Make header files self contained.

This CL adds #includes to header files in order to make them self contained after the preprocessor pass. Bug: b/251890128 Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38327}
2025-05-12 21:30:45 +01:00 · 2022-10-07 21:47:49 +00:00 · 2022-10-07 21:47:49 +00:00 · 9d9c2d5795
commit 9d9c2d5795
parent 61ad0044df
45 changed files with 85 additions and 7 deletions
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@ -158,6 +158,7 @@ rtc_library("candidate") {
 rtc_source_set("turn_customizer") {
  visibility = [ "*" ]
  sources = [ "turn_customizer.h" ]
+  deps = [ "transport:stun_types" ]
 }

 rtc_source_set("ice_transport_interface") {
--- a/api/DEPS
+++ b/api/DEPS
@ -307,6 +307,10 @@ specific_include_rules = {
    "+rtc_base/containers/flat_map.h",
  ],

+  "video_track_source_proxy_factory.h": [
+    "+rtc_base/thread.h",
+  ],
+
  # .cc files in api/ should not be restricted in what they can #include,
  # so we re-add all the top-level directories here. (That's because .h
  # files leak their #includes to whoever's #including them, but .cc files
--- a/api/audio_codecs/g722/BUILD.gn
+++ b/api/audio_codecs/g722/BUILD.gn
@ -15,6 +15,7 @@ if (is_android) {
 rtc_source_set("audio_encoder_g722_config") {
  visibility = [ "*" ]
  sources = [ "audio_encoder_g722_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
 }

 rtc_library("audio_encoder_g722") {
--- a/api/audio_codecs/g722/audio_encoder_g722_config.h
+++ b/api/audio_codecs/g722/audio_encoder_g722_config.h
@ -11,6 +11,8 @@
 #ifndef API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_
 #define API_AUDIO_CODECS_G722_AUDIO_ENCODER_G722_CONFIG_H_

+#include "api/audio_codecs/audio_encoder.h"
+
 namespace webrtc {

 struct AudioEncoderG722Config {
--- a/api/audio_codecs/opus/BUILD.gn
+++ b/api/audio_codecs/opus/BUILD.gn
@ -20,9 +20,7 @@ rtc_library("audio_encoder_opus_config") {
    "audio_encoder_opus_config.cc",
    "audio_encoder_opus_config.h",
  ]
-  deps = [
-    "../../../rtc_base/system:rtc_export",
-  ]
+  deps = [ "../../../rtc_base/system:rtc_export" ]
  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
  defines = []
  if (rtc_opus_variable_complexity) {
@ -35,6 +33,7 @@ rtc_library("audio_encoder_opus_config") {
 rtc_source_set("audio_decoder_opus_config") {
  visibility = [ "*" ]
  sources = [ "audio_decoder_multi_channel_opus_config.h" ]
+  deps = [ "..:audio_codecs_api" ]
 }

 rtc_library("audio_encoder_opus") {
--- a/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h
+++ b/api/audio_codecs/opus/audio_decoder_multi_channel_opus_config.h
@ -13,6 +13,8 @@

 #include <vector>

+#include "api/audio_codecs/audio_decoder.h"
+
 namespace webrtc {
 struct AudioDecoderMultiChannelOpusConfig {
  // The number of channels that the decoder will output.
--- a/api/call/audio_sink.h
+++ b/api/call/audio_sink.h
@ -11,6 +11,7 @@
 #ifndef API_CALL_AUDIO_SINK_H_
 #define API_CALL_AUDIO_SINK_H_

+#include <stddef.h>
 #include <stdint.h>

 namespace webrtc {
--- a/api/test/mock_video_decoder_factory.h
+++ b/api/test/mock_video_decoder_factory.h
@ -15,6 +15,7 @@
 #include <vector>

 #include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "test/gmock.h"

--- a/api/test/mock_video_encoder_factory.h
+++ b/api/test/mock_video_encoder_factory.h
@ -15,6 +15,7 @@
 #include <vector>

 #include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "test/gmock.h"

--- a/api/test/video/function_video_decoder_factory.h
+++ b/api/test/video/function_video_decoder_factory.h
@ -17,6 +17,7 @@
 #include <vector>

 #include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
 #include "api/video_codecs/video_decoder_factory.h"
 #include "rtc_base/checks.h"

--- a/api/test/video/function_video_encoder_factory.h
+++ b/api/test/video/function_video_encoder_factory.h
@ -17,6 +17,7 @@
 #include <vector>

 #include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/video_encoder_factory.h"
 #include "rtc_base/checks.h"

