(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}

api/BUILD.gn

@@ -40,7 +40,7 @@ rtc_source_set("call_api") {
 rtc_source_set("callfactory_api") {
   visibility = [ "*" ]
   sources = [
-    "call/callfactoryinterface.h",
+    "call/call_factory_interface.h",
   ]
 }
 
@@ -77,59 +77,59 @@ rtc_static_library("libjingle_peerconnection_api") {
   visibility = [ "*" ]
   cflags = []
   sources = [
-    "asyncresolverfactory.h",
+    "async_resolver_factory.h",
     "bitrate_constraints.h",
     "candidate.cc",
     "candidate.h",
-    "crypto/cryptooptions.cc",
-    "crypto/cryptooptions.h",
-    "crypto/framedecryptorinterface.h",
-    "crypto/frameencryptorinterface.h",
-    "cryptoparams.h",
-    "datachannelinterface.cc",
-    "datachannelinterface.h",
-    "dtlstransportinterface.h",
-    "dtmfsenderinterface.h",
+    "crypto/crypto_options.cc",
+    "crypto/crypto_options.h",
+    "crypto/frame_decryptor_interface.h",
+    "crypto/frame_encryptor_interface.h",
+    "crypto_params.h",
+    "data_channel_interface.cc",
+    "data_channel_interface.h",
+    "dtls_transport_interface.h",
+    "dtmf_sender_interface.h",
     "jsep.cc",
     "jsep.h",
-    "jsepicecandidate.cc",
-    "jsepicecandidate.h",
-    "jsepsessiondescription.h",
+    "jsep_ice_candidate.cc",
+    "jsep_ice_candidate.h",
+    "jsep_session_description.h",
+    "media_constraints_interface.cc",
+    "media_constraints_interface.h",
+    "media_stream_interface.cc",
+    "media_stream_interface.h",
+    "media_stream_proxy.h",
+    "media_stream_track_proxy.h",
     "media_transport_interface.cc",
     "media_transport_interface.h",
-    "mediaconstraintsinterface.cc",
-    "mediaconstraintsinterface.h",
-    "mediastreaminterface.cc",
-    "mediastreaminterface.h",
-    "mediastreamproxy.h",
-    "mediastreamtrackproxy.h",
-    "mediatypes.cc",
-    "mediatypes.h",
+    "media_types.cc",
+    "media_types.h",
     "notifier.h",
-    "peerconnectionfactoryproxy.h",
-    "peerconnectioninterface.cc",
-    "peerconnectioninterface.h",
-    "peerconnectionproxy.h",
+    "peer_connection_factory_proxy.h",
+    "peer_connection_interface.cc",
+    "peer_connection_interface.h",
+    "peer_connection_proxy.h",
     "proxy.cc",
     "proxy.h",
-    "rtcerror.cc",
-    "rtcerror.h",
+    "rtc_error.cc",
+    "rtc_error.h",
     "rtp_headers.cc",
     "rtp_headers.h",
-    "rtpparameters.cc",
-    "rtpparameters.h",
-    "rtpreceiverinterface.cc",
-    "rtpreceiverinterface.h",
-    "rtpsenderinterface.cc",
-    "rtpsenderinterface.h",
-    "rtptransceiverinterface.cc",
-    "rtptransceiverinterface.h",
-    "setremotedescriptionobserverinterface.h",
-    "statstypes.cc",
-    "statstypes.h",
-    "turncustomizer.h",
-    "umametrics.h",
-    "videosourceproxy.h",
+    "rtp_parameters.cc",
+    "rtp_parameters.h",
+    "rtp_receiver_interface.cc",
+    "rtp_receiver_interface.h",
+    "rtp_sender_interface.cc",
+    "rtp_sender_interface.h",
+    "rtp_transceiver_interface.cc",
+    "rtp_transceiver_interface.h",
+    "set_remote_description_observer_interface.h",
+    "stats_types.cc",
+    "stats_types.h",
+    "turn_customizer.h",
+    "uma_metrics.h",
+    "video_track_source_proxy.h",
   ]
   deps = [
     ":array_view",
@@ -240,16 +240,16 @@ if (rtc_include_tests) {
 rtc_source_set("libjingle_logging_api") {
   visibility = [ "*" ]
   sources = [
-    "rtceventlogoutput.h",
+    "rtc_event_log_output.h",
   ]
 }
 
 rtc_source_set("ortc_api") {
   visibility = [ "*" ]
   sources = [
-    "ortc/packettransportinterface.h",
-    "ortc/rtptransportinterface.h",
-    "ortc/srtptransportinterface.h",
+    "ortc/packet_transport_interface.h",
+    "ortc/rtp_transport_interface.h",
+    "ortc/srtp_transport_interface.h",
   ]
 
   deps = [
@@ -262,10 +262,10 @@ rtc_source_set("rtc_stats_api") {
   visibility = [ "*" ]
   cflags = []
   sources = [
-    "stats/rtcstats.h",
+    "stats/rtc_stats.h",
+    "stats/rtc_stats_collector_callback.h",
+    "stats/rtc_stats_report.h",
     "stats/rtcstats_objects.h",
-    "stats/rtcstatscollectorcallback.h",
-    "stats/rtcstatsreport.h",
   ]
 
   deps = [
@@ -346,7 +346,7 @@ rtc_source_set("array_view") {
 rtc_source_set("refcountedbase") {
   visibility = [ "*" ]
   sources = [
-    "refcountedbase.h",
+    "ref_counted_base.h",
   ]
   deps = [
     "../rtc_base:rtc_base_approved",
@@ -357,7 +357,7 @@ rtc_source_set("libjingle_peerconnection_test_api") {
   visibility = [ "*" ]
   testonly = true
   sources = [
-    "test/fakeconstraints.h",
+    "test/fake_constraints.h",
   ]
 
   deps = [
@@ -666,8 +666,8 @@ if (rtc_include_tests) {
 
     sources = [
       "array_view_unittest.cc",
-      "rtcerror_unittest.cc",
-      "rtpparameters_unittest.cc",
+      "rtc_error_unittest.cc",
+      "rtp_parameters_unittest.cc",
       "test/loopback_media_transport_unittest.cc",
     ]
 

api/DEPS

@@ -57,13 +57,13 @@ specific_include_rules = {
     "+modules/audio_coding/codecs/opus/audio_encoder_opus.h",
   ],
 
-  "asyncresolverfactory\.h": [
-    "+rtc_base/asyncresolverinterface.h",
+  "async_resolver_factory\.h": [
+    "+rtc_base/async_resolver_interface.h",
   ],
 
   "candidate\.h": [
     "+rtc_base/network_constants.h",
-    "+rtc_base/socketaddress.h",
+    "+rtc_base/socket_address.h",
   ],
 
   "create_peerconnection_factory\.h": [
@@ -71,17 +71,17 @@ specific_include_rules = {
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
-  "datachannelinterface\.h": [
-    "+rtc_base/copyonwritebuffer.h",
-    "+rtc_base/refcount.h",
+  "data_channel_interface\.h": [
+    "+rtc_base/copy_on_write_buffer.h",
+    "+rtc_base/ref_count.h",
   ],
 
-  "dtlstransportinterface\.h": [
-    "+rtc_base/refcount.h",
+  "dtls_transport_interface\.h": [
+    "+rtc_base/ref_count.h",
   ],
 
-  "dtmfsenderinterface\.h": [
-    "+rtc_base/refcount.h",
+  "dtmf_sender_interface\.h": [
+    "+rtc_base/ref_count.h",
   ],
 
