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/call/DEPS b/call/DEPS
index 54dcebb..7622e24 100644
--- a/call/DEPS
+++ b/call/DEPS
@@ -1,27 +1,27 @@
include_rules = [
- "+webrtc/audio",
- "+webrtc/logging/rtc_event_log",
- "+webrtc/modules/audio_coding",
- "+webrtc/modules/audio_device",
- "+webrtc/modules/audio_mixer",
- "+webrtc/modules/audio_processing",
- "+webrtc/modules/bitrate_controller",
- "+webrtc/modules/congestion_controller",
- "+webrtc/modules/pacing",
- "+webrtc/modules/rtp_rtcp",
- "+webrtc/modules/utility",
- "+webrtc/system_wrappers",
- "+webrtc/voice_engine",
- "+webrtc/video",
+ "+audio",
+ "+logging/rtc_event_log",
+ "+modules/audio_coding",
+ "+modules/audio_device",
+ "+modules/audio_mixer",
+ "+modules/audio_processing",
+ "+modules/bitrate_controller",
+ "+modules/congestion_controller",
+ "+modules/pacing",
+ "+modules/rtp_rtcp",
+ "+modules/utility",
+ "+system_wrappers",
+ "+voice_engine",
+ "+video",
]
specific_include_rules = {
"video_receive_stream\.h": [
- "+webrtc/common_video/include",
- "+webrtc/media/base",
+ "+common_video/include",
+ "+media/base",
],
"video_send_stream\.h": [
- "+webrtc/common_video/include",
- "+webrtc/media/base",
+ "+common_video/include",
+ "+media/base",
],
}
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 6de62df..1dd59ae 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -8,23 +8,23 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
-#define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
+#ifndef CALL_AUDIO_RECEIVE_STREAM_H_
+#define CALL_AUDIO_RECEIVE_STREAM_H_
#include <map>
#include <memory>
#include <string>
#include <vector>
-#include "webrtc/api/audio_codecs/audio_decoder_factory.h"
-#include "webrtc/api/call/transport.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/api/rtpparameters.h"
-#include "webrtc/api/rtpreceiverinterface.h"
-#include "webrtc/call/rtp_config.h"
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/call/transport.h"
+#include "api/optional.h"
+#include "api/rtpparameters.h"
+#include "api/rtpreceiverinterface.h"
+#include "call/rtp_config.h"
+#include "common_types.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"
namespace webrtc {
class AudioSinkInterface;
@@ -155,4 +155,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
+#endif // CALL_AUDIO_RECEIVE_STREAM_H_
diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc
index 0bc555b..d0043a2 100644
--- a/call/audio_send_stream.cc
+++ b/call/audio_send_stream.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/audio_send_stream.h"
+#include "call/audio_send_stream.h"
#include <string>
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index ac962c5..8160318 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -8,22 +8,22 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_AUDIO_SEND_STREAM_H_
-#define WEBRTC_CALL_AUDIO_SEND_STREAM_H_
+#ifndef CALL_AUDIO_SEND_STREAM_H_
+#define CALL_AUDIO_SEND_STREAM_H_
#include <memory>
#include <string>
#include <vector>
-#include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/api/audio_codecs/audio_encoder_factory.h"
-#include "webrtc/api/audio_codecs/audio_format.h"
-#include "webrtc/api/call/transport.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/api/rtpparameters.h"
-#include "webrtc/call/rtp_config.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
+#include "api/audio_codecs/audio_encoder.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/audio_codecs/audio_format.h"
+#include "api/call/transport.h"
+#include "api/optional.h"
+#include "api/rtpparameters.h"
+#include "call/rtp_config.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"
namespace webrtc {
@@ -157,4 +157,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_CALL_AUDIO_SEND_STREAM_H_
+#endif // CALL_AUDIO_SEND_STREAM_H_
diff --git a/call/audio_state.h b/call/audio_state.h
index 826b31d..7719388 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -7,12 +7,12 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_AUDIO_STATE_H_
-#define WEBRTC_CALL_AUDIO_STATE_H_
+#ifndef CALL_AUDIO_STATE_H_
+#define CALL_AUDIO_STATE_H_
-#include "webrtc/api/audio/audio_mixer.h"
-#include "webrtc/rtc_base/refcount.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
+#include "api/audio/audio_mixer.h"
+#include "rtc_base/refcount.h"
+#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
@@ -52,4 +52,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_CALL_AUDIO_STATE_H_
+#endif // CALL_AUDIO_STATE_H_
diff --git a/call/bitrate_allocator.cc b/call/bitrate_allocator.cc
index 1e22c00..8570ae3 100644
--- a/call/bitrate_allocator.cc
+++ b/call/bitrate_allocator.cc
@@ -9,16 +9,16 @@
*
*/
-#include "webrtc/call/bitrate_allocator.h"
+#include "call/bitrate_allocator.h"
#include <algorithm>
#include <utility>
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/metrics.h"
+#include "modules/bitrate_controller/include/bitrate_controller.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/metrics.h"
namespace webrtc {
diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h
index b9bbc93..1ac3487 100644
--- a/call/bitrate_allocator.h
