-#include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
#endif
#ifdef CODEC_OPUS
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
#endif
/***********************************/
diff --git a/modules/audio_coding/neteq/test/RTPjitter.cc b/modules/audio_coding/neteq/test/RTPjitter.cc
index 391f051..baaf8f0 100644
--- a/modules/audio_coding/neteq/test/RTPjitter.cc
+++ b/modules/audio_coding/neteq/test/RTPjitter.cc
@@ -25,8 +25,8 @@
#include <assert.h>
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
/*********************/
/* Misc. definitions */
diff --git a/modules/audio_coding/neteq/test/RTPtimeshift.cc b/modules/audio_coding/neteq/test/RTPtimeshift.cc
index 3b7a53c..a2429f1 100644
--- a/modules/audio_coding/neteq/test/RTPtimeshift.cc
+++ b/modules/audio_coding/neteq/test/RTPtimeshift.cc
@@ -12,8 +12,8 @@
#include <algorithm>
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
+#include "test/gtest.h"
#define FIRSTLINELEN 40
diff --git a/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
index 2f8ca1b..c18133a 100644
--- a/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -10,12 +10,12 @@
#include <memory>
-#include "webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/flags.h"
+#include "rtc_base/safe_conversions.h"
+#include "test/testsupport/fileutils.h"
using testing::InitGoogleTest;
diff --git a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
index 5a73a6a..d88f789 100644
--- a/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
-#include "webrtc/rtc_base/flags.h"
+#include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/flags.h"
using testing::InitGoogleTest;
diff --git a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
index f4edf37..c2542b6 100644
--- a/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_inst.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
-#include "webrtc/rtc_base/flags.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/flags.h"
using testing::InitGoogleTest;
diff --git a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
index 5ea9056..c91d6f7 100644
--- a/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -10,12 +10,12 @@
#include <memory>
-#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/flags.h"
+#include "rtc_base/safe_conversions.h"
+#include "test/testsupport/fileutils.h"
using testing::InitGoogleTest;
diff --git a/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
index e64def4..f74360e 100644
--- a/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
+++ b/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/typedefs.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "test/gtest.h"
+#include "test/testsupport/perf_test.h"
+#include "typedefs.h"
+#include "system_wrappers/include/field_trial.h"
// Runs a test with 10% packet losses and 10% clock drift, to exercise
// both loss concealment and time-stretching code.
diff --git a/modules/audio_coding/neteq/test/neteq_speed_test.cc b/modules/audio_coding/neteq/test/neteq_speed_test.cc
index c58381f..1e8c1ea 100644
--- a/modules/audio_coding/neteq/test/neteq_speed_test.cc
+++ b/modules/audio_coding/neteq/test/neteq_speed_test.cc
@@ -12,10 +12,10 @@
#include <iostream>
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "rtc_base/flags.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
// Define command line flags.
DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
diff --git a/modules/audio_coding/neteq/tick_timer.cc b/modules/audio_coding/neteq/tick_timer.cc
index 4a1b9b7..17f83b1 100644
--- a/modules/audio_coding/neteq/tick_timer.cc
+++ b/modules/audio_coding/neteq/tick_timer.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/tick_timer.h b/modules/audio_coding/neteq/tick_timer.h
index 55e9eab..93f9ee4 100644
--- a/modules/audio_coding/neteq/tick_timer.h
+++ b/modules/audio_coding/neteq/tick_timer.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
#include <memory>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -107,4 +107,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TICK_TIMER_H_
diff --git a/modules/audio_coding/neteq/tick_timer_unittest.cc b/modules/audio_coding/neteq/tick_timer_unittest.cc
index 6f338d1..875f04d 100644
--- a/modules/audio_coding/neteq/tick_timer_unittest.cc
+++ b/modules/audio_coding/neteq/tick_timer_unittest.cc
@@ -10,10 +10,10 @@
#include <memory>
-#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/tick_timer.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/time_stretch.cc b/modules/audio_coding/neteq/time_stretch.cc
index 630051b..d0ea68a 100644
--- a/modules/audio_coding/neteq/time_stretch.cc
+++ b/modules/audio_coding/neteq/time_stretch.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/time_stretch.h"
+#include "modules/audio_coding/neteq/time_stretch.h"
#include <algorithm> // min, max
#include <memory>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/cross_correlation.h"
-#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/cross_correlation.h"
+#include "modules/audio_coding/neteq/dsp_helper.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/time_stretch.h b/modules/audio_coding/neteq/time_stretch.h
index ff056e6..5192cbd 100644
--- a/modules/audio_coding/neteq/time_stretch.h
+++ b/modules/audio_coding/neteq/time_stretch.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+#define MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
#include <assert.h>
#include <string.h> // memset, size_t
-#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/audio_multi_vector.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -113,4 +113,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_
diff --git a/modules/audio_coding/neteq/time_stretch_unittest.cc b/modules/audio_coding/neteq/time_stretch_unittest.cc
index f549b27..8d0f4d4 100644
--- a/modules/audio_coding/neteq/time_stretch_unittest.cc
+++ b/modules/audio_coding/neteq/time_stretch_unittest.cc
@@ -10,18 +10,18 @@
// Unit tests for Accelerate and PreemptiveExpand classes.
