Format almost everything.

This CL was generated by running

git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \
grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \
grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \
grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \
grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \
| xargs clang-format -i ; git cl format

Most of these changes are clang-format grouping and reordering includes
differently.

Bug: webrtc:9340
Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28505}

api/array_view_unittest.cc

@@ -8,13 +8,14 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/array_view.h"
+
 #include <algorithm>
 #include <array>
 #include <string>
 #include <utility>
 #include <vector>
 
-#include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/gunit.h"

api/audio/echo_canceller3_config_json.cc

@@ -10,6 +10,7 @@
 #include "api/audio/echo_canceller3_config_json.h"
 
 #include <stddef.h>
+
 #include <string>
 #include <vector>
 

api/audio/test/audio_frame_unittest.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/audio/audio_frame.h"
+
 #include <stdint.h>
 #include <string.h>  // memcmp
 
-#include "api/audio/audio_frame.h"
 #include "test/gtest.h"
 
 namespace webrtc {

api/audio/test/echo_canceller3_config_json_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/audio/echo_canceller3_config_json.h"
+
 #include "api/audio/echo_canceller3_config.h"
 #include "test/gtest.h"
 

api/audio/test/echo_canceller3_config_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/audio/echo_canceller3_config.h"
+
 #include "api/audio/echo_canceller3_config_json.h"
 #include "test/gtest.h"
 

api/audio_codecs/audio_decoder.cc

@@ -11,6 +11,7 @@
 #include "api/audio_codecs/audio_decoder.h"
 
 #include <assert.h>
+
 #include <memory>
 #include <utility>
 

api/audio_codecs/audio_decoder.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <memory>
 #include <vector>
 

api/audio_codecs/audio_format.h

@@ -12,6 +12,7 @@
 #define API_AUDIO_CODECS_AUDIO_FORMAT_H_
 
 #include <stddef.h>
+
 #include <map>
 #include <string>
 

api/audio_codecs/g711/audio_encoder_g711.cc

@@ -79,7 +79,9 @@ std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
       impl_config.payload_type = payload_type;
       return absl::make_unique<AudioEncoderPcmA>(impl_config);
     }
-    default: { return nullptr; }
+    default: {
+      return nullptr;
+    }
   }
 }
 

api/audio_codecs/test/audio_decoder_factory_template_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/audio_codecs/audio_decoder_factory_template.h"
+
 #include "absl/memory/memory.h"
 #include "api/audio_codecs/L16/audio_decoder_L16.h"
 #include "api/audio_codecs/g711/audio_decoder_g711.h"

api/audio_codecs/test/audio_encoder_factory_template_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/audio_codecs/audio_encoder_factory_template.h"
+
 #include "absl/memory/memory.h"
 #include "api/audio_codecs/L16/audio_encoder_L16.h"
 #include "api/audio_codecs/g711/audio_encoder_g711.h"

api/audio_options.h

@@ -12,6 +12,7 @@
 #define API_AUDIO_OPTIONS_H_
 
 #include <stdint.h>
+
 #include <string>
 
 #include "absl/types/optional.h"

api/call/transport.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <vector>
 
 namespace webrtc {

api/crypto/crypto_options.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/crypto/crypto_options.h"
+
 #include "rtc_base/ssl_stream_adapter.h"
 
 namespace webrtc {

api/data_channel_interface.h

@@ -16,6 +16,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <string>
 
 #include "absl/types/optional.h"

api/function_view_unittest.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/function_view.h"
+
 #include <memory>
 #include <utility>
 
-#include "api/function_view.h"
 #include "test/gtest.h"
 
 namespace rtc {

api/jsep_ice_candidate.h

@@ -15,6 +15,7 @@
 #define API_JSEP_ICE_CANDIDATE_H_
 
 #include <stddef.h>
+
 #include <string>
 #include <vector>
 

api/media_transport_interface.cc

@@ -15,11 +15,12 @@
 // enable different media transport implementations, including QUIC-based
 // media transport.
 