--- a/api/turn_customizer.h
+++ b/api/turn_customizer.h
@ -13,9 +13,10 @@

 #include <stdlib.h>

+#include "api/transport/stun.h"
+
 namespace cricket {
 class PortInterface;
-class StunMessage;
 }  // namespace cricket

 namespace webrtc {
--- a/api/video_codecs/BUILD.gn
+++ b/api/video_codecs/BUILD.gn
@ -100,6 +100,7 @@ rtc_source_set("bitstream_parser_api") {
  visibility = [ "*" ]
  sources = [ "bitstream_parser.h" ]
  deps = [ "..:array_view" ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }

 rtc_library("builtin_video_decoder_factory") {
@ -164,6 +165,7 @@ rtc_source_set("video_encoder_factory_template_libvpx_vp8_adapter") {
  public = [ "video_encoder_factory_template_libvpx_vp8_adapter.h" ]

  deps = [
+    ":video_codecs_api",
    "../../modules/video_coding:webrtc_vp8",
    "../../modules/video_coding:webrtc_vp8_scalability",
  ]
@ -194,6 +196,7 @@ rtc_source_set("video_encoder_factory_template_libaom_av1_adapter") {

  deps = [
    ":scalability_mode",
+    ":video_codecs_api",
    "../../modules/video_coding/codecs/av1:av1_svc_config",
    "../../modules/video_coding/codecs/av1:libaom_av1_encoder",
    "../../modules/video_coding/svc:scalability_mode_util",
@ -219,7 +222,10 @@ rtc_source_set("video_decoder_factory_template_libvpx_vp8_adapter") {
  allow_poison = [ "software_video_codecs" ]
  public = [ "video_decoder_factory_template_libvpx_vp8_adapter.h" ]

-  deps = [ "../../modules/video_coding:webrtc_vp8" ]
+  deps = [
+    ":video_codecs_api",
+    "../../modules/video_coding:webrtc_vp8",
+  ]
 }

 rtc_source_set("video_decoder_factory_template_libvpx_vp9_adapter") {
@ -243,7 +249,10 @@ rtc_source_set("video_decoder_factory_template_dav1d_adapter") {
  allow_poison = [ "software_video_codecs" ]
  public = [ "video_decoder_factory_template_dav1d_adapter.h" ]

-  deps = [ "../../modules/video_coding/codecs/av1:dav1d_decoder" ]
+  deps = [
+    ":video_codecs_api",
+    "../../modules/video_coding/codecs/av1:dav1d_decoder",
+  ]
 }

 rtc_library("vp8_temporal_layers_factory") {
--- a/api/video_codecs/bitstream_parser.h
+++ b/api/video_codecs/bitstream_parser.h
@ -10,9 +10,11 @@

 #ifndef API_VIDEO_CODECS_BITSTREAM_PARSER_H_
 #define API_VIDEO_CODECS_BITSTREAM_PARSER_H_
+
 #include <stddef.h>
 #include <stdint.h>

+#include "absl/types/optional.h"
 #include "api/array_view.h"

 namespace webrtc {
--- a/api/video_codecs/video_decoder_factory_template_dav1d_adapter.h
+++ b/api/video_codecs/video_decoder_factory_template_dav1d_adapter.h
@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>

+#include "api/video_codecs/sdp_video_format.h"
 #include "modules/video_coding/codecs/av1/dav1d_decoder.h"

 namespace webrtc {
--- a/api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h
+++ b/api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h
@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>

+#include "api/video_codecs/sdp_video_format.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"

 namespace webrtc {
--- a/api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h
+++ b/api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h
@ -15,6 +15,7 @@
 #include <vector>

 #include "absl/container/inlined_vector.h"
+#include "api/video_codecs/sdp_video_format.h"
 #include "modules/video_coding/codecs/av1/av1_svc_config.h"
 #include "modules/video_coding/codecs/av1/libaom_av1_encoder.h"

--- a/api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h
+++ b/api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h
@ -15,6 +15,7 @@
 #include <vector>

 #include "absl/container/inlined_vector.h"
+#include "api/video_codecs/sdp_video_format.h"
 #include "modules/video_coding/codecs/vp8/include/vp8.h"
 #include "modules/video_coding/codecs/vp8/vp8_scalability.h"

--- a/api/video_track_source_proxy_factory.h
+++ b/api/video_track_source_proxy_factory.h
@ -12,6 +12,7 @@
 #define API_VIDEO_TRACK_SOURCE_PROXY_FACTORY_H_

 #include "api/media_stream_interface.h"
+#include "rtc_base/thread.h"

 namespace webrtc {

--- a/audio/conversion.h
+++ b/audio/conversion.h
@ -11,6 +11,9 @@
 #ifndef AUDIO_CONVERSION_H_
 #define AUDIO_CONVERSION_H_

