Remove unused critical section includes.

Bug: webrtc:11567
Change-Id: Ic5e43c51ce06c0619adc265d12ad4bef73a9df76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179521
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31745}

api/BUILD.gn

@@ -579,10 +579,7 @@ rtc_source_set("bitrate_allocation") {
 rtc_source_set("simulated_network_api") {
   visibility = [ "*" ]
   sources = [ "test/simulated_network.h" ]
-  deps = [
+  deps = [ "../rtc_base" ]
-    "../rtc_base",
-    "../rtc_base:criticalsection",
-  ]
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 

api/DEPS

@@ -258,7 +258,6 @@ specific_include_rules = {
   ],
 
   "simulated_network\.h": [
-    "+rtc_base/critical_section.h",
     "+rtc_base/random.h",
     "+rtc_base/thread_annotations.h",
   ],

api/test/DEPS

@@ -13,7 +13,6 @@ specific_include_rules = {
   ],
   "loopback_media_transport\.h": [
     "+rtc_base/async_invoker.h",
-    "+rtc_base/critical_section.h",
     "+rtc_base/thread.h",
     "+rtc_base/thread_checker.h",
   ],

api/test/simulated_network.h

@@ -19,7 +19,6 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/random.h"
 #include "rtc_base/thread_annotations.h"
 

audio/audio_state.h

@@ -19,7 +19,6 @@
 #include "audio/null_audio_poller.h"
 #include "call/audio_state.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/thread_checker.h"
 

audio/voip/audio_channel.cc

@@ -17,7 +17,6 @@
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/rtp_rtcp/include/receive_statistics.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/location.h"
 #include "rtc_base/logging.h"
 

audio/voip/audio_channel.h

@@ -22,7 +22,6 @@
 #include "audio/voip/audio_ingress.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
 #include "modules/utility/include/process_thread.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/ref_count.h"
 
 namespace webrtc {

modules/audio_processing/echo_control_mobile_unittest.cc

@@ -13,7 +13,6 @@
 
 #include "modules/audio_processing/echo_control_mobile_impl.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/critical_section.h"
 #include "test/gtest.h"
 
 namespace webrtc {

modules/congestion_controller/rtp/transport_feedback_adapter.h

@@ -19,7 +19,6 @@
 #include "api/transport/network_types.h"
 #include "modules/include/module_common_types_public.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/network_route.h"
 #include "rtc_base/thread_annotations.h"

modules/desktop_capture/win/dxgi_output_duplicator.h

@@ -27,7 +27,6 @@
 #include "modules/desktop_capture/win/d3d_device.h"
 #include "modules/desktop_capture/win/dxgi_context.h"
 #include "modules/desktop_capture/win/dxgi_texture.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/thread_annotations.h"
 
 namespace webrtc {

modules/pacing/pacing_controller.h

@@ -31,7 +31,6 @@
 #include "modules/rtp_rtcp/include/rtp_packet_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/thread_annotations.h"
 

modules/pacing/packet_router.h

@@ -27,7 +27,6 @@
 #include "modules/rtp_rtcp/source/rtcp_packet.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
 

modules/pacing/task_queue_paced_sender.h

@@ -29,7 +29,6 @@
 #include "modules/pacing/packet_router.h"
 #include "modules/pacing/rtp_packet_pacer.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/synchronization/sequence_checker.h"
 #include "rtc_base/task_queue.h"

modules/video_coding/codecs/h264/h264_decoder_impl.cc

@@ -32,7 +32,6 @@ extern "C" {
 #include "common_video/include/video_frame_buffer.h"
 #include "modules/video_coding/codecs/h264/h264_color_space.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/keep_ref_until_done.h"
 #include "rtc_base/logging.h"
 #include "system_wrappers/include/field_trial.h"

modules/video_coding/frame_object.cc

@@ -17,7 +17,6 @@
 #include "api/video/encoded_image.h"
 #include "api/video/video_timing.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
 
 namespace webrtc {
 namespace video_coding {

modules/video_coding/receiver.h

@@ -20,7 +20,6 @@
 #include "modules/video_coding/jitter_buffer.h"
 #include "modules/video_coding/packet.h"
 #include "modules/video_coding/timing.h"
-#include "rtc_base/critical_section.h"
 
 namespace webrtc {
 

modules/video_coding/rtp_frame_reference_finder.h

@@ -21,7 +21,6 @@
 #include "modules/include/module_common_types_public.h"
 #include "modules/rtp_rtcp/source/rtp_video_header.h"
 #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/numerics/sequence_number_util.h"
 #include "rtc_base/thread_annotations.h"
 

modules/video_coding/video_coding_impl.cc

@@ -16,7 +16,6 @@
 #include "api/video/encoded_image.h"
 #include "modules/video_coding/include/video_codec_interface.h"
 #include "modules/video_coding/timing.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/thread_checker.h"
 #include "system_wrappers/include/clock.h"
 

pc/channel.h

@@ -38,7 +38,6 @@
 #include "pc/srtp_transport.h"
 #include "rtc_base/async_invoker.h"
 #include "rtc_base/async_udp_socket.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/network.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"
 #include "rtc_base/unique_id_generator.h"