+#include "api/media_transport_interface.h"
+
 #include <cstdint>
 #include <utility>
 
 #include "api/datagram_transport_interface.h"
-#include "api/media_transport_interface.h"
 
 namespace webrtc {
 
@@ -30,7 +31,6 @@ MediaTransportSettings& MediaTransportSettings::operator=(
     const MediaTransportSettings&) = default;
 MediaTransportSettings::~MediaTransportSettings() = default;
 
-
 SendDataParams::SendDataParams() = default;
 SendDataParams::SendDataParams(const SendDataParams&) = default;
 

api/peer_connection_interface.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/peer_connection_interface.h"
+
 #include "api/dtls_transport_interface.h"
 #include "api/sctp_transport_interface.h"
 

api/rtc_error_unittest.cc

@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/rtc_error.h"
+
 #include <utility>
 
-#include "api/rtc_error.h"
 #include "test/gtest.h"
 
 namespace {

api/rtc_event_log_output_file.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/rtc_event_log_output_file.h"
+
 #include <limits>
 #include <utility>
 
 #include "api/rtc_event_log/rtc_event_log.h"
-#include "api/rtc_event_log_output_file.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
 

api/rtc_event_log_output_file.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdio.h>
+
 #include <string>
 
 #include "api/rtc_event_log_output.h"

api/rtp_headers.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <string>
 
 #include "absl/types/optional.h"

api/rtp_packet_info_unittest.cc

@@ -9,7 +9,6 @@
  */
 
 #include "api/rtp_packet_infos.h"
-
 #include "test/gmock.h"
 #include "test/gtest.h"
 

api/rtp_parameters.h

@@ -12,6 +12,7 @@
 #define API_RTP_PARAMETERS_H_
 
 #include <stdint.h>
+
 #include <string>
 #include <unordered_map>
 #include <vector>

api/rtp_parameters_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/rtp_parameters.h"
+
 #include "test/gtest.h"
 
 namespace webrtc {

api/sctp_transport_interface.cc

@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include <utility>
-
 #include "api/sctp_transport_interface.h"
 
+#include <utility>
+
 namespace webrtc {
 
 SctpTransportInformation::SctpTransportInformation(SctpTransportState state)

api/stats/rtc_stats.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <memory>
 #include <string>
 #include <utility>

api/stats/rtc_stats_report.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <map>
 #include <memory>
 #include <string>

api/stats/rtcstats_objects.h

@@ -12,6 +12,7 @@
 #define API_STATS_RTCSTATS_OBJECTS_H_
 
 #include <stdint.h>
+
 #include <memory>
 #include <string>
 #include <vector>
@@ -376,7 +377,7 @@ class RTC_EXPORT RTCRTPStreamStats : public RTCStats {
   RTCStatsMember<std::string> associate_stats_id;
   // TODO(hbos): Remote case not supported by |RTCStatsCollector|.
   // crbug.com/657855, 657856
-  RTCStatsMember<bool> is_remote;  // = false
+  RTCStatsMember<bool> is_remote;          // = false
   RTCStatsMember<std::string> media_type;  // renamed to kind.
   RTCStatsMember<std::string> kind;
   RTCStatsMember<std::string> track_id;

api/test/create_video_quality_test_fixture.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/create_video_quality_test_fixture.h"
+
 #include <memory>
 #include <utility>
 
 #include "absl/memory/memory.h"
-#include "api/test/create_video_quality_test_fixture.h"
 #include "video/video_quality_test.h"
 
 namespace webrtc {
@@ -23,8 +24,7 @@ CreateVideoQualityTestFixture() {
   return absl::make_unique<VideoQualityTest>(nullptr);
 }
 
-std::unique_ptr<VideoQualityTestFixtureInterface>
-CreateVideoQualityTestFixture(
+std::unique_ptr<VideoQualityTestFixtureInterface> CreateVideoQualityTestFixture(
     std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory) {
   auto components = absl::make_unique<
       VideoQualityTestFixtureInterface::InjectionComponents>();
@@ -39,5 +39,3 @@ std::unique_ptr<VideoQualityTestFixtureInterface> CreateVideoQualityTestFixture(
 }
 