+++ b/call/bitrate_allocator.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_BITRATE_ALLOCATOR_H_
-#define WEBRTC_CALL_BITRATE_ALLOCATOR_H_
+#ifndef CALL_BITRATE_ALLOCATOR_H_
+#define CALL_BITRATE_ALLOCATOR_H_
#include <stdint.h>
@@ -17,7 +17,7 @@
#include <utility>
#include <vector>
-#include "webrtc/rtc_base/sequenced_task_checker.h"
+#include "rtc_base/sequenced_task_checker.h"
namespace webrtc {
@@ -169,4 +169,4 @@
uint32_t total_requested_min_bitrate_ RTC_GUARDED_BY(&sequenced_checker_);
};
} // namespace webrtc
-#endif // WEBRTC_CALL_BITRATE_ALLOCATOR_H_
+#endif // CALL_BITRATE_ALLOCATOR_H_
diff --git a/call/bitrate_allocator_unittest.cc b/call/bitrate_allocator_unittest.cc
index ead4cbd..03ad78c 100644
--- a/call/bitrate_allocator_unittest.cc
+++ b/call/bitrate_allocator_unittest.cc
@@ -12,10 +12,10 @@
#include <memory>
#include <vector>
-#include "webrtc/call/bitrate_allocator.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "call/bitrate_allocator.h"
+#include "modules/bitrate_controller/include/bitrate_controller.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
using testing::NiceMock;
diff --git a/call/bitrate_estimator_tests.cc b/call/bitrate_estimator_tests.cc
index 49f149b..4c39787 100644
--- a/call/bitrate_estimator_tests.cc
+++ b/call/bitrate_estimator_tests.cc
@@ -12,18 +12,18 @@
#include <memory>
#include <string>
-#include "webrtc/call/call.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/event.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/test/call_test.h"
-#include "webrtc/test/direct_transport.h"
-#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_decoder.h"
-#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/frame_generator_capturer.h"
-#include "webrtc/test/gtest.h"
+#include "call/call.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/event.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/call_test.h"
+#include "test/direct_transport.h"
+#include "test/encoder_settings.h"
+#include "test/fake_decoder.h"
+#include "test/fake_encoder.h"
+#include "test/frame_generator_capturer.h"
+#include "test/gtest.h"
namespace webrtc {
namespace {
diff --git a/call/call.cc b/call/call.cc
index b7cd059..102df03 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -16,46 +16,46 @@
#include <utility>
#include <vector>
-#include "webrtc/api/optional.h"
-#include "webrtc/audio/audio_receive_stream.h"
-#include "webrtc/audio/audio_send_stream.h"
-#include "webrtc/audio/audio_state.h"
-#include "webrtc/audio/scoped_voe_interface.h"
-#include "webrtc/audio/time_interval.h"
-#include "webrtc/call/bitrate_allocator.h"
-#include "webrtc/call/call.h"
-#include "webrtc/call/flexfec_receive_stream_impl.h"
-#include "webrtc/call/rtp_stream_receiver_controller.h"
-#include "webrtc/call/rtp_transport_controller_send.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/constructormagic.h"
-#include "webrtc/rtc_base/location.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/rtc_base/sequenced_task_checker.h"
-#include "webrtc/rtc_base/task_queue.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/rtc_base/trace_event.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/cpu_info.h"
-#include "webrtc/system_wrappers/include/metrics.h"
-#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
-#include "webrtc/system_wrappers/include/trace.h"
-#include "webrtc/video/call_stats.h"
-#include "webrtc/video/send_delay_stats.h"
-#include "webrtc/video/stats_counter.h"
-#include "webrtc/video/video_receive_stream.h"
-#include "webrtc/video/video_send_stream.h"
+#include "api/optional.h"
+#include "audio/audio_receive_stream.h"
+#include "audio/audio_send_stream.h"
+#include "audio/audio_state.h"
+#include "audio/scoped_voe_interface.h"
+#include "audio/time_interval.h"
+#include "call/bitrate_allocator.h"
+#include "call/call.h"
+#include "call/flexfec_receive_stream_impl.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "call/rtp_transport_controller_send.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/bitrate_controller/include/bitrate_controller.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/utility/include/process_thread.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/constructormagic.h"
+#include "rtc_base/location.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/ptr_util.h"
+#include "rtc_base/sequenced_task_checker.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/cpu_info.h"
+#include "system_wrappers/include/metrics.h"
+#include "system_wrappers/include/rw_lock_wrapper.h"
+#include "system_wrappers/include/trace.h"
+#include "video/call_stats.h"
+#include "video/send_delay_stats.h"
+#include "video/stats_counter.h"
+#include "video/video_receive_stream.h"
+#include "video/video_send_stream.h"
namespace webrtc {
diff --git a/call/call.h b/call/call.h
index 74ea2eb..9901745 100644
--- a/call/call.h
+++ b/call/call.h
@@ -7,26 +7,26 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_CALL_H_
-#define WEBRTC_CALL_CALL_H_
+#ifndef CALL_CALL_H_
+#define CALL_CALL_H_
#include <algorithm>
#include <memory>
#include <string>
#include <vector>
-#include "webrtc/api/rtcerror.h"
-#include "webrtc/call/audio_receive_stream.h"
-#include "webrtc/call/audio_send_stream.h"
