Delete root header file typedef.h.

Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <[email protected]>
Reviewed-by: Fredrik Solenberg <[email protected]>
Commit-Queue: Niels Moller <[email protected]>
Cr-Commit-Position: refs/heads/master@{#24100}

Author: Niels Möller

BUILD.gn

@@ -384,22 +384,11 @@ if (!build_with_chromium) {
   }
 }
 
-rtc_source_set("typedefs") {
-  sources = [
-    "typedefs.h",
-  ]
-  deps = [
-    "rtc_base/system:arch",
-    "rtc_base/system:unused",
-  ]
-}
-
 rtc_source_set("webrtc_common") {
   sources = [
     "common_types.h",
   ]
   deps = [
-    ":typedefs",
     "api:array_view",
     "api/video:video_bitrate_allocation",
     "rtc_base:checks",

api/BUILD.gn

@@ -111,7 +111,6 @@ rtc_static_library("libjingle_peerconnection_api") {
     # Basically, don't add stuff here. You might break sensitive downstream
     # targets like pnacl. API should not depend on anything outside of this
     # file, really. All these should arguably go away in time.
-    "..:typedefs",
     "..:webrtc_common",
     "../logging:rtc_event_log_api",
     "../media:rtc_media_config",

api/audio/BUILD.gn

@@ -16,7 +16,6 @@ rtc_source_set("audio_frame_api") {
   ]
 
   deps = [
-    "../../:typedefs",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",
   ]

api/audio/audio_frame.h

@@ -12,9 +12,9 @@
 #define API_AUDIO_AUDIO_FRAME_H_
 
 #include <stddef.h>
+#include <stdint.h>
 
 #include "rtc_base/constructormagic.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

api/audio_codecs/BUILD.gn

@@ -31,7 +31,6 @@ rtc_source_set("audio_codecs_api") {
   deps = [
     "..:array_view",
     "../..:webrtc_common",
-    "../../:typedefs",
     "../../rtc_base:checks",
     "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",

api/audio_codecs/audio_decoder.h

@@ -18,7 +18,6 @@
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructormagic.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

api/audio_codecs/audio_encoder.h

@@ -20,7 +20,6 @@
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/deprecation.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

api/rtp_headers.h

@@ -25,7 +25,6 @@
 #include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/checks.h"
 #include "rtc_base/deprecation.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

api/video/BUILD.gn

@@ -84,7 +84,6 @@ rtc_source_set("video_bitrate_allocation") {
     "video_bitrate_allocation.h",
   ]
   deps = [
-    "../..:typedefs",
     "../../rtc_base:checks",
     "../../rtc_base:safe_conversions",
     "../../rtc_base:stringutils",

api/video/video_bitrate_allocation.h

@@ -16,7 +16,6 @@
 #include <vector>
 
 #include "absl/types/optional.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

audio/utility/BUILD.gn

@@ -22,7 +22,6 @@ rtc_static_library("audio_frame_operations") {
 
   deps = [
     "../..:webrtc_common",
-    "../../:typedefs",
     "../../api/audio:audio_frame_api",
     "../../rtc_base:checks",
     "../../rtc_base:rtc_base_approved",

audio/utility/audio_frame_operations.h

@@ -14,7 +14,6 @@
 #include <stddef.h>
 
 #include "api/audio/audio_frame.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

call/BUILD.gn

@@ -30,7 +30,6 @@ rtc_source_set("call_interfaces") {
     ":rtp_interfaces",
     ":video_stream_api",
     "..:webrtc_common",
-    "../:typedefs",
     "../api:fec_controller_api",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
@@ -236,7 +235,6 @@ rtc_source_set("video_stream_api") {
   ]
   deps = [
     ":rtp_interfaces",
-    "../:typedefs",
     "../:webrtc_common",
     "../api:libjingle_peerconnection_api",
     "../api:transport_api",
@@ -258,7 +256,6 @@ rtc_source_set("fake_network") {
   ]
   deps = [
     ":call_interfaces",
-    "..:typedefs",
     "..:webrtc_common",
     "../api:simulated_network_api",
     "../api:transport_api",