+#include <stddef.h>
+#include <stdint.h>
+
 namespace webrtc {

 // Convert fixed point number with 8 bit fractional part, to floating point.
--- a/common_audio/signal_processing/include/spl_inl.h
+++ b/common_audio/signal_processing/include/spl_inl.h
@ -14,6 +14,8 @@
 #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_
 #define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_H_

+#include <stdint.h>
+
 #include "rtc_base/compile_assert_c.h"

 extern const int8_t kWebRtcSpl_CountLeadingZeros32_Table[64];
--- a/common_audio/signal_processing/include/spl_inl_armv7.h
+++ b/common_audio/signal_processing/include/spl_inl_armv7.h
@ -15,6 +15,8 @@
 #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_
 #define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_ARMV7_H_

+#include <stdint.h>
+
 /* TODO(kma): Replace some assembly code with GCC intrinsics
 * (e.g. __builtin_clz).
 */
--- a/common_video/frame_counts.h
+++ b/common_video/frame_counts.h
@ -11,6 +11,8 @@
 #ifndef COMMON_VIDEO_FRAME_COUNTS_H_
 #define COMMON_VIDEO_FRAME_COUNTS_H_

+#include <cstdint>
+
 namespace webrtc {

 struct FrameCounts {
--- a/modules/audio_coding/codecs/g711/g711_interface.h
+++ b/modules/audio_coding/codecs/g711/g711_interface.h
@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
 #define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_

+#include <stddef.h>
 #include <stdint.h>

 // Comfort noise constants
--- a/modules/audio_coding/codecs/g722/g722_interface.h
+++ b/modules/audio_coding/codecs/g722/g722_interface.h
@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
 #define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_

+#include <stddef.h>
 #include <stdint.h>

 /*
--- a/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_decoder_isac_t_impl.h
@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_
 #define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_DECODER_ISAC_T_IMPL_H_

+#include "modules/audio_coding/codecs/isac/audio_decoder_isac_t.h"
 #include "rtc_base/checks.h"

 namespace webrtc {
--- a/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_
 #define MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_

+#include "modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"

--- a/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
+++ b/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
@ -20,6 +20,8 @@

 #include <stdint.h>

+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
+
 /* indices of KLT coefficients used */
 extern const uint16_t WebRtcIsacfix_kSelIndGain[12];

--- a/modules/audio_coding/codecs/isac/main/source/filter_functions.h
+++ b/modules/audio_coding/codecs/isac/main/source/filter_functions.h
@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
 #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_

+#include <stddef.h>
+
 #include "modules/audio_coding/codecs/isac/main/source/structs.h"

 void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
--- a/modules/audio_processing/aec3/delay_estimate.h
+++ b/modules/audio_processing/aec3/delay_estimate.h
@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_
 #define MODULES_AUDIO_PROCESSING_AEC3_DELAY_ESTIMATE_H_

+#include <stddef.h>
+
 namespace webrtc {

 // Stores delay_estimates.
--- a/modules/audio_processing/agc/legacy/gain_control.h
+++ b/modules/audio_processing/agc/legacy/gain_control.h
@ -11,6 +11,9 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
 #define MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_

+#include <stddef.h>
+#include <stdint.h>
+
 namespace webrtc {

 enum {
--- a/modules/audio_processing/vad/vad_audio_proc_internal.h
+++ b/modules/audio_processing/vad/vad_audio_proc_internal.h
@ -11,6 +11,8 @@
 #ifndef MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_
 #define MODULES_AUDIO_PROCESSING_VAD_VAD_AUDIO_PROC_INTERNAL_H_

+#include <stddef.h>
+
 namespace webrtc {

 // These values should match MATLAB counterparts for unit-tests to pass.
--- a/modules/third_party/g722/g722_enc_dec.h
+++ b/modules/third_party/g722/g722_enc_dec.h
@ -30,6 +30,7 @@
 #ifndef MODULES_THIRD_PARTY_G722_G722_H_
 #define MODULES_THIRD_PARTY_G722_G722_H_

+#include <stddef.h>
 #include <stdint.h>

 /*! \page g722_page G.722 encoding and decoding
--- a/modules/video_coding/utility/ivf_defines.h
+++ b/modules/video_coding/utility/ivf_defines.h
@ -16,6 +16,8 @@
 #ifndef MODULES_VIDEO_CODING_UTILITY_IVF_DEFINES_H_
 #define MODULES_VIDEO_CODING_UTILITY_IVF_DEFINES_H_