-#include "webrtc/call/audio_state.h"
-#include "webrtc/call/flexfec_receive_stream.h"
-#include "webrtc/call/rtp_transport_controller_send_interface.h"
-#include "webrtc/call/video_receive_stream.h"
-#include "webrtc/call/video_send_stream.h"
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/networkroute.h"
-#include "webrtc/rtc_base/platform_file.h"
-#include "webrtc/rtc_base/socket.h"
+#include "api/rtcerror.h"
+#include "call/audio_receive_stream.h"
+#include "call/audio_send_stream.h"
+#include "call/audio_state.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "call/video_receive_stream.h"
+#include "call/video_send_stream.h"
+#include "common_types.h"
+#include "rtc_base/networkroute.h"
+#include "rtc_base/platform_file.h"
+#include "rtc_base/socket.h"
namespace webrtc {
@@ -204,4 +204,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_CALL_H_
+#endif // CALL_CALL_H_
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 46611a1..58435f0 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -13,33 +13,33 @@
#include <memory>
#include <string>
-#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
-#include "webrtc/call/call.h"
-#include "webrtc/call/video_config.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
-#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/rtc_base/thread_annotations.h"
-#include "webrtc/system_wrappers/include/metrics_default.h"
-#include "webrtc/test/call_test.h"
-#include "webrtc/test/direct_transport.h"
-#include "webrtc/test/drifting_clock.h"
-#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/fake_audio_device.h"
-#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/field_trial.h"
-#include "webrtc/test/frame_generator.h"
-#include "webrtc/test/frame_generator_capturer.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/rtp_rtcp_observer.h"
-#include "webrtc/test/single_threaded_task_queue.h"
-#include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/perf_test.h"
-#include "webrtc/video/transport_adapter.h"
-#include "webrtc/voice_engine/include/voe_base.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "call/call.h"
+#include "call/video_config.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ptr_util.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/metrics_default.h"
+#include "test/call_test.h"
+#include "test/direct_transport.h"
+#include "test/drifting_clock.h"
+#include "test/encoder_settings.h"
+#include "test/fake_audio_device.h"
+#include "test/fake_encoder.h"
+#include "test/field_trial.h"
+#include "test/frame_generator.h"
+#include "test/frame_generator_capturer.h"
+#include "test/gtest.h"
+#include "test/rtp_rtcp_observer.h"
+#include "test/single_threaded_task_queue.h"
+#include "test/testsupport/fileutils.h"
+#include "test/testsupport/perf_test.h"
+#include "video/transport_adapter.h"
+#include "voice_engine/include/voe_base.h"
using webrtc::test::DriftingClock;
using webrtc::test::FakeAudioDevice;
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index e342dfa..b1ebb82 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -13,22 +13,22 @@
#include <memory>
#include <utility>
-#include "webrtc/api/test/mock_audio_mixer.h"
-#include "webrtc/call/audio_state.h"
-#include "webrtc/call/call.h"
-#include "webrtc/call/fake_rtp_transport_controller_send.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/modules/audio_device/include/mock_audio_device.h"
-#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
-#include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
-#include "webrtc/modules/pacing/mock/mock_paced_sender.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/test/fake_encoder.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_audio_decoder_factory.h"
-#include "webrtc/test/mock_transport.h"
-#include "webrtc/test/mock_voice_engine.h"
+#include "api/test/mock_audio_mixer.h"
+#include "call/audio_state.h"
+#include "call/call.h"
+#include "call/fake_rtp_transport_controller_send.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "modules/audio_device/include/mock_audio_device.h"
+#include "modules/audio_mixer/audio_mixer_impl.h"
+#include "modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
+#include "modules/pacing/mock/mock_paced_sender.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "rtc_base/ptr_util.h"
+#include "test/fake_encoder.h"
+#include "test/gtest.h"
+#include "test/mock_audio_decoder_factory.h"
+#include "test/mock_transport.h"
+#include "test/mock_voice_engine.h"
namespace {
diff --git a/call/callfactory.cc b/call/callfactory.cc
index 7399cfc..82acb65 100644
--- a/call/callfactory.cc
+++ b/call/callfactory.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/callfactory.h"
+#include "call/callfactory.h"
#include <memory>
diff --git a/call/callfactory.h b/call/callfactory.h
index 98eb68a..167b82a 100644
--- a/call/callfactory.h
+++ b/call/callfactory.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_CALLFACTORY_H_
-#define WEBRTC_CALL_CALLFACTORY_H_
+#ifndef CALL_CALLFACTORY_H_
+#define CALL_CALLFACTORY_H_
-#include "webrtc/call/callfactoryinterface.h"
+#include "call/callfactoryinterface.h"
namespace webrtc {
