(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:
tools_webrtc/do-renames.sh update all-renames.txt && git cl format
Then manually updating:
tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}diff --git a/call/BUILD.gn b/call/BUILD.gn
index d457052..6db37a4 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -191,8 +191,8 @@
rtc_static_library("call") {
sources = [
"call.cc",
- "callfactory.cc",
- "callfactory.h",
+ "call_factory.cc",
+ "call_factory.h",
"degraded_call.cc",
"degraded_call.h",
"flexfec_receive_stream_impl.cc",
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 476ddab..e850954 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -19,10 +19,10 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/call/transport.h"
-#include "api/crypto/cryptooptions.h"
+#include "api/crypto/crypto_options.h"
#include "api/media_transport_interface.h"
-#include "api/rtpparameters.h"
-#include "api/rtpreceiverinterface.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
#include "call/rtp_config.h"
#include "rtc_base/scoped_ref_ptr.h"
diff --git a/call/audio_send_stream.cc b/call/audio_send_stream.cc
index 303b49c..0a3555b 100644
--- a/call/audio_send_stream.cc
+++ b/call/audio_send_stream.cc
@@ -12,7 +12,7 @@
#include <stddef.h>
-#include "rtc_base/stringencode.h"
+#include "rtc_base/string_encode.h"
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/strings/string_builder.h"
diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
index f34a51d..044ec99 100644
--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
@@ -21,10 +21,10 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/audio_format.h"
#include "api/call/transport.h"
-#include "api/crypto/cryptooptions.h"
-#include "api/crypto/frameencryptorinterface.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_encryptor_interface.h"
#include "api/media_transport_interface.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
#include "call/rtp_config.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "rtc_base/scoped_ref_ptr.h"
diff --git a/call/audio_state.h b/call/audio_state.h
index 104f493..1332c05 100644
--- a/call/audio_state.h
+++ b/call/audio_state.h
@@ -13,7 +13,7 @@
#include "api/audio/audio_mixer.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
#include "rtc_base/scoped_ref_ptr.h"
namespace webrtc {
diff --git a/call/bitrate_allocator.h b/call/bitrate_allocator.h
index 56d8bd5..0259da7 100644
--- a/call/bitrate_allocator.h
+++ b/call/bitrate_allocator.h
@@ -20,7 +20,7 @@
#include <vector>
#include "api/call/bitrate_allocation.h"
-#include "rtc_base/bitrateallocationstrategy.h"
+#include "rtc_base/bitrate_allocation_strategy.h"
#include "rtc_base/sequenced_task_checker.h"
namespace webrtc {
diff --git a/call/call.cc b/call/call.cc
index 3ffea84..a57a41c 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -45,7 +45,7 @@
#include "modules/utility/include/process_thread.h"
#include "modules/video_coding/fec_controller_default.h"
#include "rtc_base/checks.h"
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
@@ -54,7 +54,7 @@
#include "rtc_base/synchronization/rw_lock_wrapper.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/thread_annotations.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/cpu_info.h"
diff --git a/call/call.h b/call/call.h
index c0e31c0..ab834f3 100644
--- a/call/call.h
+++ b/call/call.h
@@ -15,7 +15,7 @@
#include <string>
#include <vector>
-#include "api/mediatypes.h"
+#include "api/media_types.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call_config.h"