-#include "webrtc/modules/audio_coding/neteq/accelerate.h"
-#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/accelerate.h"
+#include "modules/audio_coding/neteq/preemptive_expand.h"
#include <map>
#include <memory>
-#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
-#include "webrtc/modules/audio_coding/neteq/background_noise.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "modules/audio_coding/neteq/background_noise.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/checks.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/timestamp_scaler.cc b/modules/audio_coding/neteq/timestamp_scaler.cc
index f246cbe..3b67a38 100644
--- a/modules/audio_coding/neteq/timestamp_scaler.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/modules/audio_coding/neteq/decoder_database.h"
+#include "modules/audio_coding/neteq/decoder_database.h"
namespace webrtc {
diff --git a/modules/audio_coding/neteq/timestamp_scaler.h b/modules/audio_coding/neteq/timestamp_scaler.h
index bfb8d9c..fa67717 100644
--- a/modules/audio_coding/neteq/timestamp_scaler.h
+++ b/modules/audio_coding/neteq/timestamp_scaler.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -65,4 +65,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TIMESTAMP_SCALER_H_
diff --git a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
index 13c4554..b3c1bb0 100644
--- a/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
+++ b/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
-#include "webrtc/modules/audio_coding/neteq/packet.h"
-#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
+#include "modules/audio_coding/neteq/packet.h"
+#include "modules/audio_coding/neteq/timestamp_scaler.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
using ::testing::Return;
using ::testing::ReturnNull;
diff --git a/modules/audio_coding/neteq/tools/DEPS b/modules/audio_coding/neteq/tools/DEPS
index 0f16a4f..4db1e1d 100644
--- a/modules/audio_coding/neteq/tools/DEPS
+++ b/modules/audio_coding/neteq/tools/DEPS
@@ -1,3 +1,3 @@
include_rules = [
- "+webrtc/logging/rtc_event_log",
+ "+logging/rtc_event_log",
]
diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h
index 48fde65..9ec8358 100644
--- a/modules/audio_coding/neteq/tools/audio_checksum.h
+++ b/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
#include <string>
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/md5digest.h"
-#include "webrtc/rtc_base/stringencode.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/md5digest.h"
+#include "rtc_base/stringencode.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -56,4 +56,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
diff --git a/modules/audio_coding/neteq/tools/audio_loop.cc b/modules/audio_coding/neteq/tools/audio_loop.cc
index 56765aa..b5ad881 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.cc
+++ b/modules/audio_coding/neteq/tools/audio_loop.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include <assert.h>
#include <stdio.h>
diff --git a/modules/audio_coding/neteq/tools/audio_loop.h b/modules/audio_coding/neteq/tools/audio_loop.h
index d226df4..b26f310 100644
--- a/modules/audio_coding/neteq/tools/audio_loop.h
+++ b/modules/audio_coding/neteq/tools/audio_loop.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
#include <memory>
#include <string>
-#include "webrtc/api/array_view.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "api/array_view.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -56,4 +56,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_LOOP_H_
diff --git a/modules/audio_coding/neteq/tools/audio_sink.cc b/modules/audio_coding/neteq/tools/audio_sink.cc
index 665dc41..7d7af7e 100644
--- a/modules/audio_coding/neteq/tools/audio_sink.cc
+++ b/modules/audio_coding/neteq/tools/audio_sink.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h
index 076b408..17b9e67 100644
--- a/modules/audio_coding/neteq/tools/audio_sink.h
+++ b/modules/audio_coding/neteq/tools/audio_sink.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -68,4 +68,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
index 29e3402..6b325b6 100644
--- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
+++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
+#include "modules/audio_coding/neteq/tools/constant_pcm_packet_source.h"
#include <algorithm>
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
index 1250333..7c5aa67 100644
--- a/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
+++ b/modules/audio_coding/neteq/tools/constant_pcm_packet_source.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
#include <stdio.h>
#include <string>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace test {
@@ -52,4 +52,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_CONSTANT_PCM_PACKET_SOURCE_H_
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
index 41bed7c..d04e818 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h"
+#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
#include <utility>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/encode_neteq_input.h b/modules/audio_coding/neteq/tools/encode_neteq_input.h
index df78e2a..b44d4ac 100644
--- a/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ b/modules/audio_coding/neteq/tools/encode_neteq_input.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
#include <memory>
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/include/module_common_types.h"
namespace webrtc {
namespace test {
@@ -68,4 +68,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
index eef32df..6779e5e 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/safe_conversions.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/safe_conversions.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
index 09c2e78..7aa8e6e 100644
--- a/modules/audio_coding/neteq/tools/fake_decode_from_file.h
+++ b/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
#include <memory>
-#include "webrtc/api/array_view.h"
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "api/array_view.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "api/optional.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
namespace webrtc {
namespace test {
@@ -70,4 +70,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
diff --git a/modules/audio_coding/neteq/tools/input_audio_file.cc b/modules/audio_coding/neteq/tools/input_audio_file.cc
index 97d20bc..8c8b72c 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file.cc
+++ b/modules/audio_coding/neteq/tools/input_audio_file.cc
@@ -8,9 +8,9 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/input_audio_file.h b/modules/audio_coding/neteq/tools/input_audio_file.h
index 271c513..a0ba7f6 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/input_audio_file.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
#include <stdio.h>
#include <string>
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -55,4 +55,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_INPUT_AUDIO_FILE_H_
diff --git a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
index ff795d8..e0ee265 100644
--- a/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
+++ b/modules/audio_coding/neteq/tools/input_audio_file_unittest.cc
@@ -10,9 +10,9 @@
// Unit tests for test InputAudioFile class.