 }  // namespace webrtc
-
-

api/test/create_video_quality_test_fixture.h

@@ -20,8 +20,7 @@ namespace webrtc {
 std::unique_ptr<VideoQualityTestFixtureInterface>
 CreateVideoQualityTestFixture();
 
-std::unique_ptr<VideoQualityTestFixtureInterface>
-CreateVideoQualityTestFixture(
+std::unique_ptr<VideoQualityTestFixtureInterface> CreateVideoQualityTestFixture(
     std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory);
 
 std::unique_ptr<VideoQualityTestFixtureInterface> CreateVideoQualityTestFixture(

api/test/fake_frame_decryptor.cc

@@ -9,7 +9,9 @@
  */
 
 #include "api/test/fake_frame_decryptor.h"
+
 #include <vector>
+
 #include "rtc_base/checks.h"
 
 namespace webrtc {

api/test/fake_frame_decryptor.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <vector>
 
 #include "api/array_view.h"

api/test/fake_frame_encryptor.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/test/fake_frame_encryptor.h"
+
 #include "rtc_base/checks.h"
 
 namespace webrtc {

api/test/loopback_media_transport_unittest.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/loopback_media_transport.h"
+
 #include <algorithm>
 #include <memory>
 #include <vector>
 
-#include "api/test/loopback_media_transport.h"
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/mock_audio_mixer.h

@@ -12,7 +12,6 @@
 #define API_TEST_MOCK_AUDIO_MIXER_H_
 
 #include "api/audio/audio_mixer.h"
-
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/mock_frame_encryptor.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/test/mock_frame_encryptor.h"
+
 #include "test/gmock.h"
 
 namespace webrtc {

api/test/neteq_simulator.h

@@ -12,6 +12,7 @@
 #define API_TEST_NETEQ_SIMULATOR_H_
 
 #include <stdint.h>
+
 #include <map>
 #include <vector>
 

api/test/simulated_network.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <deque>
 #include <queue>
 #include <vector>

api/test/test_dependency_factory.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/test/test_dependency_factory.h"
+
 #include <memory>
 #include <utility>
 
 #include "absl/memory/memory.h"
-#include "api/test/test_dependency_factory.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/thread_checker.h"
 

api/test/videocodec_test_stats.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <string>
 #include <vector>
 

api/transport/field_trial_based_config.cc

@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "api/transport/field_trial_based_config.h"
+
 #include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {

api/transport/field_trial_based_config.h

@@ -11,6 +11,7 @@
 #define API_TRANSPORT_FIELD_TRIAL_BASED_CONFIG_H_
 
 #include <string>
+
 #include "absl/strings/string_view.h"
 #include "api/transport/webrtc_key_value_config.h"
 

api/transport/media/audio_transport.cc

@@ -15,10 +15,10 @@
 // enable different media transport implementations, including QUIC-based
 // media transport.
 
-#include <utility>
-
 #include "api/transport/media/audio_transport.h"
 
+#include <utility>
+
 namespace webrtc {
 
 MediaTransportEncodedAudioFrame::~MediaTransportEncodedAudioFrame() {}

api/transport/media/video_transport.cc

@@ -15,10 +15,10 @@
 // enable different media transport implementations, including QUIC-based
 // media transport.
 