@@ -23,4 +23,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_CALLFACTORY_H_
+#endif // CALL_CALLFACTORY_H_
diff --git a/call/callfactoryinterface.h b/call/callfactoryinterface.h
index 6a0329a..a3cf6eb 100644
--- a/call/callfactoryinterface.h
+++ b/call/callfactoryinterface.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_CALLFACTORYINTERFACE_H_
-#define WEBRTC_CALL_CALLFACTORYINTERFACE_H_
+#ifndef CALL_CALLFACTORYINTERFACE_H_
+#define CALL_CALLFACTORYINTERFACE_H_
#include <memory>
-#include "webrtc/call/call.h"
+#include "call/call.h"
namespace webrtc {
@@ -31,4 +31,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_CALLFACTORYINTERFACE_H_
+#endif // CALL_CALLFACTORYINTERFACE_H_
diff --git a/call/fake_rtp_transport_controller_send.h b/call/fake_rtp_transport_controller_send.h
index 6cff173..b6174c6 100644
--- a/call/fake_rtp_transport_controller_send.h
+++ b/call/fake_rtp_transport_controller_send.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
-#define WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#ifndef CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#define CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
-#include "webrtc/call/rtp_transport_controller_send_interface.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
-#include "webrtc/modules/pacing/packet_router.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "common_types.h"
+#include "modules/congestion_controller/include/send_side_congestion_controller.h"
+#include "modules/pacing/packet_router.h"
namespace webrtc {
@@ -65,4 +65,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#endif // CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index 400bca4..ee83c26 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
-#define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
+#ifndef CALL_FLEXFEC_RECEIVE_STREAM_H_
+#define CALL_FLEXFEC_RECEIVE_STREAM_H_
#include <stdint.h>
#include <string>
#include <vector>
-#include "webrtc/api/call/transport.h"
-#include "webrtc/api/rtpparameters.h"
-#include "webrtc/call/rtp_packet_sink_interface.h"
-#include "webrtc/common_types.h"
+#include "api/call/transport.h"
+#include "api/rtpparameters.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "common_types.h"
namespace webrtc {
@@ -84,4 +84,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_H_
+#endif // CALL_FLEXFEC_RECEIVE_STREAM_H_
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index d6b41ed..bb68f6e 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -8,20 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/flexfec_receive_stream_impl.h"
+#include "call/flexfec_receive_stream_impl.h"
#include <string>
-#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
-#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/modules/utility/include/process_thread.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/location.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/system_wrappers/include/clock.h"
+#include "call/rtp_stream_receiver_controller_interface.h"
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/utility/include/process_thread.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/location.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
namespace webrtc {
diff --git a/call/flexfec_receive_stream_impl.h b/call/flexfec_receive_stream_impl.h
index f60e810..6bcbc7c 100644
--- a/call/flexfec_receive_stream_impl.h
+++ b/call/flexfec_receive_stream_impl.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_
-#define WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_
+#ifndef CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_
+#define CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_
#include <memory>
-#include "webrtc/call/flexfec_receive_stream.h"
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "call/flexfec_receive_stream.h"
+#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
@@ -61,4 +61,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_
+#endif // CALL_FLEXFEC_RECEIVE_STREAM_IMPL_H_
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc
index c3bf2e0..21dbeb7 100644
--- a/call/flexfec_receive_stream_unittest.cc
+++ b/call/flexfec_receive_stream_unittest.cc
@@ -8,24 +8,24 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/flexfec_receive_stream_impl.h"
+#include "call/flexfec_receive_stream_impl.h"
#include <stdint.h>
#include <memory>
-#include "webrtc/api/array_view.h"
-#include "webrtc/call/rtp_stream_receiver_controller.h"
-#include "webrtc/modules/pacing/packet_router.h"
-#include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
-#include "webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
-#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "webrtc/modules/utility/include/mock/mock_process_thread.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/mock_transport.h"
+#include "api/array_view.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
+#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/utility/include/mock/mock_process_thread.h"
+#include "rtc_base/ptr_util.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_transport.h"
namespace webrtc {
diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc