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
index a0f5aa7..72a539f 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
#include <algorithm>
#include <fstream>
@@ -17,7 +17,7 @@
#include <limits>
#include <utility>
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
index e142a3b..cbac836 100644
--- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
+++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
#include <map>
#include <set>
#include <string>
#include <vector>
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
-#include "webrtc/typedefs.h"
+#include "api/optional.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -68,4 +68,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_DELAY_ANALYZER_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 8d88b08..68dde52 100644
--- a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -9,11 +9,11 @@
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
+#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/test/gtest.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "rtc_base/format_macros.h"
+#include "test/gtest.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
index fc66c0f..61e256e 100644
--- a/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
#include <memory>
#include <string>
-#include "webrtc/api/audio_codecs/audio_decoder.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/include/module_common_types.h"
+#include "api/audio_codecs/audio_decoder.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/include/module_common_types.h"
namespace webrtc {
namespace test {
@@ -62,4 +62,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_input.cc b/modules/audio_coding/neteq/tools/neteq_input.cc
index 752bb29..44513ab 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_input.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
#include <sstream>
diff --git a/modules/audio_coding/neteq/tools/neteq_input.h b/modules/audio_coding/neteq/tools/neteq_input.h
index 08f554b..cf28386 100644
--- a/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_input.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
#include <algorithm>
#include <memory>
#include <string>
-#include "webrtc/api/optional.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/rtc_base/buffer.h"
+#include "api/optional.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "rtc_base/buffer.h"
namespace webrtc {
namespace test {
@@ -80,4 +80,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
index 13a061f..12a12c4 100644
--- a/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include <algorithm>
#include <limits>
-#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_packet_source_input.h b/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
index 35b54fa..b482556 100644
--- a/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
#include <map>
#include <string>
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
namespace test {
@@ -81,4 +81,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 8d0cf90..d3de4de 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_loop.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
using webrtc::NetEq;
using webrtc::test::AudioLoop;
diff --git a/modules/audio_coding/neteq/tools/neteq_performance_test.h b/modules/audio_coding/neteq/tools/neteq_performance_test.h
index d094db0..a02e40c 100644
--- a/modules/audio_coding/neteq/tools/neteq_performance_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_performance_test.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
-#include "webrtc/typedefs.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -29,4 +29,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 85adb59..7df46ff 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -11,13 +11,13 @@
#include <math.h>
#include <stdio.h>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.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/resample_input_audio_file.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.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/resample_input_audio_file.h"
+#include "rtc_base/checks.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_quality_test.h b/modules/audio_coding/neteq/tools/neteq_quality_test.h
index c1964b6..f920191 100644
--- a/modules/audio_coding/neteq/tools/neteq_quality_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_quality_test.h
@@ -8,21 +8,21 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
#include <fstream>
#include <memory>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/flags.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -137,4 +137,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_QUALITY_TEST_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
index 553c71f..a3e3413 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
-#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/modules/audio_coding/neteq/tools/neteq_replacement_input.h
index ee1e4ae..3a89399 100644
--- a/modules/audio_coding/neteq/tools/neteq_replacement_input.h
+++ b/modules/audio_coding/neteq/tools/neteq_replacement_input.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
#include <memory>
#include <set>
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
namespace webrtc {
namespace test {
@@ -48,4 +48,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_REPLACEMENT_INPUT_H_
diff --git a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
index d6647e4..eb057cc 100644
--- a/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
+++ b/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
@@ -22,21 +22,21 @@
#include <numeric>
#include <string>
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.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/rtp_file_source.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.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/rtp_file_source.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/flags.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_test.cc b/modules/audio_coding/neteq/tools/neteq_test.cc
index 598ffd7..39d9549 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.cc
+++ b/modules/audio_coding/neteq/tools/neteq_test.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