   "fec_controller\.h": [
@@ -89,118 +89,118 @@ specific_include_rules = {
   ],
 
   "jsep\.h": [
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
   ],
 
-  "jsepicecandidate\.h": [
-    "+rtc_base/constructormagic.h",
+  "jsep_ice_candidate\.h": [
+    "+rtc_base/constructor_magic.h",
   ],
 
-  "jsepsessiondescription\.h": [
-    "+rtc_base/constructormagic.h",
+  "jsep_session_description\.h": [
+    "+rtc_base/constructor_magic.h",
   ],
 
-  "mediastreaminterface\.h": [
+  "media_stream_interface\.h": [
     "+modules/audio_processing/include/audio_processing_statistics.h",
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
   "media_transport_interface\.h": [
-    "+rtc_base/copyonwritebuffer.h",  # As used by datachannelinterface.h
-    "+rtc_base/networkroute.h",
+    "+rtc_base/copy_on_write_buffer.h",  # As used by datachannelinterface.h
+    "+rtc_base/network_route.h",
     "+rtc_base/deprecation.h",
   ],
 
-  "peerconnectionfactoryproxy\.h": [
+  "peer_connection_factory_proxy\.h": [
     "+rtc_base/bind.h",
   ],
 
-  "peerconnectioninterface\.h": [
+  "peer_connection_interface\.h": [
     "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
-    "+media/base/mediaconfig.h",
-    "+media/base/videocapturer.h",
-    "+media/base/mediaengine.h",
-    "+p2p/base/portallocator.h",
-    "+rtc_base/bitrateallocationstrategy.h",
+    "+media/base/media_config.h",
+    "+media/base/video_capturer.h",
+    "+media/base/media_engine.h",
+    "+p2p/base/port_allocator.h",
+    "+rtc_base/bitrate_allocation_strategy.h",
     "+rtc_base/network.h",
     "+rtc_base/platform_file.h",
-    "+rtc_base/rtccertificate.h",
-    "+rtc_base/rtccertificategenerator.h",
-    "+rtc_base/socketaddress.h",
-    "+rtc_base/sslcertificate.h",
-    "+rtc_base/sslstreamadapter.h",
+    "+rtc_base/rtc_certificate.h",
+    "+rtc_base/rtc_certificate_generator.h",
+    "+rtc_base/socket_address.h",
+    "+rtc_base/ssl_certificate.h",
+    "+rtc_base/ssl_stream_adapter.h",
   ],
 
   "proxy\.h": [
     "+rtc_base/event.h",
-    "+rtc_base/messagehandler.h",  # Inherits from it.
-    "+rtc_base/messagequeue.h",    # Inherits from MessageData.
-    "+rtc_base/refcountedobject.h",
+    "+rtc_base/message_handler.h",  # Inherits from it.
+    "+rtc_base/message_queue.h",    # Inherits from MessageData.
+    "+rtc_base/ref_counted_object.h",
     "+rtc_base/scoped_ref_ptr.h",
     "+rtc_base/thread.h",
   ],
 
-  "refcountedbase\.h": [
-    "+rtc_base/constructormagic.h",
-    "+rtc_base/refcount.h",
-    "+rtc_base/refcounter.h",
+  "ref_counted_base\.h": [
+    "+rtc_base/constructor_magic.h",
+    "+rtc_base/ref_count.h",
+    "+rtc_base/ref_counter.h",
   ],
 
-  "rtcerror\.h": [
+  "rtc_error\.h": [
     "+rtc_base/logging.h",
   ],
 
-  "rtpreceiverinterface\.h": [
-    "+rtc_base/refcount.h",
+  "rtp_receiver_interface\.h": [
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
-  "rtpsenderinterface\.h": [
-    "+rtc_base/refcount.h",
+  "rtp_sender_interface\.h": [
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
-  "rtptransceiverinterface\.h": [
-    "+rtc_base/refcount.h",
+  "rtp_transceiver_interface\.h": [
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
-  "setremotedescriptionobserverinterface\.h": [
-    "+rtc_base/refcount.h",
+  "set_remote_description_observer_interface\.h": [
+    "+rtc_base/ref_count.h",
   ],
 
-  "statstypes\.h": [
-    "+rtc_base/constructormagic.h",
-    "+rtc_base/refcount.h",
+  "stats_types\.h": [
+    "+rtc_base/constructor_magic.h",
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
-    "+rtc_base/stringencode.h",
+    "+rtc_base/string_encode.h",
     "+rtc_base/thread_checker.h",
   ],
 
-  "umametrics\.h": [
-    "+rtc_base/refcount.h",
+  "uma_metrics\.h": [
+    "+rtc_base/ref_count.h",
   ],
 
   "audio_frame\.h": [
-    "+rtc_base/constructormagic.h",
+    "+rtc_base/constructor_magic.h",
   ],
 
   "audio_mixer\.h": [
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
   ],
 
   "audio_decoder\.h": [
     "+rtc_base/buffer.h",
-    "+rtc_base/constructormagic.h",
+    "+rtc_base/constructor_magic.h",
   ],
 
   "audio_decoder_factory\.h": [
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
   ],
 
   "audio_decoder_factory_template\.h": [
-    "+rtc_base/refcountedobject.h",
+    "+rtc_base/ref_counted_object.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
@@ -210,11 +210,11 @@ specific_include_rules = {
   ],
 
   "audio_encoder_factory\.h": [
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
   ],
 
   "audio_encoder_factory_template\.h": [
-    "+rtc_base/refcountedobject.h",
+    "+rtc_base/ref_counted_object.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
@@ -226,22 +226,22 @@ specific_include_rules = {
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
-  "framedecryptorinterface\.h": [
-    "+rtc_base/refcount.h",
+  "frame_decryptor_interface\.h": [
+    "+rtc_base/ref_count.h",
   ],
 
-  "frameencryptorinterface\.h": [
-    "+rtc_base/refcount.h",
+  "frame_encryptor_interface\.h": [
+    "+rtc_base/ref_count.h",
   ],
 
-  "rtcstatscollectorcallback\.h": [
-    "+rtc_base/refcount.h",
+  "rtc_stats_collector_callback\.h": [
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
-  "rtcstatsreport\.h": [
-    "+rtc_base/refcount.h",
-    "+rtc_base/refcountedobject.h",
+  "rtc_stats_report\.h": [
+    "+rtc_base/ref_count.h",
+    "+rtc_base/ref_counted_object.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 
@@ -250,11 +250,11 @@ specific_include_rules = {
   ],
 
   "fake_frame_decryptor\.h": [
-    "+rtc_base/refcountedobject.h",
+    "+rtc_base/ref_counted_object.h",
   ],
 
   "fake_frame_encryptor\.h": [
-    "+rtc_base/refcountedobject.h",
+    "+rtc_base/ref_counted_object.h",
   ],
 
   "mock.*\.h": [
@@ -262,7 +262,7 @@ specific_include_rules = {
   ],
 
   "simulated_network\.h": [
-    "+rtc_base/criticalsection.h",
+    "+rtc_base/critical_section.h",
     "+rtc_base/random.h",
     "+rtc_base/thread_annotations.h",
   ],
@@ -280,7 +280,7 @@ specific_include_rules = {
   ],
 
   "video_encoder_config\.h": [
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 

api/async_resolver_factory.h

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_ASYNCRESOLVERFACTORY_H_
-#define API_ASYNCRESOLVERFACTORY_H_
+#ifndef API_ASYNC_RESOLVER_FACTORY_H_
+#define API_ASYNC_RESOLVER_FACTORY_H_
 