-#include <utility>
-
 #include "api/transport/media/video_transport.h"
 
+#include <utility>
+
 namespace webrtc {
 
 MediaTransportEncodedVideoFrame::MediaTransportEncodedVideoFrame() = default;

api/transport/network_control.h

@@ -11,6 +11,7 @@
 #ifndef API_TRANSPORT_NETWORK_CONTROL_H_
 #define API_TRANSPORT_NETWORK_CONTROL_H_
 #include <stdint.h>
+
 #include <memory>
 
 #include "api/transport/network_types.h"

api/transport/network_types.h

@@ -11,6 +11,7 @@
 #ifndef API_TRANSPORT_NETWORK_TYPES_H_
 #define API_TRANSPORT_NETWORK_TYPES_H_
 #include <stdint.h>
+
 #include <vector>
 
 #include "absl/types/optional.h"

api/transport/test/create_feedback_generator.cc

@@ -10,7 +10,6 @@
 #include "api/transport/test/create_feedback_generator.h"
 
 #include "absl/memory/memory.h"
-
 #include "test/network/feedback_generator.h"
 
 namespace webrtc {

api/transport/webrtc_key_value_config.h

@@ -11,6 +11,7 @@
 #define API_TRANSPORT_WEBRTC_KEY_VALUE_CONFIG_H_
 
 #include <string>
+
 #include "absl/strings/string_view.h"
 
 namespace webrtc {

api/units/data_rate_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/units/data_rate.h"
+
 #include "rtc_base/logging.h"
 #include "test/gtest.h"
 

api/units/frequency.cc

@@ -8,6 +8,7 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include "api/units/frequency.h"
+
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {

api/video/encoded_frame.h

@@ -11,11 +11,11 @@
 #ifndef API_VIDEO_ENCODED_FRAME_H_
 #define API_VIDEO_ENCODED_FRAME_H_
 
-#include "modules/video_coding/encoded_frame.h"
-
 #include <stddef.h>
 #include <stdint.h>
 
+#include "modules/video_coding/encoded_frame.h"
+
 namespace webrtc {
 namespace video_coding {
 

api/video/encoded_image.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_ENCODED_IMAGE_H_
 
 #include <stdint.h>
+
 #include <map>
 #include <utility>
 
@@ -95,9 +96,7 @@ class RTC_EXPORT EncodedImage {
 
   void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
 
-  absl::optional<int> SpatialIndex() const {
-    return spatial_index_;
-  }
+  absl::optional<int> SpatialIndex() const { return spatial_index_; }
   void SetSpatialIndex(absl::optional<int> spatial_index) {
     RTC_DCHECK_GE(spatial_index.value_or(0), 0);
     RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
@@ -208,7 +207,7 @@ class RTC_EXPORT EncodedImage {
   // TODO(bugs.webrtc.org/9378): We're transitioning to always owning the
   // encoded data.
   rtc::scoped_refptr<EncodedImageBufferInterface> encoded_data_;
-  size_t size_;      // Size of encoded frame data.
+  size_t size_;  // Size of encoded frame data.
   // Non-null when used with an un-owned buffer.
   uint8_t* buffer_;
   // Allocated size of _buffer; relevant only if it's non-null.

api/video/i010_buffer.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_I010_BUFFER_H_
 
 #include <stdint.h>
+
 #include <memory>
 
 #include "api/scoped_refptr.h"

api/video/i420_buffer.cc

@@ -10,6 +10,7 @@
 #include "api/video/i420_buffer.h"
 
 #include <string.h>
+
 #include <algorithm>
 #include <utility>
 

api/video/i420_buffer.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_I420_BUFFER_H_
 
 #include <stdint.h>
+
 #include <memory>
 
 #include "api/scoped_refptr.h"

api/video/test/color_space_unittest.cc

@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/video/color_space.h"
+
 #include <stdint.h>
 
-#include "api/video/color_space.h"
 #include "test/gtest.h"
 
 namespace webrtc {

api/video/test/video_bitrate_allocation_unittest.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/video/video_bitrate_allocation.h"
+
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "api/video/video_bitrate_allocation.h"
 #include "test/gtest.h"
 
 namespace webrtc {

api/video/video_bitrate_allocation.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <limits>
 #include <string>
 #include <vector>

api/video/video_bitrate_allocator_factory.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
 
 #include <memory>
+
 #include "api/video/video_bitrate_allocator.h"
 #include "api/video_codecs/video_codec.h"
 

api/video/video_frame.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_VIDEO_FRAME_H_
 