+#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include <iostream>
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/neteq_test.h b/modules/audio_coding/neteq/tools/neteq_test.h
index 18fad9a..2c0a07c 100644
--- a/modules/audio_coding/neteq/tools/neteq_test.h
+++ b/modules/audio_coding/neteq/tools/neteq_test.h
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
#include <map>
#include <memory>
#include <string>
#include <utility>
-#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
+#include "modules/audio_coding/neteq/include/neteq.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "modules/audio_coding/neteq/tools/neteq_input.h"
namespace webrtc {
namespace test {
@@ -103,4 +103,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_TEST_H_
diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h
index 7934470..7e65bc2 100644
--- a/modules/audio_coding/neteq/tools/output_audio_file.h
+++ b/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
#include <assert.h>
#include <stdio.h>
#include <string>
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace test {
@@ -47,4 +47,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
diff --git a/modules/audio_coding/neteq/tools/output_wav_file.h b/modules/audio_coding/neteq/tools/output_wav_file.h
index 7b3431d..031a8cb 100644
--- a/modules/audio_coding/neteq/tools/output_wav_file.h
+++ b/modules/audio_coding/neteq/tools/output_wav_file.h
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
#include <string>
-#include "webrtc/common_audio/wav_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "common_audio/wav_file.h"
+#include "modules/audio_coding/neteq/tools/audio_sink.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace test {
@@ -40,4 +40,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_WAV_FILE_H_
diff --git a/modules/audio_coding/neteq/tools/packet.cc b/modules/audio_coding/neteq/tools/packet.cc
index b5b1b2f..71337b6 100644
--- a/modules/audio_coding/neteq/tools/packet.cc
+++ b/modules/audio_coding/neteq/tools/packet.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
#include <string.h>
#include <memory>
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/checks.h"
+#include "modules/include/module_common_types.h"
+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/packet.h b/modules/audio_coding/neteq/tools/packet.h
index ce9ef5a..583ef56 100644
--- a/modules/audio_coding/neteq/tools/packet.h
+++ b/modules/audio_coding/neteq/tools/packet.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
#include <list>
#include <memory>
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
@@ -114,4 +114,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_H_
diff --git a/modules/audio_coding/neteq/tools/packet_source.cc b/modules/audio_coding/neteq/tools/packet_source.cc
index d6cb37e..30bf431 100644
--- a/modules/audio_coding/neteq/tools/packet_source.cc
+++ b/modules/audio_coding/neteq/tools/packet_source.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/packet_source.h b/modules/audio_coding/neteq/tools/packet_source.h
index 233ddec..3e825d4 100644
--- a/modules/audio_coding/neteq/tools/packet_source.h
+++ b/modules/audio_coding/neteq/tools/packet_source.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
#include <bitset>
#include <memory>
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -47,4 +47,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
diff --git a/modules/audio_coding/neteq/tools/packet_unittest.cc b/modules/audio_coding/neteq/tools/packet_unittest.cc
index 940276b..ce6a3b9 100644
--- a/modules/audio_coding/neteq/tools/packet_unittest.cc
+++ b/modules/audio_coding/neteq/tools/packet_unittest.cc
@@ -10,9 +10,9 @@
// Unit tests for test Packet class.
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/test/gtest.h"
+#include "test/gtest.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.cc b/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
index 45c5b20..5050e1f 100644
--- a/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
+++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.cc
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
#include <memory>
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/modules/audio_coding/neteq/tools/resample_input_audio_file.h
index b008d19..0099979 100644
--- a/modules/audio_coding/neteq/tools/resample_input_audio_file.h
+++ b/modules/audio_coding/neteq/tools/resample_input_audio_file.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
#include <string>
-#include "webrtc/common_audio/resampler/include/resampler.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "common_audio/resampler/include/resampler.h"
+#include "modules/audio_coding/neteq/tools/input_audio_file.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -48,4 +48,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RESAMPLE_INPUT_AUDIO_FILE_H_
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index f54f91e..1603ee8 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
#include <assert.h>
#include <string.h>
#include <iostream>
#include <limits>
-#include "webrtc/call/call.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/checks.h"
+#include "call/call.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
index 15e3ce7..df01e06 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.h
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
#include <memory>
#include <string>
-#include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "logging/rtc_event_log/rtc_event_log_parser.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -62,4 +62,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
diff --git a/modules/audio_coding/neteq/tools/rtp_analyze.cc b/modules/audio_coding/neteq/tools/rtp_analyze.cc
index 23f96c5..12721cc 100644
--- a/modules/audio_coding/neteq/tools/rtp_analyze.cc
+++ b/modules/audio_coding/neteq/tools/rtp_analyze.cc
@@ -14,9 +14,9 @@
#include <memory>
#include <vector>
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
-#include "webrtc/rtc_base/flags.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/flags.h"
// Define command line flags.