index 2b85452..da7095c 100644
--- a/call/rampup_tests.cc
+++ b/call/rampup_tests.cc
@@ -8,14 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rampup_tests.h"
+#include "call/rampup_tests.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/platform_thread.h"
-#include "webrtc/test/encoder_settings.h"
-#include "webrtc/test/gtest.h"
-#include "webrtc/test/testsupport/perf_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/platform_thread.h"
+#include "test/encoder_settings.h"
+#include "test/gtest.h"
+#include "test/testsupport/perf_test.h"
namespace webrtc {
namespace {
diff --git a/call/rampup_tests.h b/call/rampup_tests.h
index 8358822..1339f1f 100644
--- a/call/rampup_tests.h
+++ b/call/rampup_tests.h
@@ -8,17 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RAMPUP_TESTS_H_
-#define WEBRTC_CALL_RAMPUP_TESTS_H_
+#ifndef CALL_RAMPUP_TESTS_H_
+#define CALL_RAMPUP_TESTS_H_
#include <map>
#include <string>
#include <vector>
-#include "webrtc/call/call.h"
-#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
-#include "webrtc/rtc_base/event.h"
-#include "webrtc/test/call_test.h"
+#include "call/call.h"
+#include "logging/rtc_event_log/rtc_event_log.h"
+#include "rtc_base/event.h"
+#include "test/call_test.h"
namespace webrtc {
@@ -157,4 +157,4 @@
std::vector<int> loss_rates_;
};
} // namespace webrtc
-#endif // WEBRTC_CALL_RAMPUP_TESTS_H_
+#endif // CALL_RAMPUP_TESTS_H_
diff --git a/call/rtcp_demuxer.cc b/call/rtcp_demuxer.cc
index 90620c32..6533599 100644
--- a/call/rtcp_demuxer.cc
+++ b/call/rtcp_demuxer.cc
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtcp_demuxer.h"
+#include "call/rtcp_demuxer.h"
-#include "webrtc/call/rtcp_packet_sink_interface.h"
-#include "webrtc/call/rtp_rtcp_demuxer_helper.h"
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/checks.h"
+#include "call/rtcp_packet_sink_interface.h"
+#include "call/rtp_rtcp_demuxer_helper.h"
+#include "common_types.h"
+#include "rtc_base/checks.h"
namespace webrtc {
diff --git a/call/rtcp_demuxer.h b/call/rtcp_demuxer.h
index 7b5af9f..87b5816 100644
--- a/call/rtcp_demuxer.h
+++ b/call/rtcp_demuxer.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTCP_DEMUXER_H_
-#define WEBRTC_CALL_RTCP_DEMUXER_H_
+#ifndef CALL_RTCP_DEMUXER_H_
+#define CALL_RTCP_DEMUXER_H_
#include <map>
#include <string>
#include <vector>
-#include "webrtc/api/array_view.h"
-#include "webrtc/call/ssrc_binding_observer.h"
-#include "webrtc/rtc_base/basictypes.h"
+#include "api/array_view.h"
+#include "call/ssrc_binding_observer.h"
+#include "rtc_base/basictypes.h"
namespace webrtc {
@@ -82,4 +82,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTCP_DEMUXER_H_
+#endif // CALL_RTCP_DEMUXER_H_
diff --git a/call/rtcp_demuxer_unittest.cc b/call/rtcp_demuxer_unittest.cc
index 91adf99..9125f0e 100644
--- a/call/rtcp_demuxer_unittest.cc
+++ b/call/rtcp_demuxer_unittest.cc
@@ -8,20 +8,20 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtcp_demuxer.h"
+#include "call/rtcp_demuxer.h"
#include <memory>
#include <set>
-#include "webrtc/call/rtcp_packet_sink_interface.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "call/rtcp_packet_sink_interface.h"
+#include "common_types.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ptr_util.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/call/rtcp_packet_sink_interface.h b/call/rtcp_packet_sink_interface.h
index 22a404d..8ea3f7d 100644
--- a/call/rtcp_packet_sink_interface.h
+++ b/call/rtcp_packet_sink_interface.h
@@ -7,10 +7,10 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
-#define WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
+#ifndef CALL_RTCP_PACKET_SINK_INTERFACE_H_
+#define CALL_RTCP_PACKET_SINK_INTERFACE_H_
-#include "webrtc/api/array_view.h"
+#include "api/array_view.h"
namespace webrtc {
@@ -26,4 +26,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTCP_PACKET_SINK_INTERFACE_H_
+#endif // CALL_RTCP_PACKET_SINK_INTERFACE_H_
diff --git a/call/rtp_config.cc b/call/rtp_config.cc
index dc3ea21..3621f72 100644
--- a/call/rtp_config.cc
+++ b/call/rtp_config.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtp_config.h"
+#include "call/rtp_config.h"
#include <sstream>
diff --git a/call/rtp_config.h b/call/rtp_config.h
index f33103d..86d32ac 100644
--- a/call/rtp_config.h
+++ b/call/rtp_config.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_CONFIG_H_
-#define WEBRTC_CALL_RTP_CONFIG_H_
+#ifndef CALL_RTP_CONFIG_H_
+#define CALL_RTP_CONFIG_H_
#include <string>
@@ -45,4 +45,4 @@
int red_rtx_payload_type;
};
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_CONFIG_H_
+#endif // CALL_RTP_CONFIG_H_
diff --git a/call/rtp_demuxer.cc b/call/rtp_demuxer.cc
index 1939a0f..0f9173e 100644
--- a/call/rtp_demuxer.cc
+++ b/call/rtp_demuxer.cc
@@ -8,15 +8,15 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtp_demuxer.h"
+#include "call/rtp_demuxer.h"
-#include "webrtc/call/rtp_packet_sink_interface.h"
-#include "webrtc/call/rtp_rtcp_demuxer_helper.h"
-#include "webrtc/call/ssrc_binding_observer.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/logging.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "call/rtp_rtcp_demuxer_helper.h"