 #include <stdint.h>
+
 #include <utility>
 
 #include "absl/types/optional.h"

api/video/video_stream_encoder_create.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_VIDEO_STREAM_ENCODER_CREATE_H_
 
 #include <stdint.h>
+
 #include <memory>
 
 #include "api/task_queue/task_queue_factory.h"

api/video_codecs/sdp_video_format.cc

@@ -9,6 +9,7 @@
  */
 
 #include "api/video_codecs/sdp_video_format.h"
+
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {

api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc

@@ -8,8 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
+
 #include <stddef.h>
 #include <stdint.h>
+
 #include <memory>
 #include <string>
 #include <vector>
@@ -27,7 +30,6 @@
 #include "api/video/video_rotation.h"
 #include "api/video_codecs/video_codec.h"
 #include "api/video_codecs/video_encoder.h"
-#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
 #include "modules/include/module_common_types.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "modules/video_coding/include/video_codec_interface.h"

api/video_codecs/video_codec.cc

@@ -11,6 +11,7 @@
 #include "api/video_codecs/video_codec.h"
 
 #include <string.h>
+
 #include <string>
 
 #include "absl/strings/match.h"

api/video_codecs/video_codec.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <string>
 
 #include "api/video/video_bitrate_allocation.h"
@@ -107,9 +108,9 @@ class RTC_EXPORT VideoCodec {
   uint16_t width;
   uint16_t height;
 
-  unsigned int startBitrate;   // kilobits/sec.
-  unsigned int maxBitrate;     // kilobits/sec.
-  unsigned int minBitrate;     // kilobits/sec.
+  unsigned int startBitrate;  // kilobits/sec.
+  unsigned int maxBitrate;    // kilobits/sec.
+  unsigned int minBitrate;    // kilobits/sec.
 
   uint32_t maxFramerate;
 

api/video_codecs/video_decoder_software_fallback_wrapper.cc

@@ -11,6 +11,7 @@
 #include "api/video_codecs/video_decoder_software_fallback_wrapper.h"
 
 #include <stdint.h>
+
 #include <string>
 #include <utility>
 

api/video_codecs/video_encoder_config.h

@@ -12,6 +12,7 @@
 #define API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
 
 #include <stddef.h>
+
 #include <string>
 #include <vector>
 

api/video_codecs/video_encoder_software_fallback_wrapper.cc

@@ -11,6 +11,7 @@
 #include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
 
 #include <stdint.h>
+
 #include <cstdio>
 #include <string>
 #include <utility>

audio/audio_level.cc

@@ -17,8 +17,7 @@ namespace webrtc {
 namespace voe {
 
 AudioLevel::AudioLevel()
-    : abs_max_(0), count_(0), current_level_full_range_(0) {
-}
+    : abs_max_(0), count_(0), current_level_full_range_(0) {}
 
 AudioLevel::~AudioLevel() {}
 

audio/audio_receive_stream_unittest.cc

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/audio_receive_stream.h"
+
 #include <map>
 #include <string>
 #include <utility>
@@ -15,7 +17,6 @@
 
 #include "api/test/mock_audio_mixer.h"
 #include "api/test/mock_frame_decryptor.h"
-#include "audio/audio_receive_stream.h"
 #include "audio/conversion.h"
 #include "audio/mock_voe_channel_proxy.h"
 #include "call/rtp_stream_receiver_controller.h"

audio/audio_send_stream_unittest.cc

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/audio_send_stream.h"
+
 #include <string>
 #include <utility>
 #include <vector>
@@ -15,7 +17,6 @@
 #include "absl/memory/memory.h"
 #include "api/task_queue/default_task_queue_factory.h"
 #include "api/test/mock_frame_encryptor.h"
-#include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
 #include "audio/conversion.h"
 #include "audio/mock_voe_channel_proxy.h"

audio/audio_state_unittest.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/audio_state.h"
+
 #include <memory>
 #include <vector>
 
