Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump

webrtc/modules/audio_processing/BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("//build/config/arm.gni")
 import("//third_party/protobuf/proto_library.gni")
 import("../../build/webrtc.gni")
 
 declare_args() {
-  # Outputs some low-level debug files.
-  aec_debug_dump = false
-
   # Disables the usual mode where we trust the reported system delay
   # values the AEC receives. The corresponding define is set appropriately
   # in the code, but it can be force-enabled here for testing.
   aec_untrusted_delay_for_testing = false
 }
 
 source_set("audio_processing") {
   sources = [
     "aec/aec_core.cc",
     "aec/aec_core.h",
@@ skipping 129 matching lines @@
 
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
 
   defines = []
   deps = [
     "../..:webrtc_common",
     "../audio_coding:isac",
   ]
 
-  if (aec_debug_dump) {
-    defines += [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+  if (apm_debug_dump) {
+    defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
   } else {
-    defines += [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
+    defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
   }
 
   if (aec_untrusted_delay_for_testing) {
     defines += [ "WEBRTC_UNTRUSTED_DELAY" ]
   }
 
   if (rtc_enable_protobuf) {
     defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
     deps += [ ":audioproc_debug_proto" ]
   }
@@ skipping 83 matching lines @@
       "aec/aec_rdft_sse2.cc",
     ]
 
     if (is_posix) {
       cflags = [ "-msse2" ]
     }
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
-    if (aec_debug_dump) {
-      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+    if (apm_debug_dump) {
+      defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
     } else {
-      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
+      defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
     }
   }
 }
 
 if (rtc_build_with_neon) {
   source_set("audio_processing_neon") {
     sources = [
       "aec/aec_core_neon.cc",
       "aec/aec_rdft_neon.cc",
       "aecm/aecm_core_neon.cc",
@@ skipping 17 matching lines @@
       ]
     }
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
     deps = [
       "../../common_audio",
     ]
 
-    if (aec_debug_dump) {
-      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+    if (apm_debug_dump) {
+      defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ]
     } else {
-      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
+      defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ]
     }
   }
 }
 
 if (rtc_include_tests) {
   executable("audioproc") {
     testonly = true
     sources = [
       "test/process_test.cc",
     ]
@@ skipping 155 matching lines @@
         "test/protobuf_utils.cc",
         "test/protobuf_utils.h",
       ]
 
       deps = [
         ":audioproc_debug_proto",
       ]
     }
   }
 }