Replaced the data logging functionality in the AEC with a generic logging functionality

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")
@@ skipping 63 matching lines @@
     "gain_control_impl.h",
     "high_pass_filter_impl.cc",
     "high_pass_filter_impl.h",
     "include/audio_processing.h",
     "intelligibility/intelligibility_enhancer.cc",
     "intelligibility/intelligibility_enhancer.h",
     "intelligibility/intelligibility_utils.cc",
     "intelligibility/intelligibility_utils.h",
     "level_estimator_impl.cc",
     "level_estimator_impl.h",
-    "logging/aec_logging.h",
-    "logging/aec_logging_file_handling.cc",
-    "logging/aec_logging_file_handling.h",
+    "logging/apm_data_dumper.cc",
+    "logging/apm_data_dumper.h",
     "noise_suppression_impl.cc",
     "noise_suppression_impl.h",
     "render_queue_item_verifier.h",
     "rms_level.cc",
     "rms_level.h",
     "splitting_filter.cc",
     "splitting_filter.h",
     "three_band_filter_bank.cc",
     "three_band_filter_bank.h",
     "transient/common.h",
@@ skipping 45 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" ]
+    defines += [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+  } else {
+    defines += [ "WEBRTC_AEC_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 80 matching lines @@
       "aec/aec_core_sse2.cc",
       "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" ]
+    } else {
+      defines = [ "WEBRTC_AEC_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",
       "ns/nsx_core_neon.c",
@@ skipping 15 matching lines @@
         "-ffat-lto-objects",
       ]
     }
 
     configs += [ "../..:common_config" ]
     public_configs = [ "../..:common_inherited_config" ]
 
     deps = [
       "../../common_audio",
     ]
+
+    if (aec_debug_dump) {
+      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
+    } else {
+      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
+    }
   }
 }