-#include "rtc_base/asyncresolverinterface.h"
+#include "rtc_base/async_resolver_interface.h"
 
 namespace webrtc {
 
@@ -29,4 +29,4 @@ class AsyncResolverFactory {
 
 }  // namespace webrtc
 
-#endif  // API_ASYNCRESOLVERFACTORY_H_
+#endif  // API_ASYNC_RESOLVER_FACTORY_H_

api/audio/audio_frame.cc

@@ -13,7 +13,7 @@
 #include <string.h>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 

api/audio/audio_frame.h

@@ -14,7 +14,7 @@
 #include <stddef.h>
 #include <stdint.h>
 
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 

api/audio/audio_mixer.h

@@ -14,7 +14,7 @@
 #include <memory>
 
 #include "api/audio/audio_frame.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 

api/audio_codecs/audio_decoder.h

@@ -19,7 +19,7 @@
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 

api/audio_codecs/audio_decoder_factory.h

@@ -18,7 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_decoder.h"
 #include "api/audio_codecs/audio_format.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 

api/audio_codecs/audio_decoder_factory_template.h

@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "api/audio_codecs/audio_decoder_factory.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {

api/audio_codecs/audio_encoder_factory.h

@@ -18,7 +18,7 @@
 #include "api/audio_codecs/audio_codec_pair_id.h"
 #include "api/audio_codecs/audio_encoder.h"
 #include "api/audio_codecs/audio_format.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 

api/audio_codecs/audio_encoder_factory_template.h

@@ -15,7 +15,7 @@
 #include <vector>
 
 #include "api/audio_codecs/audio_encoder_factory.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {

api/call/call_factory_interface.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_CALL_CALLFACTORYINTERFACE_H_
-#define API_CALL_CALLFACTORYINTERFACE_H_
+#ifndef API_CALL_CALL_FACTORY_INTERFACE_H_
+#define API_CALL_CALL_FACTORY_INTERFACE_H_
 
 #include <memory>
 
@@ -33,4 +33,4 @@ std::unique_ptr<CallFactoryInterface> CreateCallFactory();
 
 }  // namespace webrtc
 
-#endif  // API_CALL_CALLFACTORYINTERFACE_H_
+#endif  // API_CALL_CALL_FACTORY_INTERFACE_H_

api/candidate.cc

@@ -11,7 +11,7 @@
 #include "api/candidate.h"
 
 #include "rtc_base/helpers.h"
-#include "rtc_base/ipaddress.h"
+#include "rtc_base/ip_address.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace cricket {

api/candidate.h

@@ -19,7 +19,7 @@
 
 #include "rtc_base/checks.h"
 #include "rtc_base/network_constants.h"
-#include "rtc_base/socketaddress.h"
+#include "rtc_base/socket_address.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace cricket {

api/create_peerconnection_factory.cc

@@ -16,16 +16,16 @@
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
-#include "api/call/callfactoryinterface.h"
+#include "api/call/call_factory_interface.h"
 #include "api/fec_controller.h"
-#include "api/peerconnectioninterface.h"
+#include "api/peer_connection_interface.h"
 #include "api/transport/network_control.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "logging/rtc_event_log/rtc_event_log_factory.h"
 #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
-#include "media/base/mediaengine.h"
-#include "media/engine/webrtcmediaengine.h"
+#include "media/base/media_engine.h"
+#include "media/engine/webrtc_media_engine.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "rtc_base/scoped_ref_ptr.h"

api/create_peerconnection_factory.h

@@ -17,7 +17,7 @@
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/fec_controller.h"
-#include "api/peerconnectioninterface.h"
+#include "api/peer_connection_interface.h"
 #include "api/transport/network_control.h"
 #include "rtc_base/deprecation.h"
 #include "rtc_base/scoped_ref_ptr.h"

api/crypto/crypto_options.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/crypto/cryptooptions.h"
-#include "rtc_base/sslstreamadapter.h"
+#include "api/crypto/crypto_options.h"
+#include "rtc_base/ssl_stream_adapter.h"
 
 namespace webrtc {
 

api/crypto/crypto_options.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_CRYPTO_CRYPTOOPTIONS_H_
-#define API_CRYPTO_CRYPTOOPTIONS_H_
+#ifndef API_CRYPTO_CRYPTO_OPTIONS_H_
+#define API_CRYPTO_CRYPTO_OPTIONS_H_
 
 #include <vector>
 
@@ -63,4 +63,4 @@ struct CryptoOptions {
 
 }  // namespace webrtc
 
-#endif  // API_CRYPTO_CRYPTOOPTIONS_H_
+#endif  // API_CRYPTO_CRYPTO_OPTIONS_H_

api/crypto/frame_decryptor_interface.h

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_
-#define API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_
+#ifndef API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
+#define API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
 
 #include <vector>
 
 #include "api/array_view.h"
-#include "api/mediatypes.h"
-#include "rtc_base/refcount.h"
+#include "api/media_types.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -56,4 +56,4 @@ class FrameDecryptorInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_
+#endif  // API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_

api/crypto/frame_encryptor_interface.h

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_
-#define API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_
+#ifndef API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_
+#define API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_
 
 #include "api/array_view.h"
-#include "api/mediatypes.h"
-#include "rtc_base/refcount.h"
+#include "api/media_types.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -52,4 +52,4 @@ class FrameEncryptorInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_
+#endif  // API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_

api/crypto_params.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_CRYPTOPARAMS_H_
-#define API_CRYPTOPARAMS_H_
+#ifndef API_CRYPTO_PARAMS_H_
+#define API_CRYPTO_PARAMS_H_
 
 #include <string>
 
@@ -38,4 +38,4 @@ struct CryptoParams {
 
 }  // namespace cricket
 
-#endif  // API_CRYPTOPARAMS_H_
+#endif  // API_CRYPTO_PARAMS_H_

api/data_channel_interface.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/datachannelinterface.h"
+#include "api/data_channel_interface.h"
 
 namespace webrtc {
 

api/data_channel_interface.h

@@ -11,16 +11,16 @@
 // This file contains interfaces for DataChannels
 // http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
 
-#ifndef API_DATACHANNELINTERFACE_H_
-#define API_DATACHANNELINTERFACE_H_
+#ifndef API_DATA_CHANNEL_INTERFACE_H_
+#define API_DATA_CHANNEL_INTERFACE_H_
 
 #include <stddef.h>
 #include <stdint.h>
 #include <string>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/copyonwritebuffer.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -177,4 +177,4 @@ class DataChannelInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_DATACHANNELINTERFACE_H_
+#endif  // API_DATA_CHANNEL_INTERFACE_H_

api/dtls_transport_interface.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_DTLSTRANSPORTINTERFACE_H_
-#define API_DTLSTRANSPORTINTERFACE_H_
+#ifndef API_DTLS_TRANSPORT_INTERFACE_H_
+#define API_DTLS_TRANSPORT_INTERFACE_H_
 
-#include "api/rtcerror.h"
-#include "rtc_base/refcount.h"
+#include "api/rtc_error.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -66,4 +66,4 @@ class DtlsTransportInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_DTLSTRANSPORTINTERFACE_H_
+#endif  // API_DTLS_TRANSPORT_INTERFACE_H_

api/dtmf_sender_interface.h

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_DTMFSENDERINTERFACE_H_
-#define API_DTMFSENDERINTERFACE_H_
+#ifndef API_DTMF_SENDER_INTERFACE_H_
+#define API_DTMF_SENDER_INTERFACE_H_
 
 #include <string>
 
-#include "api/mediastreaminterface.h"
-#include "rtc_base/refcount.h"
+#include "api/media_stream_interface.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -97,4 +97,4 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_DTMFSENDERINTERFACE_H_
+#endif  // API_DTMF_SENDER_INTERFACE_H_

api/jsep.h

@@ -27,8 +27,8 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "api/rtcerror.h"
-#include "rtc_base/refcount.h"
+#include "api/rtc_error.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace cricket {

api/jsep_ice_candidate.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/jsepicecandidate.h"
+#include "api/jsep_ice_candidate.h"
 
 #include <algorithm>
 #include <utility>

api/jsep_ice_candidate.h

@@ -11,8 +11,8 @@
 // TODO(deadbeef): Move this out of api/; it's an implementation detail and
 // shouldn't be used externally.
 