pc/srtp_filter.h

@@ -24,7 +24,6 @@
 #include "pc/session_description.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/ssl_stream_adapter.h"
 #include "rtc_base/thread_checker.h"
 

pc/srtp_session_unittest.cc

@@ -17,7 +17,6 @@
 #include "media/base/fake_rtp.h"
 #include "pc/test/srtp_test_util.h"
 #include "rtc_base/byte_order.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/ssl_stream_adapter.h"  // For rtc::SRTP_*
 #include "system_wrappers/include/metrics.h"
 #include "test/gmock.h"

rtc_base/async_invoker_inl.h

@@ -13,7 +13,6 @@
 
 #include "api/scoped_refptr.h"
 #include "rtc_base/bind.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/ref_counted_object.h"

rtc_base/logging.cc

@@ -42,7 +42,6 @@ static const int kMaxLogLineSize = 1024 - 60;
 
 #include "absl/base/attributes.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/platform_thread_types.h"
 #include "rtc_base/string_encode.h"
 #include "rtc_base/string_utils.h"

rtc_base/stream.h

@@ -15,7 +15,6 @@
 
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/message_handler.h"
 #include "rtc_base/system/rtc_export.h"
 #include "rtc_base/third_party/sigslot/sigslot.h"

rtc_base/system/file_wrapper.h

@@ -14,7 +14,7 @@
 #include <stddef.h>
 #include <stdio.h>
 
-#include "rtc_base/critical_section.h"
+#include <string>
 
 // Implementation that can read (exclusive) or write from/to a file.
 

sdk/objc/native/src/audio/audio_device_ios.mm

@@ -21,7 +21,6 @@
 #include "rtc_base/atomic_ops.h"
 #include "rtc_base/bind.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"

sdk/objc/native/src/audio/audio_device_module_ios.h

@@ -19,7 +19,6 @@
 #include "modules/audio_device/audio_device_buffer.h"
 #include "modules/audio_device/include/audio_device.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/critical_section.h"
 
 namespace webrtc {
 

test/network/emulated_network_manager.h

@@ -16,7 +16,6 @@
 
 #include "api/test/network_emulation_manager.h"
 #include "api/test/time_controller.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/ip_address.h"
 #include "rtc_base/network.h"
 #include "rtc_base/socket_server.h"

test/pc/e2e/analyzer/video/default_encoded_image_data_injector.h

@@ -19,7 +19,6 @@
 #include <vector>
 
 #include "api/video/encoded_image.h"
-#include "rtc_base/critical_section.h"
 #include "test/pc/e2e/analyzer/video/encoded_image_data_injector.h"
 
 namespace webrtc {

test/rtp_rtcp_observer.h

@@ -18,7 +18,6 @@
 #include "api/test/simulated_network.h"
 #include "call/simulated_packet_receiver.h"
 #include "call/video_send_stream.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "system_wrappers/include/field_trial.h"
 #include "test/direct_transport.h"

test/testsupport/video_frame_writer.h

@@ -16,7 +16,6 @@
 
 #include "api/video/video_frame.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/critical_section.h"
 #include "test/testsupport/frame_writer.h"
 
 namespace webrtc {

video/adaptation/quality_scaler_resource.h

@@ -23,7 +23,6 @@
 #include "call/adaptation/degradation_preference_provider.h"
 #include "call/adaptation/resource_adaptation_processor_interface.h"
 #include "modules/video_coding/utility/quality_scaler.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/ref_counted_object.h"
 #include "rtc_base/task_queue.h"
 #include "video/adaptation/video_stream_encoder_resource.h"

video/call_stats2.h

@@ -18,7 +18,6 @@
 #include "modules/include/module_common_types.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/synchronization/sequence_checker.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/task_utils/pending_task_safety_flag.h"

video/video_receive_stream2_unittest.cc

@@ -25,7 +25,6 @@
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/utility/include/process_thread.h"
 #include "modules/video_coding/encoded_frame.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "system_wrappers/include/clock.h"
 #include "test/fake_decoder.h"

video/video_receive_stream_unittest.cc

@@ -25,7 +25,6 @@
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "modules/utility/include/process_thread.h"
 #include "modules/video_coding/encoded_frame.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "system_wrappers/include/clock.h"
 #include "test/fake_decoder.h"

video/video_send_stream.h

@@ -20,7 +20,6 @@
 #include "call/bitrate_allocator.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"

video/video_stream_decoder2.h

@@ -20,7 +20,6 @@
 #include "api/video/video_sink_interface.h"
 #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "modules/video_coding/include/video_coding_defines.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/platform_thread.h"
 
 namespace webrtc {

video/video_stream_encoder.h

@@ -34,7 +34,6 @@
 #include "call/adaptation/video_source_restrictions.h"
 #include "call/adaptation/video_stream_input_state_provider.h"
 #include "modules/video_coding/utility/frame_dropper.h"
-#include "rtc_base/critical_section.h"
 #include "rtc_base/event.h"
 #include "rtc_base/experiments/rate_control_settings.h"
 #include "rtc_base/numerics/exp_filter.h"