-#include "audio/audio_state.h"
 #include "call/test/mock_audio_send_stream.h"
 #include "modules/audio_device/include/mock_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"

audio/channel_receive.cc

@@ -549,8 +549,8 @@ void ChannelReceive::StopPlayout() {
   _outputAudioLevel.ResetLevelFullRange();
 }
 
-absl::optional<std::pair<int, SdpAudioFormat>>
-    ChannelReceive::GetReceiveCodec() const {
+absl::optional<std::pair<int, SdpAudioFormat>> ChannelReceive::GetReceiveCodec()
+    const {
   RTC_DCHECK(worker_thread_checker_.IsCurrent());
   return audio_coding_->ReceiveCodec();
 }

audio/channel_receive.h

@@ -85,8 +85,8 @@ class ChannelReceiveInterface : public RtpPacketSinkInterface {
   virtual void StopPlayout() = 0;
 
   // Payload type and format of last received RTP packet, if any.
-  virtual absl::optional<std::pair<int, SdpAudioFormat>>
-      GetReceiveCodec() const = 0;
+  virtual absl::optional<std::pair<int, SdpAudioFormat>> GetReceiveCodec()
+      const = 0;
 
   virtual void ReceivedRTCPPacket(const uint8_t* data, size_t length) = 0;
 

audio/remix_resample_unittest.cc

@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/remix_resample.h"
+
 #include <cmath>
 
-#include "audio/remix_resample.h"
 #include "common_audio/resampler/include/push_resampler.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/checks.h"

audio/test/audio_end_to_end_test.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/test/audio_end_to_end_test.h"
+
 #include <algorithm>
 
 #include "absl/memory/memory.h"
-#include "audio/test/audio_end_to_end_test.h"
 #include "call/fake_network_pipe.h"
 #include "call/simulated_network.h"
 #include "system_wrappers/include/sleep.h"

audio/transport_feedback_packet_loss_tracker.cc

@@ -225,8 +225,7 @@ void TransportFeedbackPacketLossTracker::UpdateRplr(
       UpdateCounter(&rplr_state_.num_acked_pairs_, apply);
       if (prev->second.status == PacketStatus::Lost &&
           it->second.status == PacketStatus::Received) {
-        UpdateCounter(
-            &rplr_state_.num_recoverable_losses_, apply);
+        UpdateCounter(&rplr_state_.num_recoverable_losses_, apply);
       }
     }
   }

audio/transport_feedback_packet_loss_tracker_unittest.cc

@@ -8,12 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/transport_feedback_packet_loss_tracker.h"
+
 #include <limits>
 #include <memory>
 #include <numeric>
 #include <vector>
 
-#include "audio/transport_feedback_packet_loss_tracker.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "rtc_base/checks.h"

audio/utility/audio_frame_operations.cc

@@ -11,6 +11,7 @@
 #include "audio/utility/audio_frame_operations.h"
 
 #include <string.h>
+
 #include <algorithm>
 #include <cstdint>
 #include <utility>

audio/utility/audio_frame_operations_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include "audio/utility/audio_frame_operations.h"
+
 #include "rtc_base/checks.h"
 #include "test/gtest.h"
 

audio/utility/channel_mixer.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <memory>
 #include <vector>
 

audio/utility/channel_mixer_unittest.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "audio/utility/channel_mixer.h"
+
 #include <memory>
 
 #include "api/audio/audio_frame.h"
 #include "api/audio/channel_layout.h"
-#include "audio/utility/channel_mixer.h"
 #include "audio/utility/channel_mixing_matrix.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/strings/string_builder.h"

call/bitrate_allocator_unittest.cc

@@ -8,11 +8,12 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "call/bitrate_allocator.h"
+
 #include <algorithm>
 #include <memory>
 #include <vector>
 