-#ifndef API_JSEPICECANDIDATE_H_
-#define API_JSEPICECANDIDATE_H_
+#ifndef API_JSEP_ICE_CANDIDATE_H_
+#define API_JSEP_ICE_CANDIDATE_H_
 
 #include <stddef.h>
 #include <string>
@@ -20,7 +20,7 @@
 
 #include "api/candidate.h"
 #include "api/jsep.h"
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
 
 namespace webrtc {
 
@@ -82,4 +82,4 @@ class JsepCandidateCollection : public IceCandidateCollection {
 
 }  // namespace webrtc
 
-#endif  // API_JSEPICECANDIDATE_H_
+#endif  // API_JSEP_ICE_CANDIDATE_H_

api/jsep_session_description.h

@@ -11,8 +11,8 @@
 // TODO(deadbeef): Move this out of api/; it's an implementation detail and
 // shouldn't be used externally.
 
-#ifndef API_JSEPSESSIONDESCRIPTION_H_
-#define API_JSEPSESSIONDESCRIPTION_H_
+#ifndef API_JSEP_SESSION_DESCRIPTION_H_
+#define API_JSEP_SESSION_DESCRIPTION_H_
 
 #include <memory>
 #include <string>
@@ -21,8 +21,8 @@
 #include "absl/strings/string_view.h"
 #include "api/candidate.h"
 #include "api/jsep.h"
-#include "api/jsepicecandidate.h"
-#include "rtc_base/constructormagic.h"
+#include "api/jsep_ice_candidate.h"
+#include "rtc_base/constructor_magic.h"
 
 namespace cricket {
 class SessionDescription;
@@ -88,4 +88,4 @@ class JsepSessionDescription : public SessionDescriptionInterface {
 
 }  // namespace webrtc
 
-#endif  // API_JSEPSESSIONDESCRIPTION_H_
+#endif  // API_JSEP_SESSION_DESCRIPTION_H_

api/media_constraints_interface.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/mediaconstraintsinterface.h"
+#include "api/media_constraints_interface.h"
 
 #include "absl/types/optional.h"
-#include "api/peerconnectioninterface.h"
-#include "media/base/mediaconfig.h"
-#include "rtc_base/stringencode.h"
+#include "api/peer_connection_interface.h"
+#include "media/base/media_config.h"
+#include "rtc_base/string_encode.h"
 
 namespace {
 

api/media_constraints_interface.h

@@ -18,15 +18,15 @@
 // be removed from WebRTC.
 // https://bugs.chromium.org/p/webrtc/issues/detail?id=9239
 
-#ifndef API_MEDIACONSTRAINTSINTERFACE_H_
-#define API_MEDIACONSTRAINTSINTERFACE_H_
+#ifndef API_MEDIA_CONSTRAINTS_INTERFACE_H_
+#define API_MEDIA_CONSTRAINTS_INTERFACE_H_
 
 #include <stddef.h>
 #include <string>
 #include <vector>
 
 #include "api/audio_options.h"
-#include "api/peerconnectioninterface.h"
+#include "api/peer_connection_interface.h"
 
 namespace webrtc {
 
@@ -156,4 +156,4 @@ bool CopyConstraintsIntoOfferAnswerOptions(
 
 }  // namespace webrtc
 
-#endif  // API_MEDIACONSTRAINTSINTERFACE_H_
+#endif  // API_MEDIA_CONSTRAINTS_INTERFACE_H_

api/media_stream_interface.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/mediastreaminterface.h"
+#include "api/media_stream_interface.h"
 
 namespace webrtc {
 

api/media_stream_interface.h

@@ -14,8 +14,8 @@
 // interfaces must be used only with PeerConnection. PeerConnectionManager
 // interface provides the factory methods to create MediaStream and MediaTracks.
 
-#ifndef API_MEDIASTREAMINTERFACE_H_
-#define API_MEDIASTREAMINTERFACE_H_
+#ifndef API_MEDIA_STREAM_INTERFACE_H_
+#define API_MEDIA_STREAM_INTERFACE_H_
 
 #include <stddef.h>
 
@@ -28,7 +28,7 @@
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
@@ -294,4 +294,4 @@ class MediaStreamInterface : public rtc::RefCountInterface,
 
 }  // namespace webrtc
 
-#endif  // API_MEDIASTREAMINTERFACE_H_
+#endif  // API_MEDIA_STREAM_INTERFACE_H_

api/media_stream_proxy.h

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_MEDIASTREAMPROXY_H_
-#define API_MEDIASTREAMPROXY_H_
+#ifndef API_MEDIA_STREAM_PROXY_H_
+#define API_MEDIA_STREAM_PROXY_H_
 
 #include <string>
 
-#include "api/mediastreaminterface.h"
+#include "api/media_stream_interface.h"
 #include "api/proxy.h"
 
 namespace webrtc {
@@ -41,4 +41,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_MEDIASTREAMPROXY_H_
+#endif  // API_MEDIA_STREAM_PROXY_H_

api/media_stream_track_proxy.h

@@ -11,12 +11,12 @@
 // This file includes proxy classes for tracks. The purpose is
 // to make sure tracks are only accessed from the signaling thread.
 
-#ifndef API_MEDIASTREAMTRACKPROXY_H_
-#define API_MEDIASTREAMTRACKPROXY_H_
+#ifndef API_MEDIA_STREAM_TRACK_PROXY_H_
+#define API_MEDIA_STREAM_TRACK_PROXY_H_
 
 #include <string>
 
-#include "api/mediastreaminterface.h"
+#include "api/media_stream_interface.h"
 #include "api/proxy.h"
 
 namespace webrtc {
@@ -62,4 +62,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_MEDIASTREAMTRACKPROXY_H_
+#endif  // API_MEDIA_STREAM_TRACK_PROXY_H_

api/media_transport_interface.h

@@ -25,11 +25,11 @@
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
-#include "api/rtcerror.h"
+#include "api/rtc_error.h"
 #include "api/video/encoded_image.h"
-#include "rtc_base/copyonwritebuffer.h"
+#include "rtc_base/copy_on_write_buffer.h"
 #include "rtc_base/deprecation.h"
-#include "rtc_base/networkroute.h"
+#include "rtc_base/network_route.h"
 
 namespace rtc {
 class PacketTransportInternal;

api/media_types.cc

@@ -8,9 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/mediatypes.h"
+#include "api/media_types.h"
 
-#include "api/mediastreaminterface.h"
+#include "api/media_stream_interface.h"
 #include "rtc_base/checks.h"
 
 namespace {

api/media_types.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_MEDIATYPES_H_
-#define API_MEDIATYPES_H_
+#ifndef API_MEDIA_TYPES_H_
+#define API_MEDIA_TYPES_H_
 