+#include "call/ssrc_binding_observer.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
namespace webrtc {
diff --git a/call/rtp_demuxer.h b/call/rtp_demuxer.h
index ddd4a2f..971c151 100644
--- a/call/rtp_demuxer.h
+++ b/call/rtp_demuxer.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_DEMUXER_H_
-#define WEBRTC_CALL_RTP_DEMUXER_H_
+#ifndef CALL_RTP_DEMUXER_H_
+#define CALL_RTP_DEMUXER_H_
#include <map>
#include <set>
@@ -201,4 +201,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_DEMUXER_H_
+#endif // CALL_RTP_DEMUXER_H_
diff --git a/call/rtp_demuxer_unittest.cc b/call/rtp_demuxer_unittest.cc
index 6fe5fc8..6202413 100644
--- a/call/rtp_demuxer_unittest.cc
+++ b/call/rtp_demuxer_unittest.cc
@@ -8,25 +8,25 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtp_demuxer.h"
+#include "call/rtp_demuxer.h"
#include <memory>
#include <set>
#include <string>
-#include "webrtc/call/ssrc_binding_observer.h"
-#include "webrtc/call/test/mock_rtp_packet_sink_interface.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/checks.h"
-#include "webrtc/rtc_base/ptr_util.h"
-#include "webrtc/rtc_base/safe_conversions.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "call/ssrc_binding_observer.h"
+#include "call/test/mock_rtp_packet_sink_interface.h"
+#include "common_types.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/ptr_util.h"
+#include "rtc_base/safe_conversions.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/call/rtp_packet_sink_interface.h b/call/rtp_packet_sink_interface.h
index 0b3e64e..ffbd58c 100644
--- a/call/rtp_packet_sink_interface.h
+++ b/call/rtp_packet_sink_interface.h
@@ -7,8 +7,8 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_H_
-#define WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_H_
+#ifndef CALL_RTP_PACKET_SINK_INTERFACE_H_
+#define CALL_RTP_PACKET_SINK_INTERFACE_H_
namespace webrtc {
@@ -23,4 +23,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_PACKET_SINK_INTERFACE_H_
+#endif // CALL_RTP_PACKET_SINK_INTERFACE_H_
diff --git a/call/rtp_rtcp_demuxer_helper.cc b/call/rtp_rtcp_demuxer_helper.cc
index e8d3cbf..d36242b 100644
--- a/call/rtp_rtcp_demuxer_helper.cc
+++ b/call/rtp_rtcp_demuxer_helper.cc
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtp_rtcp_demuxer_helper.h"
+#include "call/rtp_rtcp_demuxer_helper.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
namespace webrtc {
diff --git a/call/rtp_rtcp_demuxer_helper.h b/call/rtp_rtcp_demuxer_helper.h
index 0a4767d..32408e8 100644
--- a/call/rtp_rtcp_demuxer_helper.h
+++ b/call/rtp_rtcp_demuxer_helper.h
@@ -8,16 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_RTCP_DEMUXER_HELPER_H_
-#define WEBRTC_CALL_RTP_RTCP_DEMUXER_HELPER_H_
+#ifndef CALL_RTP_RTCP_DEMUXER_HELPER_H_
+#define CALL_RTP_RTCP_DEMUXER_HELPER_H_
#include <algorithm>
#include <map>
#include <utility>
-#include "webrtc/api/array_view.h"
-#include "webrtc/api/optional.h"
-#include "webrtc/rtc_base/basictypes.h"
+#include "api/array_view.h"
+#include "api/optional.h"
+#include "rtc_base/basictypes.h"
namespace webrtc {
@@ -95,4 +95,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_RTCP_DEMUXER_HELPER_H_
+#endif // CALL_RTP_RTCP_DEMUXER_HELPER_H_
diff --git a/call/rtp_rtcp_demuxer_helper_unittest.cc b/call/rtp_rtcp_demuxer_helper_unittest.cc
index 96a1383..cb8a092 100644
--- a/call/rtp_rtcp_demuxer_helper_unittest.cc
+++ b/call/rtp_rtcp_demuxer_helper_unittest.cc
@@ -10,19 +10,19 @@
#include <cstdio>
-#include "webrtc/call/rtp_rtcp_demuxer_helper.h"
+#include "call/rtp_rtcp_demuxer_helper.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
-#include "webrtc/rtc_base/arraysize.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/buffer.h"
-#include "webrtc/test/gtest.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/pli.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/buffer.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/call/rtp_stream_receiver_controller.cc b/call/rtp_stream_receiver_controller.cc
index 94fa83b..44a1622 100644
--- a/call/rtp_stream_receiver_controller.cc
+++ b/call/rtp_stream_receiver_controller.cc
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtp_stream_receiver_controller.h"
+#include "call/rtp_stream_receiver_controller.h"
-#include "webrtc/rtc_base/logging.h"
-#include "webrtc/rtc_base/ptr_util.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/ptr_util.h"
namespace webrtc {
diff --git a/call/rtp_stream_receiver_controller.h b/call/rtp_stream_receiver_controller.h
index feec5e4..c523e3f 100644
--- a/call/rtp_stream_receiver_controller.h
+++ b/call/rtp_stream_receiver_controller.h
@@ -7,14 +7,14 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
-#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
+#ifndef CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
+#define CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
#include <memory>
-#include "webrtc/call/rtp_demuxer.h"
-#include "webrtc/call/rtp_stream_receiver_controller_interface.h"
-#include "webrtc/rtc_base/criticalsection.h"
+#include "call/rtp_demuxer.h"
+#include "call/rtp_stream_receiver_controller_interface.h"
+#include "rtc_base/criticalsection.h"