@@ -24,10 +24,10 @@
#include "call/rtp_transport_controller_send_interface.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
-#include "rtc_base/bitrateallocationstrategy.h"
-#include "rtc_base/copyonwritebuffer.h"
+#include "rtc_base/bitrate_allocation_strategy.h"
+#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
-#include "rtc_base/networkroute.h"
+#include "rtc_base/network_route.h"
namespace webrtc {
diff --git a/call/call_config.h b/call/call_config.h
index 5834175..67ccd51 100644
--- a/call/call_config.h
+++ b/call/call_config.h
@@ -12,7 +12,7 @@
#include "api/bitrate_constraints.h"
#include "api/fec_controller.h"
-#include "api/rtcerror.h"
+#include "api/rtc_error.h"
#include "api/transport/network_control.h"
#include "call/audio_state.h"
diff --git a/call/call_factory.cc b/call/call_factory.cc
index 4c46a40..309063d 100644
--- a/call/call_factory.cc
+++ b/call/call_factory.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "call/callfactory.h"
+#include "call/call_factory.h"
#include <stdio.h>
#include <memory>
diff --git a/call/call_factory.h b/call/call_factory.h
index 448ffdc..f0d695c 100644
--- a/call/call_factory.h
+++ b/call/call_factory.h
@@ -8,10 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef CALL_CALLFACTORY_H_
-#define CALL_CALLFACTORY_H_
+#ifndef CALL_CALL_FACTORY_H_
+#define CALL_CALL_FACTORY_H_
-#include "api/call/callfactoryinterface.h"
+#include "api/call/call_factory_interface.h"
#include "call/call.h"
#include "call/call_config.h"
@@ -25,4 +25,4 @@
} // namespace webrtc
-#endif // CALL_CALLFACTORY_H_
+#endif // CALL_CALL_FACTORY_H_
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 72aa78c..c584bff 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -27,7 +27,7 @@
#include "modules/audio_device/include/test_audio_device.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "rtc_base/bitrateallocationstrategy.h"
+#include "rtc_base/bitrate_allocation_strategy.h"
#include "rtc_base/checks.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/metrics.h"
@@ -43,7 +43,7 @@
#include "test/null_transport.h"
#include "test/rtp_rtcp_observer.h"
#include "test/single_threaded_task_queue.h"
-#include "test/testsupport/fileutils.h"
+#include "test/testsupport/file_utils.h"
#include "test/testsupport/perf_test.h"
#include "test/video_encoder_proxy_factory.h"
#include "video/transport_adapter.h"
diff --git a/call/degraded_call.h b/call/degraded_call.h
index a5db264..8f062c3 100644
--- a/call/degraded_call.h
+++ b/call/degraded_call.h
@@ -18,7 +18,7 @@
#include "absl/types/optional.h"
#include "api/call/transport.h"
#include "api/fec_controller.h"
-#include "api/mediatypes.h"
+#include "api/media_types.h"
#include "api/rtp_headers.h"
#include "api/test/simulated_network.h"
#include "api/video_codecs/video_encoder_config.h"
@@ -33,8 +33,8 @@
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "modules/utility/include/process_thread.h"
-#include "rtc_base/bitrateallocationstrategy.h"
-#include "rtc_base/copyonwritebuffer.h"
+#include "rtc_base/bitrate_allocation_strategy.h"
+#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/network/sent_packet.h"
#include "system_wrappers/include/clock.h"
diff --git a/call/fake_network_pipe.cc b/call/fake_network_pipe.cc
index 7c3c3d2..b5c0cb5 100644
--- a/call/fake_network_pipe.cc
+++ b/call/fake_network_pipe.cc
@@ -14,7 +14,7 @@
#include <utility>
#include <vector>
-#include "api/mediatypes.h"
+#include "api/media_types.h"