 #include <string>
 
@@ -33,4 +33,4 @@ enum class MediaType { ANY, AUDIO, VIDEO, DATA };
 
 }  // namespace webrtc
 
-#endif  // API_MEDIATYPES_H_
+#endif  // API_MEDIA_TYPES_H_

api/notifier.h

@@ -13,7 +13,7 @@
 
 #include <list>
 
-#include "api/mediastreaminterface.h"
+#include "api/media_stream_interface.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

api/ortc/packet_transport_interface.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_ORTC_PACKETTRANSPORTINTERFACE_H_
-#define API_ORTC_PACKETTRANSPORTINTERFACE_H_
+#ifndef API_ORTC_PACKET_TRANSPORT_INTERFACE_H_
+#define API_ORTC_PACKET_TRANSPORT_INTERFACE_H_
 
 namespace rtc {
 
@@ -36,4 +36,4 @@ class PacketTransportInterface {
 
 }  // namespace webrtc
 
-#endif  // API_ORTC_PACKETTRANSPORTINTERFACE_H_
+#endif  // API_ORTC_PACKET_TRANSPORT_INTERFACE_H_

api/ortc/rtp_transport_interface.h

@@ -8,16 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_ORTC_RTPTRANSPORTINTERFACE_H_
-#define API_ORTC_RTPTRANSPORTINTERFACE_H_
+#ifndef API_ORTC_RTP_TRANSPORT_INTERFACE_H_
+#define API_ORTC_RTP_TRANSPORT_INTERFACE_H_
 
 #include <string>
 
 #include "absl/types/optional.h"
-#include "api/ortc/packettransportinterface.h"
-#include "api/rtcerror.h"
+#include "api/ortc/packet_transport_interface.h"
+#include "api/rtc_error.h"
 #include "api/rtp_headers.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
 
 namespace webrtc {
 
@@ -91,4 +91,4 @@ class RtpTransportInterface {
 
 }  // namespace webrtc
 
-#endif  // API_ORTC_RTPTRANSPORTINTERFACE_H_
+#endif  // API_ORTC_RTP_TRANSPORT_INTERFACE_H_

api/ortc/srtp_transport_interface.h

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_ORTC_SRTPTRANSPORTINTERFACE_H_
-#define API_ORTC_SRTPTRANSPORTINTERFACE_H_
+#ifndef API_ORTC_SRTP_TRANSPORT_INTERFACE_H_
+#define API_ORTC_SRTP_TRANSPORT_INTERFACE_H_
 
-#include "api/cryptoparams.h"
-#include "api/ortc/rtptransportinterface.h"
-#include "api/rtcerror.h"
+#include "api/crypto_params.h"
+#include "api/ortc/rtp_transport_interface.h"
+#include "api/rtc_error.h"
 
 namespace webrtc {
 
@@ -45,4 +45,4 @@ class SrtpTransportInterface : public RtpTransportInterface {
 
 }  // namespace webrtc
 
-#endif  // API_ORTC_SRTPTRANSPORTINTERFACE_H_
+#endif  // API_ORTC_SRTP_TRANSPORT_INTERFACE_H_

api/peer_connection_factory_proxy.h

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_PEERCONNECTIONFACTORYPROXY_H_
-#define API_PEERCONNECTIONFACTORYPROXY_H_
+#ifndef API_PEER_CONNECTION_FACTORY_PROXY_H_
+#define API_PEER_CONNECTION_FACTORY_PROXY_H_
 
 #include <memory>
 #include <string>
 #include <utility>
 
-#include "api/peerconnectioninterface.h"
+#include "api/peer_connection_interface.h"
 #include "api/proxy.h"
 #include "rtc_base/bind.h"
 
@@ -74,4 +74,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_PEERCONNECTIONFACTORYPROXY_H_
+#endif  // API_PEER_CONNECTION_FACTORY_PROXY_H_

api/peer_connection_interface.cc

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/peerconnectioninterface.h"
-#include "api/dtlstransportinterface.h"
+#include "api/peer_connection_interface.h"
+#include "api/dtls_transport_interface.h"
 
 namespace webrtc {
 

api/peer_connection_interface.h

@@ -64,55 +64,55 @@
 // 7. Once a candidate has been gathered, the PeerConnection will call the
 // observer function OnIceCandidate. Send these candidates to the remote peer.
 
-#ifndef API_PEERCONNECTIONINTERFACE_H_
-#define API_PEERCONNECTIONINTERFACE_H_
+#ifndef API_PEER_CONNECTION_INTERFACE_H_
+#define API_PEER_CONNECTION_INTERFACE_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "api/asyncresolverfactory.h"
+#include "api/async_resolver_factory.h"
 #include "api/audio/audio_mixer.h"
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_options.h"
-#include "api/call/callfactoryinterface.h"
-#include "api/crypto/cryptooptions.h"
-#include "api/datachannelinterface.h"
+#include "api/call/call_factory_interface.h"
+#include "api/crypto/crypto_options.h"
+#include "api/data_channel_interface.h"
 #include "api/fec_controller.h"
 #include "api/jsep.h"
+#include "api/media_stream_interface.h"
 #include "api/media_transport_interface.h"
-#include "api/mediastreaminterface.h"
-#include "api/rtcerror.h"
-#include "api/rtceventlogoutput.h"
-#include "api/rtpreceiverinterface.h"
-#include "api/rtpsenderinterface.h"
-#include "api/rtptransceiverinterface.h"
-#include "api/setremotedescriptionobserverinterface.h"
-#include "api/stats/rtcstatscollectorcallback.h"
-#include "api/statstypes.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/set_remote_description_observer_interface.h"
+#include "api/stats/rtc_stats_collector_callback.h"
+#include "api/stats_types.h"
 #include "api/transport/bitrate_settings.h"
 #include "api/transport/network_control.h"
-#include "api/turncustomizer.h"
+#include "api/turn_customizer.h"
 #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
-#include "media/base/mediaconfig.h"
+#include "media/base/media_config.h"
 // TODO(bugs.webrtc.org/6353): cricket::VideoCapturer is deprecated and should
 // be deleted from the PeerConnection api.
-#include "media/base/videocapturer.h"  // nogncheck
+#include "media/base/video_capturer.h"  // nogncheck
 // TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
 // inject a PacketSocketFactory and/or NetworkManager, and not expose
 // PortAllocator in the PeerConnection api.
-#include "media/base/mediaengine.h"  // nogncheck
-#include "p2p/base/portallocator.h"  // nogncheck
+#include "media/base/media_engine.h"  // nogncheck
+#include "p2p/base/port_allocator.h"  // nogncheck
 // TODO(nisse): The interface for bitrate allocation strategy belongs in api/.
-#include "rtc_base/bitrateallocationstrategy.h"
+#include "rtc_base/bitrate_allocation_strategy.h"
 #include "rtc_base/network.h"
 #include "rtc_base/platform_file.h"
-#include "rtc_base/rtccertificate.h"
-#include "rtc_base/rtccertificategenerator.h"
-#include "rtc_base/socketaddress.h"
-#include "rtc_base/sslcertificate.h"
-#include "rtc_base/sslstreamadapter.h"
+#include "rtc_base/rtc_certificate.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/socket_address.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace rtc {
@@ -1462,4 +1462,4 @@ CreateModularPeerConnectionFactory(
 
