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

webrtc/modules/audio_processing/audio_processing.gypi

 # Copyright (c) 2012 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.
 
 {
   'variables': {
     'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets',
+    # Outputs some low-level debug files.
+    'aec_debug_dump%': 0,
   },
   'targets': [
     {
       'target_name': 'audio_processing',
       'type': 'static_library',
       'variables': {
         # Outputs some low-level debug files.
-        'aec_debug_dump%': 0,
         'agc_debug_dump%': 0,
 
         # 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%': 0,
       },
       'dependencies': [
         '<(webrtc_root)/base/base.gyp:rtc_base_approved',
         '<(webrtc_root)/common.gyp:webrtc_common',
@@ skipping 54 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 36 matching lines @@
         'vad/vad_circular_buffer.cc',
         'vad/vad_circular_buffer.h',
         'vad/voice_activity_detector.cc',
         'vad/voice_activity_detector.h',
         'vad/voice_gmm_tables.h',
         'voice_detection_impl.cc',
         'voice_detection_impl.h',
       ],
       'conditions': [
         ['aec_debug_dump==1', {
-          'defines': ['WEBRTC_AEC_DEBUG_DUMP',],
+          'defines': ['WEBRTC_AEC_DEBUG_DUMP=1',],
+        }, {
+          'defines': ['WEBRTC_AEC_DEBUG_DUMP=0',],
         }],
         ['aec_untrusted_delay_for_testing==1', {
           'defines': ['WEBRTC_UNTRUSTED_DELAY',],
         }],
         ['agc_debug_dump==1', {
           'defines': ['WEBRTC_AGC_DEBUG_DUMP',],
         }],
         ['enable_protobuf==1', {
           'dependencies': ['audioproc_debug_proto'],
           'defines': ['WEBRTC_AUDIOPROC_DEBUG_DUMP'],
@@ skipping 78 matching lines @@
     ['target_arch=="ia32" or target_arch=="x64"', {
       'targets': [
         {
           'target_name': 'audio_processing_sse2',
           'type': 'static_library',
           'sources': [
             'aec/aec_core_sse2.cc',
             'aec/aec_rdft_sse2.cc',
           ],
           'conditions': [
+            ['aec_debug_dump==1', {
+              'defines': ['WEBRTC_AEC_DEBUG_DUMP=1',],
+            }, {
+              'defines': ['WEBRTC_AEC_DEBUG_DUMP=0',],
+            }],
             ['os_posix==1', {
               'cflags': [ '-msse2', ],
               'xcode_settings': {
                 'OTHER_CFLAGS': [ '-msse2', ],
               },
             }],
           ],
         },
       ],
     }],
     ['build_with_neon==1', {
       'targets': [{
         'target_name': 'audio_processing_neon',
         'type': 'static_library',
         'includes': ['../../build/arm_neon.gypi',],
         'dependencies': [
           '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
         ],
         'sources': [
           'aec/aec_core_neon.cc',
           'aec/aec_rdft_neon.cc',
           'aecm/aecm_core_neon.cc',
           'ns/nsx_core_neon.c',
         ],
+        'conditions': [
+          ['aec_debug_dump==1', {
+            'defines': ['WEBRTC_AEC_DEBUG_DUMP=1',],
+          }],
+          ['aec_debug_dump==0', {
+            'defines': ['WEBRTC_AEC_DEBUG_DUMP=0',],
+          }],
+        ],
       }],
     }],
   ],
 }