(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}
This commit is contained in:
Steve Anton
2019-01-11 09:11:00 -08:00
parent 1c05765831
commit 10542f21c8
1635 changed files with 5341 additions and 5341 deletions

View File

@ -23,19 +23,19 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/call/callfactoryinterface.h"
#include "api/call/call_factory_interface.h"
#include "api/create_peerconnection_factory.h"
#include "api/datachannelinterface.h"
#include "api/data_channel_interface.h"
#include "api/jsep.h"
#include "api/jsepsessiondescription.h"
#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/peerconnectioninterface.h"
#include "api/rtcerror.h"
#include "api/rtceventlogoutput.h"
#include "api/rtpreceiverinterface.h"
#include "api/rtpsenderinterface.h"
#include "api/rtptransceiverinterface.h"
#include "api/jsep_session_description.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
@ -45,54 +45,54 @@
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/base/codec.h"
#include "media/base/fakevideocapturer.h"
#include "media/base/mediaconfig.h"
#include "media/base/mediaengine.h"
#include "media/base/streamparams.h"
#include "media/base/videocapturer.h"
#include "media/engine/webrtcmediaengine.h"
#include "media/sctp/sctptransportinternal.h"
#include "media/base/fake_video_capturer.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "media/base/video_capturer.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/sctp/sctp_transport_internal.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/fakeportallocator.h"
#include "p2p/base/p2pconstants.h"
#include "p2p/base/fake_port_allocator.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "p2p/base/portallocator.h"
#include "p2p/base/transportdescription.h"
#include "p2p/base/transportinfo.h"
#include "pc/audiotrack.h"
#include "pc/mediasession.h"
#include "pc/mediastream.h"
#include "pc/peerconnection.h"
#include "pc/peerconnectionfactory.h"
#include "pc/rtcstatscollector.h"
#include "pc/rtpsender.h"
#include "pc/sessiondescription.h"
#include "pc/streamcollection.h"
#include "pc/test/fakeaudiocapturemodule.h"
#include "pc/test/fakertccertificategenerator.h"
#include "pc/test/fakevideotracksource.h"
#include "pc/test/mockpeerconnectionobservers.h"
#include "pc/test/testsdpstrings.h"
#include "pc/videotrack.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/audio_track.h"
#include "pc/media_session.h"
#include "pc/media_stream.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_factory.h"
#include "pc/rtc_stats_collector.h"
#include "pc/rtp_sender.h"
#include "pc/session_description.h"
#include "pc/stream_collection.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_rtc_certificate_generator.h"
#include "pc/test/fake_video_track_source.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "pc/test/test_sdp_strings.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/gunit.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/refcountedobject.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/socketaddress.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/string_utils.h"
#include "rtc_base/thread.h"
#include "rtc_base/timeutils.h"
#include "rtc_base/virtualsocketserver.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/fileutils.h"
#include "test/testsupport/file_utils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#include "pc/test/android_test_initializer.h"
#endif
namespace webrtc {