diff --git a/BUILD.gn b/BUILD.gn index 188a70e598..f1ac0bb32b 100644 --- a/BUILD.gn +++ b/BUILD.gn @@ -447,14 +447,6 @@ rtc_source_set("webrtc_common") { sources = [ "common_types.h", ] - deps = [ - "api:array_view", - "api/video:video_frame", - "modules/audio_coding:audio_coding_module_typedefs", - "rtc_base:checks", - "rtc_base:deprecation", - "//third_party/abseil-cpp/absl/strings", - ] } if (use_libfuzzer || use_afl) { diff --git a/DEPS b/DEPS index 9abf5ea657..e428d18116 100644 --- a/DEPS +++ b/DEPS @@ -1632,11 +1632,3 @@ include_rules = [ "+absl/types/optional.h", "+absl/types/variant.h", ] - -specific_include_rules = { - "common_types\.h" : [ - # TODO(bugs.webrtc.org/5876). For migration of AudioFrameType. - # Delete when downstream code is updated. - "+modules/audio_coding/include/audio_coding_module_typedefs.h" - ], -} diff --git a/common_types.h b/common_types.h index 3caf1553e9..f93769c8d6 100644 --- a/common_types.h +++ b/common_types.h @@ -14,18 +14,6 @@ #include // For size_t #include -// TODO(bugs.webrtc.org/7660): Delete include once downstream code is updated. -#include "api/video/video_codec_type.h" -// TODO(bugs.webrtc.org/5876): For AudioFrameType. Delete when downstream code -// is updated. -#include "modules/audio_coding/include/audio_coding_module_typedefs.h" - -#if defined(_MSC_VER) -// Disable "new behavior: elements of array will be default initialized" -// warning. Affects OverUseDetectorOptions. -#pragma warning(disable : 4351) -#endif - namespace webrtc { // Statistics for RTCP packet types. diff --git a/test/scenario/scenario_config.h b/test/scenario/scenario_config.h index 5532dba809..e62c61086c 100644 --- a/test/scenario/scenario_config.h +++ b/test/scenario/scenario_config.h @@ -19,6 +19,7 @@ #include "api/units/data_rate.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" +#include "api/video/video_codec_type.h" #include "common_types.h" // NOLINT(build/include) #include "test/frame_generator.h" #include "test/scenario/quality_info.h"