 }  // namespace webrtc
 
-#endif  // API_PEERCONNECTIONINTERFACE_H_
+#endif  // API_PEER_CONNECTION_INTERFACE_H_

api/peer_connection_proxy.h

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_PEERCONNECTIONPROXY_H_
-#define API_PEERCONNECTIONPROXY_H_
+#ifndef API_PEER_CONNECTION_PROXY_H_
+#define API_PEER_CONNECTION_PROXY_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
-#include "api/peerconnectioninterface.h"
+#include "api/peer_connection_interface.h"
 #include "api/proxy.h"
 
 namespace webrtc {
@@ -141,4 +141,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_PEERCONNECTIONPROXY_H_
+#endif  // API_PEER_CONNECTION_PROXY_H_

api/proxy.h

@@ -57,9 +57,9 @@
 #include <utility>
 
 #include "rtc_base/event.h"
-#include "rtc_base/messagehandler.h"
-#include "rtc_base/messagequeue.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/message_handler.h"
+#include "rtc_base/message_queue.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/scoped_ref_ptr.h"
 #include "rtc_base/thread.h"
 

api/ref_counted_base.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 API_REFCOUNTEDBASE_H_
-#define API_REFCOUNTEDBASE_H_
+#ifndef API_REF_COUNTED_BASE_H_
+#define API_REF_COUNTED_BASE_H_
 
-#include "rtc_base/constructormagic.h"
-#include "rtc_base/refcount.h"
-#include "rtc_base/refcounter.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/ref_counter.h"
 
 namespace rtc {
 
@@ -40,4 +40,4 @@ class RefCountedBase {
 
 }  // namespace rtc
 
-#endif  // API_REFCOUNTEDBASE_H_
+#endif  // API_REF_COUNTED_BASE_H_

api/rtc_error.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/rtcerror.h"
+#include "api/rtc_error.h"
 
 #include "rtc_base/arraysize.h"
 

api/rtc_error.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_RTCERROR_H_
-#define API_RTCERROR_H_
+#ifndef API_RTC_ERROR_H_
+#define API_RTC_ERROR_H_
 
 #ifdef UNIT_TEST
 #include <ostream>
@@ -291,4 +291,4 @@ class RTCErrorOr {
 
 }  // namespace webrtc
 
-#endif  // API_RTCERROR_H_
+#endif  // API_RTC_ERROR_H_

api/rtc_error_unittest.cc

@@ -10,7 +10,7 @@
 
 #include <utility>
 
-#include "api/rtcerror.h"
+#include "api/rtc_error.h"
 #include "test/gtest.h"
 
 namespace {

api/rtc_event_log_output.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_RTCEVENTLOGOUTPUT_H_
-#define API_RTCEVENTLOGOUTPUT_H_
+#ifndef API_RTC_EVENT_LOG_OUTPUT_H_
+#define API_RTC_EVENT_LOG_OUTPUT_H_
 
 #include <string>
 
@@ -36,4 +36,4 @@ class RtcEventLogOutput {
 
 }  // namespace webrtc
 
-#endif  // API_RTCEVENTLOGOUTPUT_H_
+#endif  // API_RTC_EVENT_LOG_OUTPUT_H_

api/rtp_parameters.cc

@@ -7,7 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
 
 #include <algorithm>
 #include <string>

api/rtp_parameters.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_RTPPARAMETERS_H_
-#define API_RTPPARAMETERS_H_
+#ifndef API_RTP_PARAMETERS_H_
+#define API_RTP_PARAMETERS_H_
 
 #include <stdint.h>
 #include <string>
@@ -17,7 +17,7 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "api/mediatypes.h"
+#include "api/media_types.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -673,4 +673,4 @@ struct RTC_EXPORT RtpParameters {
 
 }  // namespace webrtc
 
-#endif  // API_RTPPARAMETERS_H_
+#endif  // API_RTP_PARAMETERS_H_

api/rtp_parameters_unittest.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
 #include "test/gtest.h"
 
 namespace webrtc {

api/rtp_receiver_interface.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/rtpreceiverinterface.h"
+#include "api/rtp_receiver_interface.h"
 
 namespace webrtc {
 

api/rtp_receiver_interface.h

@@ -11,18 +11,18 @@
 // This file contains interfaces for RtpReceivers
 // http://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
 
-#ifndef API_RTPRECEIVERINTERFACE_H_
-#define API_RTPRECEIVERINTERFACE_H_
+#ifndef API_RTP_RECEIVER_INTERFACE_H_
+#define API_RTP_RECEIVER_INTERFACE_H_
 
 #include <string>
 #include <vector>
 
-#include "api/crypto/framedecryptorinterface.h"
-#include "api/mediastreaminterface.h"
-#include "api/mediatypes.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/media_types.h"
 #include "api/proxy.h"
-#include "api/rtpparameters.h"
-#include "rtc_base/refcount.h"
+#include "api/rtp_parameters.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
@@ -164,4 +164,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_RTPRECEIVERINTERFACE_H_
+#endif  // API_RTP_RECEIVER_INTERFACE_H_

api/rtp_sender_interface.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/rtpsenderinterface.h"
+#include "api/rtp_sender_interface.h"
 
 namespace webrtc {
 

api/rtp_sender_interface.h

@@ -11,20 +11,20 @@
 // This file contains interfaces for RtpSenders
 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface
 
-#ifndef API_RTPSENDERINTERFACE_H_
-#define API_RTPSENDERINTERFACE_H_
+#ifndef API_RTP_SENDER_INTERFACE_H_
+#define API_RTP_SENDER_INTERFACE_H_
 
 #include <string>
 #include <vector>
 
-#include "api/crypto/frameencryptorinterface.h"
-#include "api/dtmfsenderinterface.h"
-#include "api/mediastreaminterface.h"
-#include "api/mediatypes.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/dtmf_sender_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/media_types.h"
 #include "api/proxy.h"
-#include "api/rtcerror.h"
-#include "api/rtpparameters.h"
-#include "rtc_base/refcount.h"
+#include "api/rtc_error.h"
+#include "api/rtp_parameters.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
@@ -108,4 +108,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_RTPSENDERINTERFACE_H_
+#endif  // API_RTP_SENDER_INTERFACE_H_

api/rtp_transceiver_interface.cc

@@ -8,7 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/rtptransceiverinterface.h"
+#include "api/rtp_transceiver_interface.h"
 
 #include "rtc_base/checks.h"
 

api/rtp_transceiver_interface.h

@@ -8,19 +8,19 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_RTPTRANSCEIVERINTERFACE_H_
-#define API_RTPTRANSCEIVERINTERFACE_H_
+#ifndef API_RTP_TRANSCEIVER_INTERFACE_H_
+#define API_RTP_TRANSCEIVER_INTERFACE_H_
 
 #include <string>
 #include <vector>
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
-#include "api/mediatypes.h"
-#include "api/rtpparameters.h"
-#include "api/rtpreceiverinterface.h"
-#include "api/rtpsenderinterface.h"
-#include "rtc_base/refcount.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
+#include "api/rtp_sender_interface.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
@@ -137,4 +137,4 @@ class RtpTransceiverInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_RTPTRANSCEIVERINTERFACE_H_
+#endif  // API_RTP_TRANSCEIVER_INTERFACE_H_

api/set_remote_description_observer_interface.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_SETREMOTEDESCRIPTIONOBSERVERINTERFACE_H_
-#define API_SETREMOTEDESCRIPTIONOBSERVERINTERFACE_H_
+#ifndef API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_
+#define API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_
 
-#include "api/rtcerror.h"
-#include "rtc_base/refcount.h"
+#include "api/rtc_error.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -28,4 +28,4 @@ class SetRemoteDescriptionObserverInterface : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_SETREMOTEDESCRIPTIONOBSERVERINTERFACE_H_
+#endif  // API_SET_REMOTE_DESCRIPTION_OBSERVER_INTERFACE_H_

api/stats/rtc_stats.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_STATS_RTCSTATS_H_
-#define API_STATS_RTCSTATS_H_
+#ifndef API_STATS_RTC_STATS_H_
+#define API_STATS_RTC_STATS_H_
 