DEFINE_int(red, 117, "RTP payload type for RED");
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.cc b/modules/audio_coding/neteq/tools/rtp_file_source.cc
index 5ea8de9..c9ae5f2 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.cc
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include <assert.h>
#include <string.h>
@@ -20,10 +20,10 @@
#include <memory>
-#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/rtp_file_reader.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "rtc_base/checks.h"
+#include "test/rtp_file_reader.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/rtp_file_source.h b/modules/audio_coding/neteq/tools/rtp_file_source.h
index 9b0160f..6daa748 100644
--- a/modules/audio_coding/neteq/tools/rtp_file_source.h
+++ b/modules/audio_coding/neteq/tools/rtp_file_source.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
#include <stdio.h>
#include <memory>
#include <string>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "common_types.h"
+#include "modules/audio_coding/neteq/tools/packet_source.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
@@ -63,4 +63,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.cc b/modules/audio_coding/neteq/tools/rtp_generator.cc
index a6e883d..cedd7ae 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.cc
+++ b/modules/audio_coding/neteq/tools/rtp_generator.cc
@@ -10,7 +10,7 @@
#include <assert.h>
-#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "modules/audio_coding/neteq/tools/rtp_generator.h"
namespace webrtc {
namespace test {
diff --git a/modules/audio_coding/neteq/tools/rtp_generator.h b/modules/audio_coding/neteq/tools/rtp_generator.h
index 154a38c..75dcf03 100644
--- a/modules/audio_coding/neteq/tools/rtp_generator.h
+++ b/modules/audio_coding/neteq/tools/rtp_generator.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "rtc_base/constructormagic.h"
+#include "typedefs.h"
namespace webrtc {
namespace test {
@@ -80,4 +80,4 @@
} // namespace test
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
+#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_GENERATOR_H_
diff --git a/modules/audio_coding/neteq/tools/rtpcat.cc b/modules/audio_coding/neteq/tools/rtpcat.cc
index d903ca2..431de55 100644
--- a/modules/audio_coding/neteq/tools/rtpcat.cc
+++ b/modules/audio_coding/neteq/tools/rtpcat.cc
@@ -12,9 +12,9 @@
#include <memory>
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/test/rtp_file_reader.h"
-#include "webrtc/test/rtp_file_writer.h"
+#include "rtc_base/checks.h"
+#include "test/rtp_file_reader.h"
+#include "test/rtp_file_writer.h"
using webrtc::test::RtpFileReader;
using webrtc::test::RtpFileWriter;
diff --git a/modules/audio_coding/test/ACMTest.h b/modules/audio_coding/test/ACMTest.h
index d7e87d3..3fc97ca 100644
--- a/modules/audio_coding/test/ACMTest.h
+++ b/modules/audio_coding/test/ACMTest.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#define MODULES_AUDIO_CODING_TEST_ACMTEST_H_
class ACMTest {
public:
@@ -18,4 +18,4 @@
virtual void Perform() = 0;
};
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ACMTEST_H_
+#endif // MODULES_AUDIO_CODING_TEST_ACMTEST_H_
diff --git a/modules/audio_coding/test/APITest.cc b/modules/audio_coding/test/APITest.cc
index 7e67e3f..fe94e59 100644
--- a/modules/audio_coding/test/APITest.cc
+++ b/modules/audio_coding/test/APITest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/APITest.h"
+#include "modules/audio_coding/test/APITest.h"
#include <ctype.h>
#include <stdio.h>
@@ -19,16 +19,16 @@
#include <ostream>
#include <string>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/rtc_base/platform_thread.h"
-#include "webrtc/rtc_base/timeutils.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/test/utility.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/timeutils.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "system_wrappers/include/trace.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/APITest.h b/modules/audio_coding/test/APITest.h
index 99a7201..652d7c4 100644
--- a/modules/audio_coding/test/APITest.h
+++ b/modules/audio_coding/test/APITest.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_APITEST_H_
+#define MODULES_AUDIO_CODING_TEST_APITEST_H_
#include <memory>
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "system_wrappers/include/rw_lock_wrapper.h"
namespace webrtc {
@@ -159,4 +159,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_APITEST_H_
+#endif // MODULES_AUDIO_CODING_TEST_APITEST_H_
diff --git a/modules/audio_coding/test/Channel.cc b/modules/audio_coding/test/Channel.cc
index 4c65953..7d5e6e2 100644
--- a/modules/audio_coding/test/Channel.cc
+++ b/modules/audio_coding/test/Channel.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/Channel.h"
#include <assert.h>
#include <iostream>
-#include "webrtc/rtc_base/format_macros.h"
-#include "webrtc/rtc_base/timeutils.h"
+#include "rtc_base/format_macros.h"
+#include "rtc_base/timeutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/Channel.h b/modules/audio_coding/test/Channel.h
index aeb535a..f64bf37 100644
--- a/modules/audio_coding/test/Channel.h
+++ b/modules/audio_coding/test/Channel.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#ifndef MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#define MODULES_AUDIO_CODING_TEST_CHANNEL_H_
#include <stdio.h>
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/criticalsection.h"
+#include "typedefs.h"
namespace webrtc {
@@ -126,4 +126,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_CHANNEL_H_
+#endif // MODULES_AUDIO_CODING_TEST_CHANNEL_H_
diff --git a/modules/audio_coding/test/EncodeDecodeTest.cc b/modules/audio_coding/test/EncodeDecodeTest.cc
index 24d0719..1125fbe 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.cc