+#include <stddef.h>
+
 namespace webrtc {
 constexpr size_t kIvfHeaderSize = 32;
 }  // namespace webrtc
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@ -2631,6 +2631,7 @@ if (rtc_include_tests && !build_with_chromium) {
      ":stream_collection",
      ":video_track_source",
      "../api:audio_options_api",
+      "../api:call_api",
      "../api:create_frame_generator",
      "../api:create_peerconnection_factory",
      "../api:field_trials_view",
--- a/pc/test/mock_voice_media_channel.h
+++ b/pc/test/mock_voice_media_channel.h
@ -14,6 +14,7 @@
 #include <string>
 #include <vector>

+#include "api/call/audio_sink.h"
 #include "media/base/media_channel.h"
 #include "rtc_base/gunit.h"
 #include "test/gmock.h"
--- a/pc/test/rtc_stats_obtainer.h
+++ b/pc/test/rtc_stats_obtainer.h
@ -11,6 +11,9 @@
 #ifndef PC_TEST_RTC_STATS_OBTAINER_H_
 #define PC_TEST_RTC_STATS_OBTAINER_H_

+#include "api/make_ref_counted.h"
+#include "api/sequence_checker.h"
+#include "api/stats/rtc_stats_collector_callback.h"
 #include "api/stats/rtc_stats_report.h"
 #include "rtc_base/gunit.h"

--- a/rtc_base/ssl_roots.h
+++ b/rtc_base/ssl_roots.h
@ -11,6 +11,8 @@
 #ifndef RTC_BASE_SSL_ROOTS_H_
 #define RTC_BASE_SSL_ROOTS_H_

+#include <cstddef>
+
 // This file is the root certificates in C form that are needed to connect to
 // Google.

--- a/rtc_base/system_time.h
+++ b/rtc_base/system_time.h
@ -11,6 +11,8 @@
 #ifndef RTC_BASE_SYSTEM_TIME_H_
 #define RTC_BASE_SYSTEM_TIME_H_

+#include <cstdint>
+
 namespace rtc {

 // Returns the actual system time, even if a clock is set for testing.
--- a/sdk/android/native_api/jni/jni_int_wrapper.h
+++ b/sdk/android/native_api/jni/jni_int_wrapper.h
@ -14,6 +14,8 @@
 #ifndef SDK_ANDROID_NATIVE_API_JNI_JNI_INT_WRAPPER_H_
 #define SDK_ANDROID_NATIVE_API_JNI_JNI_INT_WRAPPER_H_

+#include <cstdint>
+
 // Wrapper used to receive int when calling Java from native. The wrapper
 // disallows automatic conversion of anything besides int32_t to a jint.
 // Checking is only done in debugging builds.
--- a/sdk/objc/native/src/audio/helpers.h
+++ b/sdk/objc/native/src/audio/helpers.h
@ -11,6 +11,9 @@
 #ifndef SDK_OBJC_NATIVE_SRC_AUDIO_HELPERS_H_
 #define SDK_OBJC_NATIVE_SRC_AUDIO_HELPERS_H_

+#import <Foundation/Foundation.h>
+#include <objc/objc.h>
+
 #include <string>

 namespace webrtc {
--- a/test/mock_audio_encoder_factory.h
+++ b/test/mock_audio_encoder_factory.h
@ -15,6 +15,7 @@
 #include <vector>

 #include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/make_ref_counted.h"
 #include "api/scoped_refptr.h"
 #include "test/gmock.h"

--- a/test/pc/sctp/BUILD.gn
+++ b/test/pc/sctp/BUILD.gn
@ -11,5 +11,8 @@ import("../../../webrtc.gni")
 rtc_source_set("fake_sctp_transport") {
  visibility = [ "*" ]
  sources = [ "fake_sctp_transport.h" ]
-  deps = [ "../../../media:rtc_data_sctp_transport_internal" ]
+  deps = [
+    "../../../api/transport:sctp_transport_factory_interface",
+    "../../../media:rtc_data_sctp_transport_internal",
+  ]
 }
--- a/test/pc/sctp/fake_sctp_transport.h
+++ b/test/pc/sctp/fake_sctp_transport.h
@ -13,6 +13,7 @@

 #include <memory>

+#include "api/transport/sctp_transport_factory_interface.h"
 #include "media/sctp/sctp_transport_internal.h"

 // Used for tests in this file to verify that PeerConnection responds to signals
--- a/test/testsupport/mock/mock_frame_reader.h
+++ b/test/testsupport/mock/mock_frame_reader.h
@ -11,6 +11,7 @@
 #ifndef TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
 #define TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_

+#include "api/video/i420_buffer.h"
 #include "test/gmock.h"
 #include "test/testsupport/frame_reader.h"