 #include <stddef.h>
 #include <stdint.h>
@@ -348,4 +348,4 @@ class RTCNonStandardStatsMember : public RTCStatsMember<T> {
 };
 }  // namespace webrtc
 
-#endif  // API_STATS_RTCSTATS_H_
+#endif  // API_STATS_RTC_STATS_H_

api/stats/rtc_stats_collector_callback.h

@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_STATS_RTCSTATSCOLLECTORCALLBACK_H_
-#define API_STATS_RTCSTATSCOLLECTORCALLBACK_H_
+#ifndef API_STATS_RTC_STATS_COLLECTOR_CALLBACK_H_
+#define API_STATS_RTC_STATS_COLLECTOR_CALLBACK_H_
 
-#include "api/stats/rtcstatsreport.h"
-#include "rtc_base/refcount.h"
+#include "api/stats/rtc_stats_report.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {
@@ -27,4 +27,4 @@ class RTCStatsCollectorCallback : public virtual rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_STATS_RTCSTATSCOLLECTORCALLBACK_H_
+#endif  // API_STATS_RTC_STATS_COLLECTOR_CALLBACK_H_

api/stats/rtc_stats_report.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_STATS_RTCSTATSREPORT_H_
-#define API_STATS_RTCSTATSREPORT_H_
+#ifndef API_STATS_RTC_STATS_REPORT_H_
+#define API_STATS_RTC_STATS_REPORT_H_
 
 #include <stddef.h>
 #include <stdint.h>
@@ -18,9 +18,9 @@
 #include <string>
 #include <vector>
 
-#include "api/stats/rtcstats.h"
-#include "rtc_base/refcount.h"
-#include "rtc_base/refcountedobject.h"
+#include "api/stats/rtc_stats.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/scoped_ref_ptr.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -117,4 +117,4 @@ class RTC_EXPORT RTCStatsReport : public rtc::RefCountInterface {
 
 }  // namespace webrtc
 
-#endif  // API_STATS_RTCSTATSREPORT_H_
+#endif  // API_STATS_RTC_STATS_REPORT_H_

api/stats/rtcstats_objects.h

@@ -16,7 +16,7 @@
 #include <string>
 #include <vector>
 
-#include "api/stats/rtcstats.h"
+#include "api/stats/rtc_stats.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {

api/stats_types.cc

@@ -8,12 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "api/statstypes.h"
+#include "api/stats_types.h"
 
 #include <string.h>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/ref_counted_object.h"
 
 // TODO(tommi): Could we have a static map of value name -> expected type
 // and use this to RTC_DCHECK on correct usage (somewhat strongly typed values)?

api/stats_types.h

@@ -11,8 +11,8 @@
 // This file contains structures used for retrieving statistics from an ongoing
 // libjingle session.
 
-#ifndef API_STATSTYPES_H_
-#define API_STATSTYPES_H_
+#ifndef API_STATS_TYPES_H_
+#define API_STATS_TYPES_H_
 
 #include <algorithm>
 #include <list>
@@ -20,10 +20,10 @@
 #include <string>
 #include <vector>
 
-#include "rtc_base/constructormagic.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
-#include "rtc_base/stringencode.h"
+#include "rtc_base/string_encode.h"
 #include "rtc_base/thread_checker.h"
 
 namespace webrtc {
@@ -449,4 +449,4 @@ class StatsCollection {
 
 }  // namespace webrtc
 
-#endif  // API_STATSTYPES_H_
+#endif  // API_STATS_TYPES_H_

api/test/DEPS

@@ -5,12 +5,12 @@ specific_include_rules = {
   ".*": [
     "+video"
   ],
-  "fakeconstraints\.h": [
-    "+rtc_base/stringencode.h",
+  "fake_constraints\.h": [
+    "+rtc_base/string_encode.h",
   ],
   "loopback_media_transport\.h": [
-    "+rtc_base/asyncinvoker.h",
-    "+rtc_base/criticalsection.h",
+    "+rtc_base/async_invoker.h",
+    "+rtc_base/critical_section.h",
     "+rtc_base/thread.h",
     "+rtc_base/thread_checker.h",
   ],

api/test/fake_constraints.h

@@ -8,14 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_TEST_FAKECONSTRAINTS_H_
-#define API_TEST_FAKECONSTRAINTS_H_
+#ifndef API_TEST_FAKE_CONSTRAINTS_H_
+#define API_TEST_FAKE_CONSTRAINTS_H_
 
 #include <string>
 #include <vector>
 
-#include "api/mediaconstraintsinterface.h"
-#include "rtc_base/stringencode.h"
+#include "api/media_constraints_interface.h"
+#include "rtc_base/string_encode.h"
 
 namespace webrtc {
 
@@ -109,4 +109,4 @@ class FakeConstraints : public webrtc::MediaConstraintsInterface {
 
 }  // namespace webrtc
 
-#endif  // API_TEST_FAKECONSTRAINTS_H_
+#endif  // API_TEST_FAKE_CONSTRAINTS_H_

api/test/fake_frame_decryptor.h

@@ -16,9 +16,9 @@
 #include <vector>
 
 #include "api/array_view.h"
-#include "api/crypto/framedecryptorinterface.h"
-#include "api/mediatypes.h"
-#include "rtc_base/refcountedobject.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/media_types.h"
+#include "rtc_base/ref_counted_object.h"
 
 namespace webrtc {
 

api/test/fake_frame_encryptor.h

@@ -15,9 +15,9 @@
 #include <stdint.h>
 
 #include "api/array_view.h"
-#include "api/crypto/frameencryptorinterface.h"
-#include "api/mediatypes.h"
-#include "rtc_base/refcountedobject.h"
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/media_types.h"
+#include "rtc_base/ref_counted_object.h"
 
 namespace webrtc {
 

api/test/loopback_media_transport.cc

@@ -11,7 +11,7 @@
 #include "api/test/loopback_media_transport.h"
 
 #include "absl/memory/memory.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 

api/test/loopback_media_transport.h

@@ -17,8 +17,8 @@
 
 #include "absl/memory/memory.h"
 #include "api/media_transport_interface.h"
-#include "rtc_base/asyncinvoker.h"
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/async_invoker.h"
+#include "rtc_base/critical_section.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_checker.h"
 

api/test/mock_frame_decryptor.h

@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "api/crypto/framedecryptorinterface.h"
+#include "api/crypto/frame_decryptor_interface.h"
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/mock_frame_encryptor.h

@@ -11,7 +11,7 @@
 #ifndef API_TEST_MOCK_FRAME_ENCRYPTOR_H_
 #define API_TEST_MOCK_FRAME_ENCRYPTOR_H_
 
-#include "api/crypto/frameencryptorinterface.h"
+#include "api/crypto/frame_encryptor_interface.h"
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/mock_peerconnectioninterface.h

@@ -16,7 +16,7 @@
 #include <utility>
 #include <vector>
 
-#include "api/peerconnectioninterface.h"
+#include "api/peer_connection_interface.h"
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/mock_rtpreceiver.h

@@ -14,7 +14,7 @@
 #include <string>
 #include <vector>
 
-#include "api/rtpreceiverinterface.h"
+#include "api/rtp_receiver_interface.h"
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/mock_rtpsender.h