-#include "call/bitrate_allocator.h"
 #include "system_wrappers/include/clock.h"
 #include "test/gmock.h"
 #include "test/gtest.h"

call/call.cc

@@ -8,7 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "call/call.h"
+
 #include <string.h>
+
 #include <algorithm>
 #include <map>
 #include <memory>
@@ -23,7 +26,6 @@
 #include "audio/audio_send_stream.h"
 #include "audio/audio_state.h"
 #include "call/bitrate_allocator.h"
-#include "call/call.h"
 #include "call/flexfec_receive_stream_impl.h"
 #include "call/receive_time_calculator.h"
 #include "call/rtp_stream_receiver_controller.h"

call/call_factory.cc

@@ -11,6 +11,7 @@
 #include "call/call_factory.h"
 
 #include <stdio.h>
+
 #include <memory>
 #include <string>
 

call/call_unittest.cc

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "call/call.h"
+
 #include <list>
 #include <map>
 #include <memory>
@@ -21,7 +23,6 @@
 #include "audio/audio_receive_stream.h"
 #include "audio/audio_send_stream.h"
 #include "call/audio_state.h"
-#include "call/call.h"
 #include "logging/rtc_event_log/rtc_event_log.h"
 #include "modules/audio_device/include/mock_audio_device.h"
 #include "modules/audio_processing/include/mock_audio_processing.h"

call/degraded_call.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "call/degraded_call.h"
+
 #include <utility>
 
 #include "absl/memory/memory.h"
-#include "call/degraded_call.h"
 #include "rtc_base/location.h"
 
 namespace webrtc {

call/degraded_call.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <memory>
 
 #include "absl/types/optional.h"

call/fake_network_pipe.cc

@@ -8,14 +8,16 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "call/fake_network_pipe.h"
+
 #include <string.h>
+
 #include <algorithm>
 #include <queue>
 #include <utility>
 #include <vector>
 
 #include "api/media_types.h"
-#include "call/fake_network_pipe.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"

call/flexfec_receive_stream.h

@@ -12,6 +12,7 @@
 #define CALL_FLEXFEC_RECEIVE_STREAM_H_
 
 #include <stdint.h>
+
 #include <string>
 #include <vector>
 

call/flexfec_receive_stream_impl.cc

@@ -11,6 +11,7 @@
 #include "call/flexfec_receive_stream_impl.h"
 
 #include <stddef.h>
+
 #include <cstdint>
 #include <string>
 #include <vector>

call/flexfec_receive_stream_unittest.cc

@@ -8,6 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "call/flexfec_receive_stream.h"
+
 #include <cstdint>
 #include <memory>
 #include <vector>
@@ -17,7 +19,6 @@
 #include "api/call/transport.h"
 #include "api/rtp_headers.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"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"

call/receive_time_calculator.h

@@ -11,6 +11,7 @@
 #define CALL_RECEIVE_TIME_CALCULATOR_H_
 
 #include <stdint.h>
+
 #include <memory>
 
 #include "api/units/time_delta.h"

call/receive_time_calculator_unittest.cc

@@ -11,6 +11,7 @@
 #include "call/receive_time_calculator.h"
 
 #include <stdlib.h>
+
 #include <algorithm>
 #include <cmath>
 #include <cstdint>

call/rtcp_demuxer.cc

@@ -11,6 +11,7 @@
 #include "call/rtcp_demuxer.h"
 
 #include <stddef.h>
+
 #include <algorithm>
 #include <utility>
 

call/rtcp_demuxer_unittest.cc

@@ -11,6 +11,7 @@
 #include "call/rtcp_demuxer.h"
 
 #include <stddef.h>
+
 #include <set>
 
 #include "api/rtp_headers.h"

call/rtp_bitrate_configurator_unittest.cc

@@ -7,9 +7,10 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+#include "call/rtp_bitrate_configurator.h"
+
 #include <memory>
 
-#include "call/rtp_bitrate_configurator.h"
 #include "test/gtest.h"
 
 namespace webrtc {

call/rtp_config.h

@@ -13,6 +13,7 @@
 
 #include <stddef.h>
 #include <stdint.h>
+
 #include <string>
 #include <vector>