call/audio_receive_stream.h

@@ -24,7 +24,6 @@
 #include "call/rtp_config.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "rtc_base/scoped_ref_ptr.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 class AudioSinkInterface;

call/audio_send_stream.h

@@ -25,7 +25,6 @@
 #include "call/rtp_config.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
 #include "rtc_base/scoped_ref_ptr.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

call/fake_network_pipe.h

@@ -28,7 +28,6 @@
 #include "rtc_base/criticalsection.h"
 #include "rtc_base/random.h"
 #include "rtc_base/thread_annotations.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

common_audio/BUILD.gn

@@ -55,12 +55,12 @@ rtc_static_library("common_audio") {
     ":fft4g",
     ":sinc_resampler",
     "..:webrtc_common",
-    "../:typedefs",
     "../rtc_base:checks",
     "../rtc_base:gtest_prod",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/memory:aligned_array",
     "../rtc_base/memory:aligned_malloc",
+    "../rtc_base/system:arch",
     "../system_wrappers",
     "../system_wrappers:cpu_features_api",
     "//third_party/abseil-cpp/absl/types:optional",
@@ -192,11 +192,11 @@ rtc_source_set("common_audio_c") {
     ":common_audio_cc",
     ":fft4g",
     "..:webrtc_common",
-    "../:typedefs",
     "../rtc_base:checks",
     "../rtc_base:compile_assert_c",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:sanitizer",
+    "../rtc_base/system:arch",
     "../system_wrappers",
     "../system_wrappers:cpu_features_api",
   ]
@@ -218,7 +218,6 @@ rtc_source_set("common_audio_cc") {
 
   deps = [
     "..:webrtc_common",
-    "../:typedefs",
     "../rtc_base:rtc_base_approved",
     "../system_wrappers",
   ]
@@ -230,10 +229,10 @@ rtc_source_set("sinc_resampler") {
   ]
   deps = [
     "..:webrtc_common",
-    "../:typedefs",
     "../rtc_base:gtest_prod",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/memory:aligned_malloc",
+    "../rtc_base/system:arch",
     "../system_wrappers",
   ]
 }
@@ -257,6 +256,7 @@ rtc_source_set("fir_filter_factory") {
     ":fir_filter",
     "../rtc_base:checks",
     "../rtc_base:rtc_base_approved",
+    "../rtc_base/system:arch",
     "../system_wrappers:cpu_features_api",
   ]
   if (current_cpu == "x86" || current_cpu == "x64") {
@@ -353,6 +353,7 @@ if (rtc_build_with_neon) {
       ":common_audio_c",
       "../rtc_base:checks",
       "../rtc_base:rtc_base_approved",
+      "../rtc_base/system:arch",
     ]
   }
 }
@@ -404,10 +405,10 @@ if (rtc_include_tests) {
       ":fir_filter_factory",
       ":sinc_resampler",
       "..:webrtc_common",
-      "../:typedefs",
       "../rtc_base:checks",
       "../rtc_base:rtc_base_approved",
       "../rtc_base:rtc_base_tests_utils",
+      "../rtc_base/system:arch",
       "../system_wrappers:cpu_features_api",
       "../test:fileutils",
       "../test:test_main",

common_audio/audio_util.cc

@@ -10,8 +10,6 @@
 
 #include "common_audio/include/audio_util.h"
 
-#include "typedefs.h"  // NOLINT(build/include)
-
 namespace webrtc {
 
 void FloatToS16(const float* src, size_t size, int16_t* dest) {

common_audio/audio_util_unittest.cc

@@ -13,7 +13,6 @@
 #include "rtc_base/arraysize.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 namespace {

common_audio/fir_filter_factory.cc

@@ -12,6 +12,7 @@
 
 #include "common_audio/fir_filter_c.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
 #include "system_wrappers/include/cpu_features_wrapper.h"
 