+++ b/modules/audio_coding/test/EncodeDecodeTest.cc
@@ -8,20 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
#include <memory>
#include <sstream>
#include <stdio.h>
#include <stdlib.h>
-#include "webrtc/common_types.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/test/utility.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/utility.h"
+#include "system_wrappers/include/trace.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/EncodeDecodeTest.h b/modules/audio_coding/test/EncodeDecodeTest.h
index f9a9a5b..941a92f 100644
--- a/modules/audio_coding/test/EncodeDecodeTest.h
+++ b/modules/audio_coding/test/EncodeDecodeTest.h
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
#include <stdio.h>
#include <string.h>
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/RTPFile.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/RTPFile.h"
+#include "typedefs.h"
namespace webrtc {
@@ -120,4 +120,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
+#endif // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_
diff --git a/modules/audio_coding/test/PCMFile.cc b/modules/audio_coding/test/PCMFile.cc
index 03d4fa7..73c8542 100644
--- a/modules/audio_coding/test/PCMFile.cc
+++ b/modules/audio_coding/test/PCMFile.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/PCMFile.h"
#include <ctype.h>
#include <stdio.h>
#include <string.h>
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/test/gtest.h"
+#include "modules/include/module_common_types.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/PCMFile.h b/modules/audio_coding/test/PCMFile.h
index 3576dc6..c140ef2 100644
--- a/modules/audio_coding/test/PCMFile.h
+++ b/modules/audio_coding/test/PCMFile.h
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#ifndef MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#define MODULES_AUDIO_CODING_TEST_PCMFILE_H_
#include <stdio.h>
#include <stdlib.h>
#include <string>
-#include "webrtc/api/optional.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/typedefs.h"
+#include "api/optional.h"
+#include "modules/include/module_common_types.h"
+#include "typedefs.h"
namespace webrtc {
@@ -73,4 +73,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PCMFILE_H_
+#endif // MODULES_AUDIO_CODING_TEST_PCMFILE_H_
diff --git a/modules/audio_coding/test/PacketLossTest.cc b/modules/audio_coding/test/PacketLossTest.cc
index 68f2615..c80615a 100644
--- a/modules/audio_coding/test/PacketLossTest.cc
+++ b/modules/audio_coding/test/PacketLossTest.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
+#include "modules/audio_coding/test/PacketLossTest.h"
#include <memory>
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/PacketLossTest.h b/modules/audio_coding/test/PacketLossTest.h
index 705fe73..7eab442 100644
--- a/modules/audio_coding/test/PacketLossTest.h
+++ b/modules/audio_coding/test/PacketLossTest.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
#include <memory>
#include <string>
-#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
namespace webrtc {
@@ -64,4 +64,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
+#endif // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
diff --git a/modules/audio_coding/test/RTPFile.cc b/modules/audio_coding/test/RTPFile.cc
index 4bf4c98..d896e76 100644
--- a/modules/audio_coding/test/RTPFile.cc
+++ b/modules/audio_coding/test/RTPFile.cc
@@ -20,10 +20,10 @@
#endif
#include "audio_coding_module.h"
-#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
+#include "system_wrappers/include/rw_lock_wrapper.h"
// TODO(tlegrand): Consider removing usage of gtest.
-#include "webrtc/test/gtest.h"
-#include "webrtc/typedefs.h"
+#include "test/gtest.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/RTPFile.h b/modules/audio_coding/test/RTPFile.h
index 696d41e..dd6ee72 100644
--- a/modules/audio_coding/test/RTPFile.h
+++ b/modules/audio_coding/test/RTPFile.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#ifndef MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#define MODULES_AUDIO_CODING_TEST_RTPFILE_H_
#include <stdio.h>
#include <queue>
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/include/module_common_types.h"
+#include "system_wrappers/include/rw_lock_wrapper.h"
+#include "typedefs.h"
namespace webrtc {
@@ -123,4 +123,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_RTPFILE_H_
+#endif // MODULES_AUDIO_CODING_TEST_RTPFILE_H_
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index aedf82c..f840676 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -8,21 +8,21 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
+#include "modules/audio_coding/test/TestAllCodecs.h"
#include <cstdio>
#include <limits>
#include <string>
-#include "webrtc/common_types.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/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/utility.h"
+#include "rtc_base/logging.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
// Description of the test:
// In this test we set up a one-way communication channel from a participant
diff --git a/modules/audio_coding/test/TestAllCodecs.h b/modules/audio_coding/test/TestAllCodecs.h
index 7df139b..cdcc55c 100644
--- a/modules/audio_coding/test/TestAllCodecs.h
+++ b/modules/audio_coding/test/TestAllCodecs.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#ifndef MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#define MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
#include <memory>
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "typedefs.h"
namespace webrtc {
@@ -80,4 +80,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
+#endif // MODULES_AUDIO_CODING_TEST_TESTALLCODECS_H_
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 4ec3ed1..77cc24d 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/TestRedFec.h"