@@ -14,7 +14,7 @@
 #include <string>
 #include <vector>
 
-#include "api/rtpsenderinterface.h"
+#include "api/rtp_sender_interface.h"
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/simulated_network.h

@@ -18,7 +18,7 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/critical_section.h"
 #include "rtc_base/random.h"
 #include "rtc_base/thread_annotations.h"
 

api/test/video_quality_test_fixture.h

@@ -18,7 +18,7 @@
 
 #include "api/bitrate_constraints.h"
 #include "api/fec_controller.h"
-#include "api/mediatypes.h"
+#include "api/media_types.h"
 #include "api/test/simulated_network.h"
 #include "api/video_codecs/video_encoder_config.h"
 

api/turn_customizer.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_TURNCUSTOMIZER_H_
-#define API_TURNCUSTOMIZER_H_
+#ifndef API_TURN_CUSTOMIZER_H_
+#define API_TURN_CUSTOMIZER_H_
 
 #include <stdlib.h>
 
@@ -42,4 +42,4 @@ class TurnCustomizer {
 
 }  // namespace webrtc
 
-#endif  // API_TURNCUSTOMIZER_H_
+#endif  // API_TURN_CUSTOMIZER_H_

api/uma_metrics.h

@@ -10,10 +10,10 @@
 
 // This file contains enums related to IPv4/IPv6 metrics.
 
-#ifndef API_UMAMETRICS_H_
-#define API_UMAMETRICS_H_
+#ifndef API_UMA_METRICS_H_
+#define API_UMA_METRICS_H_
 
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
@@ -157,4 +157,4 @@ enum AddIceCandidateResult {
 
 }  // namespace webrtc
 
-#endif  // API_UMAMETRICS_H_
+#endif  // API_UMA_METRICS_H_

api/video/DEPS

@@ -21,7 +21,7 @@ specific_include_rules = {
   ],
 
   "video_frame_buffer\.h": [
-    "+rtc_base/refcount.h",
+    "+rtc_base/ref_count.h",
     "+rtc_base/scoped_ref_ptr.h",
   ],
 

api/video/i010_buffer.cc

@@ -13,7 +13,7 @@
 
 #include "api/video/i420_buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/ref_counted_object.h"
 #include "third_party/libyuv/include/libyuv/convert.h"
 #include "third_party/libyuv/include/libyuv/scale.h"
 

api/video/i420_buffer.cc

@@ -14,7 +14,7 @@
 #include <utility>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/ref_counted_object.h"
 #include "third_party/libyuv/include/libyuv/convert.h"
 #include "third_party/libyuv/include/libyuv/planar_functions.h"
 #include "third_party/libyuv/include/libyuv/scale.h"

api/video/video_frame.cc

@@ -11,7 +11,7 @@
 #include "api/video/video_frame.h"
 
 #include "rtc_base/checks.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 

api/video/video_frame_buffer.h

@@ -13,7 +13,7 @@
 
 #include <stdint.h>
 
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {

api/video/video_stream_encoder_interface.h

@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "api/rtpparameters.h"  // For DegradationPreference.
+#include "api/rtp_parameters.h"  // For DegradationPreference.
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
 #include "api/video_codecs/video_encoder.h"

api/video_codecs/builtin_video_decoder_factory.cc

@@ -11,7 +11,7 @@
 #include "api/video_codecs/builtin_video_decoder_factory.h"
 
 #include "absl/memory/memory.h"
-#include "media/engine/internaldecoderfactory.h"
+#include "media/engine/internal_decoder_factory.h"
 
 namespace webrtc {
 

api/video_codecs/builtin_video_encoder_factory.cc

@@ -17,9 +17,9 @@
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_encoder.h"
 #include "media/base/codec.h"
-#include "media/base/mediaconstants.h"
+#include "media/base/media_constants.h"
 #include "media/engine/encoder_simulcast_proxy.h"
-#include "media/engine/internalencoderfactory.h"
+#include "media/engine/internal_encoder_factory.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {

api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc

@@ -32,7 +32,7 @@
 #include "modules/video_coding/include/video_codec_interface.h"
 #include "modules/video_coding/include/video_error_codes.h"
 #include "modules/video_coding/utility/simulcast_rate_allocator.h"
-#include "rtc_base/fakeclock.h"
+#include "rtc_base/fake_clock.h"
 #include "rtc_base/scoped_ref_ptr.h"
 #include "test/field_trial.h"
 #include "test/gmock.h"

api/video_codecs/video_encoder_config.h

@@ -18,7 +18,7 @@
 #include "absl/types/optional.h"
 #include "api/video_codecs/sdp_video_format.h"
 #include "api/video_codecs/video_codec.h"
-#include "rtc_base/refcount.h"
+#include "rtc_base/ref_count.h"
 #include "rtc_base/scoped_ref_ptr.h"
 
 namespace webrtc {

api/video_track_source_proxy.h

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef API_VIDEOSOURCEPROXY_H_
-#define API_VIDEOSOURCEPROXY_H_
+#ifndef API_VIDEO_TRACK_SOURCE_PROXY_H_
+#define API_VIDEO_TRACK_SOURCE_PROXY_H_
 
-#include "api/mediastreaminterface.h"
+#include "api/media_stream_interface.h"
 #include "api/proxy.h"
 
 namespace webrtc {
@@ -38,4 +38,4 @@ END_PROXY_MAP()
 
 }  // namespace webrtc
 
-#endif  // API_VIDEOSOURCEPROXY_H_
+#endif  // API_VIDEO_TRACK_SOURCE_PROXY_H_

audio/audio_level.h

@@ -11,7 +11,7 @@
 #ifndef AUDIO_AUDIO_LEVEL_H_
 #define AUDIO_AUDIO_LEVEL_H_
 
-#include "rtc_base/criticalsection.h"
+#include "rtc_base/critical_section.h"
 #include "rtc_base/thread_annotations.h"
 
 namespace webrtc {

audio/audio_receive_stream.cc

@@ -17,7 +17,7 @@
 #include "api/array_view.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/call/audio_sink.h"
-#include "api/rtpparameters.h"
+#include "api/rtp_parameters.h"
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
 #include "audio/channel_receive.h"
@@ -27,7 +27,7 @@
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/string_builder.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
 
 namespace webrtc {
 

audio/audio_receive_stream.h

@@ -19,7 +19,7 @@
 #include "audio/audio_state.h"
 #include "call/audio_receive_stream.h"
 #include "call/syncable.h"
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
 #include "rtc_base/thread_checker.h"
 
 namespace webrtc {

audio/audio_send_stream.cc

@@ -19,7 +19,7 @@
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/audio_codecs/audio_format.h"
 #include "api/call/transport.h"
-#include "api/crypto/frameencryptorinterface.h"
+#include "api/crypto/frame_encryptor_interface.h"
 #include "audio/audio_state.h"
 #include "audio/channel_send.h"
 #include "audio/conversion.h"
@@ -37,7 +37,7 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/strings/audio_format_to_string.h"
 #include "rtc_base/task_queue.h"
-#include "rtc_base/timeutils.h"
+#include "rtc_base/time_utils.h"
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {

audio/audio_send_stream.h

@@ -20,7 +20,7 @@
 #include "call/audio_state.h"
 #include "call/bitrate_allocator.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "rtc_base/constructormagic.h"
+#include "rtc_base/constructor_magic.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/thread_checker.h"
 

audio/audio_state.cc

@@ -19,7 +19,7 @@
 #include "modules/audio_device/include/audio_device.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/refcountedobject.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/thread.h"
 
 namespace webrtc {