#include "call/fake_network_pipe.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/checks.h"
diff --git a/call/fake_network_pipe.h b/call/fake_network_pipe.h
index 34e1b50..2c41dbf 100644
--- a/call/fake_network_pipe.h
+++ b/call/fake_network_pipe.h
@@ -23,8 +23,8 @@
#include "api/test/simulated_network.h"
#include "call/call.h"
#include "call/simulated_packet_receiver.h"
-#include "rtc_base/constructormagic.h"
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
diff --git a/call/flexfec_receive_stream.h b/call/flexfec_receive_stream.h
index 949ad74..77c9932 100644
--- a/call/flexfec_receive_stream.h
+++ b/call/flexfec_receive_stream.h
@@ -17,7 +17,7 @@
#include "api/call/transport.h"
#include "api/rtp_headers.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
#include "call/rtp_packet_sink_interface.h"
namespace webrtc {
diff --git a/call/flexfec_receive_stream_impl.cc b/call/flexfec_receive_stream_impl.cc
index 3b33be6..8154715 100644
--- a/call/flexfec_receive_stream_impl.cc
+++ b/call/flexfec_receive_stream_impl.cc
@@ -17,7 +17,7 @@
#include "api/array_view.h"
#include "api/call/transport.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.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"
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc
index d73a804..a1bacf9 100644
--- a/call/flexfec_receive_stream_unittest.cc
+++ b/call/flexfec_receive_stream_unittest.cc
@@ -16,7 +16,7 @@
#include "api/array_view.h"
#include "api/call/transport.h"
#include "api/rtp_headers.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
#include "call/flexfec_receive_stream.h"
#include "call/flexfec_receive_stream_impl.h"
#include "call/rtp_stream_receiver_controller.h"
diff --git a/call/packet_receiver.h b/call/packet_receiver.h
index 02a0155..df57d8f 100644
--- a/call/packet_receiver.h
+++ b/call/packet_receiver.h
@@ -15,8 +15,8 @@
#include <string>
#include <vector>
-#include "api/mediatypes.h"
-#include "rtc_base/copyonwritebuffer.h"
+#include "api/media_types.h"
+#include "rtc_base/copy_on_write_buffer.h"
namespace webrtc {
diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc
index f629dbe..91788b0 100644
--- a/call/rampup_tests.cc
+++ b/call/rampup_tests.cc
@@ -17,7 +17,7 @@
#include "rtc_base/flags.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
-#include "rtc_base/stringencode.h"
+#include "rtc_base/string_encode.h"
#include "test/encoder_settings.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
diff --git a/call/receive_time_calculator_unittest.cc b/call/receive_time_calculator_unittest.cc
index 3de759c..4349ee4 100644
--- a/call/receive_time_calculator_unittest.cc
+++ b/call/receive_time_calculator_unittest.cc
@@ -18,7 +18,7 @@
#include "absl/types/optional.h"
#include "rtc_base/random.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
#include "test/gtest.h"
namespace webrtc {
diff --git a/call/rtp_bitrate_configurator.h b/call/rtp_bitrate_configurator.h
index 1981292..a16dda8 100644
--- a/call/rtp_bitrate_configurator.h
+++ b/call/rtp_bitrate_configurator.h
@@ -14,7 +14,7 @@
#include "absl/types/optional.h"
#include "api/bitrate_constraints.h"
#include "api/transport/bitrate_settings.h"
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
namespace webrtc {
diff --git a/call/rtp_config.h b/call/rtp_config.h
index 06a89f2..b6155c0 100644
--- a/call/rtp_config.h
+++ b/call/rtp_config.h
@@ -17,7 +17,7 @@
#include <vector>
#include "api/rtp_headers.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
namespace webrtc {
// Currently only VP8/VP9 specific.