namespace webrtc {
@@ -69,4 +69,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
+#endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_H_
diff --git a/call/rtp_stream_receiver_controller_interface.h b/call/rtp_stream_receiver_controller_interface.h
index 94b4f10..a5e5295 100644
--- a/call/rtp_stream_receiver_controller_interface.h
+++ b/call/rtp_stream_receiver_controller_interface.h
@@ -7,12 +7,12 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
-#define WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
+#ifndef CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
+#define CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
#include <memory>
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
@@ -44,4 +44,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
+#endif // CALL_RTP_STREAM_RECEIVER_CONTROLLER_INTERFACE_H_
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
index ec061d0..090c261 100644
--- a/call/rtp_transport_controller_send.cc
+++ b/call/rtp_transport_controller_send.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtp_transport_controller_send.h"
+#include "call/rtp_transport_controller_send.h"
namespace webrtc {
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 766a2e3..c50ea11 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
-#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
-#include "webrtc/call/rtp_transport_controller_send_interface.h"
-#include "webrtc/common_types.h"
-#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
-#include "webrtc/rtc_base/constructormagic.h"
+#include "call/rtp_transport_controller_send_interface.h"
+#include "common_types.h"
+#include "modules/congestion_controller/include/send_side_congestion_controller.h"
+#include "rtc_base/constructormagic.h"
namespace webrtc {
class Clock;
@@ -55,4 +55,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
+#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index b580421..7b0dbef 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
-#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
+#ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
+#define CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
namespace webrtc {
@@ -70,4 +70,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
+#endif // CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
diff --git a/call/rtx_receive_stream.cc b/call/rtx_receive_stream.cc
index 6a5432f..3013b62 100644
--- a/call/rtx_receive_stream.cc
+++ b/call/rtx_receive_stream.cc
@@ -10,10 +10,10 @@
#include <utility>
-#include "webrtc/call/rtx_receive_stream.h"
-#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/rtc_base/logging.h"
+#include "call/rtx_receive_stream.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "rtc_base/logging.h"
namespace webrtc {
diff --git a/call/rtx_receive_stream.h b/call/rtx_receive_stream.h
index c288a27..8ffa440 100644
--- a/call/rtx_receive_stream.h
+++ b/call/rtx_receive_stream.h
@@ -8,12 +8,12 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
-#define WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#ifndef CALL_RTX_RECEIVE_STREAM_H_
+#define CALL_RTX_RECEIVE_STREAM_H_
#include <map>
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
@@ -47,4 +47,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_RTX_RECEIVE_STREAM_H_
+#endif // CALL_RTX_RECEIVE_STREAM_H_
diff --git a/call/rtx_receive_stream_unittest.cc b/call/rtx_receive_stream_unittest.cc
index 2b2625d..65ab82b 100644
--- a/call/rtx_receive_stream_unittest.cc
+++ b/call/rtx_receive_stream_unittest.cc
@@ -8,13 +8,13 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/rtx_receive_stream.h"
-#include "webrtc/call/test/mock_rtp_packet_sink_interface.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
-#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
-#include "webrtc/test/gmock.h"
-#include "webrtc/test/gtest.h"
+#include "call/rtx_receive_stream.h"
+#include "call/test/mock_rtp_packet_sink_interface.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
namespace webrtc {
diff --git a/call/ssrc_binding_observer.h b/call/ssrc_binding_observer.h
index 205a5c9..73a8e8e 100644
--- a/call/ssrc_binding_observer.h
+++ b/call/ssrc_binding_observer.h
@@ -7,12 +7,12 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_SSRC_BINDING_OBSERVER_H_
-#define WEBRTC_CALL_SSRC_BINDING_OBSERVER_H_
+#ifndef CALL_SSRC_BINDING_OBSERVER_H_
+#define CALL_SSRC_BINDING_OBSERVER_H_
#include <string>
-#include "webrtc/rtc_base/basictypes.h"
+#include "rtc_base/basictypes.h"
namespace webrtc {
@@ -39,4 +39,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_SSRC_BINDING_OBSERVER_H_
+#endif // CALL_SSRC_BINDING_OBSERVER_H_
diff --git a/call/syncable.cc b/call/syncable.cc
index c254e4f..a821881 100644
--- a/call/syncable.cc
+++ b/call/syncable.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/syncable.h"
+#include "call/syncable.h"
namespace webrtc {
diff --git a/call/syncable.h b/call/syncable.h
index 2692a8a..a97990b 100644
--- a/call/syncable.h
+++ b/call/syncable.h
@@ -11,12 +11,12 @@
// Syncable is used by RtpStreamsSynchronizer in VideoReceiveStream, and
// implemented by AudioReceiveStream.