 #if defined(WEBRTC_HAS_NEON)

common_audio/include/audio_util.h

@@ -17,7 +17,6 @@
 #include <limits>
 
 #include "rtc_base/checks.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

common_audio/resampler/include/push_resampler.h

@@ -13,8 +13,6 @@
 
 #include <memory>
 
-#include "typedefs.h"  // NOLINT(build/include)
-
 namespace webrtc {
 
 class PushSincResampler;

common_audio/resampler/include/resampler.h

@@ -16,8 +16,7 @@
 #define COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
 
 #include <stddef.h>
-
-#include "typedefs.h"  // NOLINT(build/include)
+#include <stdint.h>
 
 namespace webrtc {
 

common_audio/resampler/push_sinc_resampler.h

@@ -15,7 +15,6 @@
 
 #include "common_audio/resampler/sinc_resampler.h"
 #include "rtc_base/constructormagic.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

common_audio/resampler/push_sinc_resampler_unittest.cc

@@ -19,7 +19,6 @@
 #include "rtc_base/timeutils.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 namespace {

common_audio/resampler/sinc_resampler.cc

@@ -93,8 +93,8 @@
 #include <limits>
 
 #include "rtc_base/checks.h"
+#include "rtc_base/system/arch.h"
 #include "system_wrappers/include/cpu_features_wrapper.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 

common_audio/resampler/sinc_resampler.h

@@ -19,7 +19,7 @@
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/gtest_prod_util.h"
 #include "rtc_base/memory/aligned_malloc.h"
-#include "typedefs.h"  // NOLINT(build/include)
+#include "rtc_base/system/arch.h"
 
 namespace webrtc {
 

common_audio/resampler/sinc_resampler_unittest.cc

@@ -23,6 +23,7 @@
 #include "common_audio/resampler/sinc_resampler.h"
 #include "common_audio/resampler/sinusoidal_linear_chirp_source.h"
 #include "rtc_base/stringize_macros.h"
+#include "rtc_base/system/arch.h"
 #include "rtc_base/timeutils.h"
 #include "system_wrappers/include/cpu_features_wrapper.h"
 #include "test/gmock.h"

common_audio/signal_processing/complex_fft.c

@@ -17,6 +17,7 @@
 
 #include "common_audio/signal_processing/complex_fft_tables.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/system/arch.h"
 
 #define CFFTSFT 14
 #define CFFTRND 1

common_audio/signal_processing/complex_fft_tables.h

@@ -11,7 +11,7 @@
 #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
 #define COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
 
-#include "typedefs.h"  // NOLINT(build/include)
+#include <stdint.h>
 
 static const int16_t kSinTable1024[] = {
     0,      201,    402,    603,    804,    1005,   1206,   1406,   1607,

common_audio/signal_processing/cross_correlation_neon.c

@@ -9,6 +9,7 @@
  */
 
 #include "common_audio/signal_processing/include/signal_processing_library.h"
+#include "rtc_base/system/arch.h"
 
 #include <arm_neon.h>
 

common_audio/signal_processing/dot_product_with_scale.h

@@ -11,10 +11,9 @@
 #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
 #define COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
 
+#include <stdint.h>
 #include <string.h>
 
-#include "typedefs.h"  // NOLINT(build/include)
-
 #ifdef __cplusplus
 extern "C" {
 #endif

common_audio/signal_processing/include/real_fft.h

@@ -11,7 +11,7 @@
 #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
 #define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
 
-#include "typedefs.h"  // NOLINT(build/include)
+#include <stdint.h>
 
 // For ComplexFFT(), the maximum fft order is 10;
 // WebRTC APM uses orders of only 7 and 8.

common_audio/signal_processing/include/signal_processing_library.h

@@ -19,7 +19,6 @@
 
 #include <string.h>
 #include "common_audio/signal_processing/dot_product_with_scale.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 // Macros specific for the fixed point implementation
 #define WEBRTC_SPL_WORD16_MAX 32767

common_audio/signal_processing/real_fft_unittest.cc

@@ -11,7 +11,6 @@
 #include "common_audio/signal_processing/include/real_fft.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "test/gtest.h"
-#include "typedefs.h"  // NOLINT(build/include)
 
 namespace webrtc {
 namespace {