+#include "modules/audio_coding/test/TestRedFec.h"
#include <assert.h>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/utility.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
#ifdef SUPPORT_RED_WB
#undef SUPPORT_RED_WB
diff --git a/modules/audio_coding/test/TestRedFec.h b/modules/audio_coding/test/TestRedFec.h
index 09d9259..98aa008 100644
--- a/modules/audio_coding/test/TestRedFec.h
+++ b/modules/audio_coding/test/TestRedFec.h
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#ifndef MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#define MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
#include <memory>
#include <string>
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
namespace webrtc {
@@ -47,4 +47,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
+#endif // MODULES_AUDIO_CODING_TEST_TESTREDFEC_H_
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index 02bc141..7f24fdc 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "modules/audio_coding/test/TestStereo.h"
#include <assert.h>
#include <string>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/utility.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index 3489421..a27d8d7 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#ifndef MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#define MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
#include <math.h>
#include <memory>
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
#define PCMA_AND_PCMU
@@ -115,4 +115,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
+#endif // MODULES_AUDIO_CODING_TEST_TESTSTEREO_H_
diff --git a/modules/audio_coding/test/TestVADDTX.cc b/modules/audio_coding/test/TestVADDTX.cc
index ad5e066..5282d1b 100644
--- a/modules/audio_coding/test/TestVADDTX.cc
+++ b/modules/audio_coding/test/TestVADDTX.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
+#include "modules/audio_coding/test/TestVADDTX.h"
#include <string>
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/TestVADDTX.h b/modules/audio_coding/test/TestVADDTX.h
index b7e9871..95d6ec4 100644
--- a/modules/audio_coding/test/TestVADDTX.h
+++ b/modules/audio_coding/test/TestVADDTX.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#ifndef MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#define MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
#include <memory>
-#include "webrtc/common_types.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/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
+#include "common_types.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
namespace webrtc {
@@ -100,4 +100,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
+#endif // MODULES_AUDIO_CODING_TEST_TESTVADDTX_H_
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 0d3f3f4..7d58b6d 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -12,20 +12,20 @@
#include <string>
#include <vector>
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/APITest.h"
-#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"
-#include "webrtc/modules/audio_coding/test/PacketLossTest.h"
-#include "webrtc/modules/audio_coding/test/TestAllCodecs.h"
-#include "webrtc/modules/audio_coding/test/TestRedFec.h"
-#include "webrtc/modules/audio_coding/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/test/TestVADDTX.h"
-#include "webrtc/modules/audio_coding/test/TwoWayCommunication.h"
-#include "webrtc/modules/audio_coding/test/iSACTest.h"
-#include "webrtc/modules/audio_coding/test/opus_test.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/APITest.h"
+#include "modules/audio_coding/test/EncodeDecodeTest.h"
+#include "modules/audio_coding/test/PacketLossTest.h"
+#include "modules/audio_coding/test/TestAllCodecs.h"
+#include "modules/audio_coding/test/TestRedFec.h"
+#include "modules/audio_coding/test/TestStereo.h"
+#include "modules/audio_coding/test/TestVADDTX.h"
+#include "modules/audio_coding/test/TwoWayCommunication.h"
+#include "modules/audio_coding/test/iSACTest.h"
+#include "modules/audio_coding/test/opus_test.h"
+#include "system_wrappers/include/trace.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
using webrtc::Trace;
diff --git a/modules/audio_coding/test/TwoWayCommunication.cc b/modules/audio_coding/test/TwoWayCommunication.cc
index 3287c91..6663b1e 100644
--- a/modules/audio_coding/test/TwoWayCommunication.cc
+++ b/modules/audio_coding/test/TwoWayCommunication.cc
@@ -20,14 +20,14 @@
#include <Windows.h>
#endif
-#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/TwoWayCommunication.h b/modules/audio_coding/test/TwoWayCommunication.h
index f9d37f7..fb23275 100644
--- a/modules/audio_coding/test/TwoWayCommunication.h
+++ b/modules/audio_coding/test/TwoWayCommunication.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#ifndef MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#define MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
#include <memory>
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
namespace webrtc {
@@ -58,4 +58,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
+#endif // MODULES_AUDIO_CODING_TEST_TWOWAYCOMMUNICATION_H_
diff --git a/modules/audio_coding/test/delay_test.cc b/modules/audio_coding/test/delay_test.cc
index 0ce7fd2..1f3dc6d 100644
--- a/modules/audio_coding/test/delay_test.cc
+++ b/modules/audio_coding/test/delay_test.cc
@@ -15,18 +15,18 @@
#include <iostream>
#include <memory>
-#include "webrtc/common_types.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/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
+#include "rtc_base/flags.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
DEFINE_string(codec, "isac", "Codec Name");
DEFINE_int(sample_rate_hz, 16000, "Sampling rate in Hertz.");