-#ifndef WEBRTC_CALL_SYNCABLE_H_
-#define WEBRTC_CALL_SYNCABLE_H_
+#ifndef CALL_SYNCABLE_H_
+#define CALL_SYNCABLE_H_
#include <stdint.h>
-#include "webrtc/api/optional.h"
+#include "api/optional.h"
namespace webrtc {
@@ -40,4 +40,4 @@
};
} // namespace webrtc
-#endif // WEBRTC_CALL_SYNCABLE_H_
+#endif // CALL_SYNCABLE_H_
diff --git a/call/test/mock_rtp_packet_sink_interface.h b/call/test/mock_rtp_packet_sink_interface.h
index a139004..add399b 100644
--- a/call/test/mock_rtp_packet_sink_interface.h
+++ b/call/test/mock_rtp_packet_sink_interface.h
@@ -7,12 +7,12 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
-#define WEBRTC_CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
+#ifndef CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
+#define CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
-#include "webrtc/call/rtp_packet_sink_interface.h"
+#include "call/rtp_packet_sink_interface.h"
-#include "webrtc/test/gmock.h"
+#include "test/gmock.h"
namespace webrtc {
@@ -23,4 +23,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
+#endif // CALL_TEST_MOCK_RTP_PACKET_SINK_INTERFACE_H_
diff --git a/call/video_config.cc b/call/video_config.cc
index e06d5e8..7cf2640 100644
--- a/call/video_config.cc
+++ b/call/video_config.cc
@@ -7,13 +7,13 @@
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/video_config.h"
+#include "call/video_config.h"
#include <algorithm>
#include <sstream>
#include <string>
-#include "webrtc/rtc_base/checks.h"
+#include "rtc_base/checks.h"
namespace webrtc {
VideoStream::VideoStream()
diff --git a/call/video_config.h b/call/video_config.h
index 4a98d01..5f660b7 100644
--- a/call/video_config.h
+++ b/call/video_config.h
@@ -8,18 +8,18 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_VIDEO_CONFIG_H_
-#define WEBRTC_CALL_VIDEO_CONFIG_H_
+#ifndef CALL_VIDEO_CONFIG_H_
+#define CALL_VIDEO_CONFIG_H_
#include <string>
#include <vector>
-#include "webrtc/api/optional.h"
-#include "webrtc/common_types.h"
-#include "webrtc/rtc_base/basictypes.h"
-#include "webrtc/rtc_base/refcount.h"
-#include "webrtc/rtc_base/scoped_ref_ptr.h"
-#include "webrtc/typedefs.h"
+#include "api/optional.h"
+#include "common_types.h"
+#include "rtc_base/basictypes.h"
+#include "rtc_base/refcount.h"
+#include "rtc_base/scoped_ref_ptr.h"
+#include "typedefs.h"
namespace webrtc {
@@ -154,4 +154,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_VIDEO_CONFIG_H_
+#endif // CALL_VIDEO_CONFIG_H_
diff --git a/call/video_receive_stream.cc b/call/video_receive_stream.cc
index ef6c608..76892ea 100644
--- a/call/video_receive_stream.cc
+++ b/call/video_receive_stream.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/video_receive_stream.h"
+#include "call/video_receive_stream.h"
namespace webrtc {
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index f199ca1..4a8eca0c 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -8,21 +8,21 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_
-#define WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_
+#ifndef CALL_VIDEO_RECEIVE_STREAM_H_
+#define CALL_VIDEO_RECEIVE_STREAM_H_
#include <limits>
#include <map>
#include <string>
#include <vector>
-#include "webrtc/api/call/transport.h"
-#include "webrtc/api/rtpparameters.h"
-#include "webrtc/call/rtp_config.h"
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/rtc_base/platform_file.h"
+#include "api/call/transport.h"
+#include "api/rtpparameters.h"
+#include "call/rtp_config.h"
+#include "common_types.h"
+#include "common_video/include/frame_callback.h"
+#include "media/base/videosinkinterface.h"
+#include "rtc_base/platform_file.h"
namespace webrtc {
@@ -257,4 +257,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_VIDEO_RECEIVE_STREAM_H_
+#endif // CALL_VIDEO_RECEIVE_STREAM_H_
diff --git a/call/video_send_stream.cc b/call/video_send_stream.cc
index f6ea13f..4f8c059 100644
--- a/call/video_send_stream.cc
+++ b/call/video_send_stream.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "webrtc/call/video_send_stream.h"
+#include "call/video_send_stream.h"
namespace webrtc {
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 6dd13ab..d4bd981 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -8,23 +8,23 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef WEBRTC_CALL_VIDEO_SEND_STREAM_H_
-#define WEBRTC_CALL_VIDEO_SEND_STREAM_H_
+#ifndef CALL_VIDEO_SEND_STREAM_H_
+#define CALL_VIDEO_SEND_STREAM_H_
#include <map>
#include <string>
#include <utility>
#include <vector>
-#include "webrtc/api/call/transport.h"
-#include "webrtc/api/rtpparameters.h"
-#include "webrtc/call/rtp_config.h"
-#include "webrtc/call/video_config.h"
-#include "webrtc/common_types.h"
-#include "webrtc/common_video/include/frame_callback.h"
-#include "webrtc/media/base/videosinkinterface.h"
-#include "webrtc/media/base/videosourceinterface.h"
-#include "webrtc/rtc_base/platform_file.h"
+#include "api/call/transport.h"
+#include "api/rtpparameters.h"
+#include "call/rtp_config.h"
+#include "call/video_config.h"
+#include "common_types.h"
+#include "common_video/include/frame_callback.h"
+#include "media/base/videosinkinterface.h"
+#include "media/base/videosourceinterface.h"
+#include "rtc_base/platform_file.h"
namespace webrtc {
@@ -283,4 +283,4 @@
} // namespace webrtc
-#endif // WEBRTC_CALL_VIDEO_SEND_STREAM_H_
+#endif // CALL_VIDEO_SEND_STREAM_H_