diff --git a/call/rtp_payload_params.cc b/call/rtp_payload_params.cc
index 95c64b4..a0f123c 100644
--- a/call/rtp_payload_params.cc
+++ b/call/rtp_payload_params.cc
@@ -25,7 +25,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/random.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
diff --git a/call/rtp_stream_receiver_controller.h b/call/rtp_stream_receiver_controller.h
index c523e3f..045af3c 100644
--- a/call/rtp_stream_receiver_controller.h
+++ b/call/rtp_stream_receiver_controller.h
@@ -14,7 +14,7 @@
#include "call/rtp_demuxer.h"
#include "call/rtp_stream_receiver_controller_interface.h"
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/critical_section.h"
namespace webrtc {
diff --git a/call/rtp_transport_controller_send.h b/call/rtp_transport_controller_send.h
index 5b14a73..2581a77 100644
--- a/call/rtp_transport_controller_send.h
+++ b/call/rtp_transport_controller_send.h
@@ -25,8 +25,8 @@
#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
#include "modules/pacing/packet_router.h"
#include "modules/utility/include/process_thread.h"
-#include "rtc_base/constructormagic.h"
-#include "rtc_base/networkroute.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/network_route.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/task_queue.h"
diff --git a/call/rtp_transport_controller_send_interface.h b/call/rtp_transport_controller_send_interface.h
index 19c1dad..ef6fad2 100644
--- a/call/rtp_transport_controller_send_interface.h
+++ b/call/rtp_transport_controller_send_interface.h
@@ -20,7 +20,7 @@
#include "absl/types/optional.h"
#include "api/bitrate_constraints.h"
-#include "api/crypto/cryptooptions.h"
+#include "api/crypto/crypto_options.h"
#include "api/fec_controller.h"
#include "api/transport/bitrate_settings.h"
#include "call/rtp_config.h"
diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index d72ef0d..f5ef77e 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -27,8 +27,8 @@
#include "modules/rtp_rtcp/include/flexfec_sender.h"
#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "modules/utility/include/process_thread.h"
-#include "rtc_base/constructormagic.h"
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/critical_section.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/thread_checker.h"
diff --git a/call/simulated_network.h b/call/simulated_network.h
index 7d9d15b..0336bfc 100644
--- a/call/simulated_network.h
+++ b/call/simulated_network.h
@@ -17,7 +17,7 @@
#include "absl/types/optional.h"
#include "api/test/simulated_network.h"
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/critical_section.h"
#include "rtc_base/random.h"
#include "rtc_base/thread_annotations.h"
diff --git a/call/test/mock_rtp_transport_controller_send.h b/call/test/mock_rtp_transport_controller_send.h
index c236041..18bb117 100644
--- a/call/test/mock_rtp_transport_controller_send.h
+++ b/call/test/mock_rtp_transport_controller_send.h
@@ -17,13 +17,13 @@
#include <vector>
#include "api/bitrate_constraints.h"
-#include "api/crypto/cryptooptions.h"
-#include "api/crypto/frameencryptorinterface.h"
+#include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_encryptor_interface.h"
#include "call/rtp_transport_controller_send_interface.h"
#include "modules/congestion_controller/include/network_changed_observer.h"
#include "modules/pacing/packet_router.h"
#include "rtc_base/network/sent_packet.h"
-#include "rtc_base/networkroute.h"
+#include "rtc_base/network_route.h"
#include "rtc_base/rate_limiter.h"
#include "test/gmock.h"
diff --git a/call/video_receive_stream.h b/call/video_receive_stream.h
index 27f6e95..b469553 100644
--- a/call/video_receive_stream.h
+++ b/call/video_receive_stream.h
@@ -17,11 +17,11 @@
#include <vector>
#include "api/call/transport.h"
-#include "api/crypto/cryptooptions.h"
+#include "api/crypto/crypto_options.h"
#include "api/media_transport_interface.h"
#include "api/rtp_headers.h"
-#include "api/rtpparameters.h"
-#include "api/rtpreceiverinterface.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
#include "api/video/video_content_type.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_timing.h"
diff --git a/call/video_send_stream.cc b/call/video_send_stream.cc
index ebd37a7..8d3f62c 100644
--- a/call/video_send_stream.cc
+++ b/call/video_send_stream.cc
@@ -12,7 +12,7 @@
#include <utility>
-#include "api/crypto/frameencryptorinterface.h"
+#include "api/crypto/frame_encryptor_interface.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
diff --git a/call/video_send_stream.h b/call/video_send_stream.h
index 8ee2991..5daec19 100644
--- a/call/video_send_stream.h
+++ b/call/video_send_stream.h
@@ -18,9 +18,9 @@
#include "absl/types/optional.h"
#include "api/call/transport.h"
-#include "api/crypto/cryptooptions.h"
+#include "api/crypto/crypto_options.h"
#include "api/media_transport_interface.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame.h"
#include "api/video/video_sink_interface.h"