diff --git a/modules/audio_coding/test/iSACTest.cc b/modules/audio_coding/test/iSACTest.cc
index 531fe96..a14f795 100644
--- a/modules/audio_coding/test/iSACTest.cc
+++ b/modules/audio_coding/test/iSACTest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/iSACTest.h"
+#include "modules/audio_coding/test/iSACTest.h"
#include <ctype.h>
#include <stdio.h>
@@ -23,10 +23,10 @@
#include <time.h>
#endif
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/test/utility.h"
+#include "system_wrappers/include/event_wrapper.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/iSACTest.h b/modules/audio_coding/test/iSACTest.h
index 7d3a77e..4fe3dc7 100644
--- a/modules/audio_coding/test/iSACTest.h
+++ b/modules/audio_coding/test/iSACTest.h
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#define MODULES_AUDIO_CODING_TEST_ISACTEST_H_
#include <string.h>
#include <memory>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
+#include "common_types.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/utility.h"
#define MAX_FILE_NAME_LENGTH_BYTE 500
#define NO_OF_CLIENTS 15
@@ -77,4 +77,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_ISACTEST_H_
+#endif // MODULES_AUDIO_CODING_TEST_ISACTEST_H_
diff --git a/modules/audio_coding/test/insert_packet_with_timing.cc b/modules/audio_coding/test/insert_packet_with_timing.cc
index db58289..5f1da58 100644
--- a/modules/audio_coding/test/insert_packet_with_timing.cc
+++ b/modules/audio_coding/test/insert_packet_with_timing.cc
@@ -13,16 +13,16 @@
#include <memory>
-#include "webrtc/common_types.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/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/rtc_base/flags.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/include/module_common_types.h"
+#include "rtc_base/flags.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
// Codec.
DEFINE_string(codec, "opus", "Codec Name");
diff --git a/modules/audio_coding/test/opus_test.cc b/modules/audio_coding/test/opus_test.cc
index 9f5720b..5ba810c 100644
--- a/modules/audio_coding/test/opus_test.cc
+++ b/modules/audio_coding/test/opus_test.cc
@@ -8,21 +8,21 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/modules/audio_coding/test/opus_test.h"
+#include "modules/audio_coding/test/opus_test.h"
#include <assert.h>
#include <string>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
-#include "webrtc/modules/audio_coding/test/TestStereo.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/typedefs.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/audio_format_conversion.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/audio_coding/test/TestStereo.h"
+#include "modules/audio_coding/test/utility.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+#include "typedefs.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/opus_test.h b/modules/audio_coding/test/opus_test.h
index ce570f6..3e9d9a7 100644
--- a/modules/audio_coding/test/opus_test.h
+++ b/modules/audio_coding/test/opus_test.h
@@ -8,19 +8,19 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#ifndef MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#define MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
#include <math.h>
#include <memory>
-#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
-#include "webrtc/modules/audio_coding/acm2/acm_resampler.h"
-#include "webrtc/modules/audio_coding/test/ACMTest.h"
-#include "webrtc/modules/audio_coding/test/Channel.h"
-#include "webrtc/modules/audio_coding/test/PCMFile.h"
-#include "webrtc/modules/audio_coding/test/TestStereo.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/acm2/acm_resampler.h"
+#include "modules/audio_coding/test/ACMTest.h"
+#include "modules/audio_coding/test/Channel.h"
+#include "modules/audio_coding/test/PCMFile.h"
+#include "modules/audio_coding/test/TestStereo.h"
namespace webrtc {
@@ -58,4 +58,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
+#endif // MODULES_AUDIO_CODING_TEST_OPUS_TEST_H_
diff --git a/modules/audio_coding/test/target_delay_unittest.cc b/modules/audio_coding/test/target_delay_unittest.cc
index 7aec5d0..2e475dd 100644
--- a/modules/audio_coding/test/target_delay_unittest.cc
+++ b/modules/audio_coding/test/target_delay_unittest.cc
@@ -10,13 +10,13 @@
#include <memory>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_coding/test/utility.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
+#include "common_types.h"
+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_coding/test/utility.h"
+#include "modules/include/module_common_types.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
namespace webrtc {
diff --git a/modules/audio_coding/test/utility.cc b/modules/audio_coding/test/utility.cc
index 043e6b9..ecade69 100644
--- a/modules/audio_coding/test/utility.cc
+++ b/modules/audio_coding/test/utility.cc
@@ -15,9 +15,9 @@
#include <stdlib.h>
#include <string.h>
-#include "webrtc/common_types.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/test/gtest.h"
+#include "common_types.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "test/gtest.h"
#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
diff --git a/modules/audio_coding/test/utility.h b/modules/audio_coding/test/utility.h
index 8af3675..07cbe71 100644
--- a/modules/audio_coding/test/utility.h
+++ b/modules/audio_coding/test/utility.h
@@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#ifndef MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#define MODULES_AUDIO_CODING_TEST_UTILITY_H_
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/test/gtest.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "test/gtest.h"
namespace webrtc {
@@ -135,4 +135,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_AUDIO_CODING_TEST_UTILITY_H_
+#endif // MODULES_AUDIO_CODING_TEST_UTILITY_H_