| Index: webrtc/modules/audio_processing/BUILD.gn
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| diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
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| index a3434836d84d9068b739d4c089d90051a80f493a..c18c1d80cfe054063253707ad24d3dede9b647bd 100644
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| --- a/webrtc/modules/audio_processing/BUILD.gn
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| +++ b/webrtc/modules/audio_processing/BUILD.gn
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| @@ -81,9 +81,8 @@ source_set("audio_processing") {
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|      "intelligibility/intelligibility_utils.h",
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|      "level_estimator_impl.cc",
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|      "level_estimator_impl.h",
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| -    "logging/aec_logging.h",
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| -    "logging/aec_logging_file_handling.cc",
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| -    "logging/aec_logging_file_handling.h",
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| +    "logging/apm_data_dumper.cc",
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| +    "logging/apm_data_dumper.h",
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|      "noise_suppression_impl.cc",
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|      "noise_suppression_impl.h",
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|      "render_queue_item_verifier.h",
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| @@ -149,7 +148,9 @@ source_set("audio_processing") {
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|    ]
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|  
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|    if (aec_debug_dump) {
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| -    defines += [ "WEBRTC_AEC_DEBUG_DUMP" ]
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| +    defines += [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
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| +  } else {
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| +    defines += [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
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|    }
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|  
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|    if (aec_untrusted_delay_for_testing) {
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| @@ -250,6 +251,12 @@ if (current_cpu == "x86" || current_cpu == "x64") {
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|  
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|      configs += [ "../..:common_config" ]
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|      public_configs = [ "../..:common_inherited_config" ]
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| +
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| +    if (aec_debug_dump) {
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| +      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
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| +    } else {
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| +      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
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| +    }
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|    }
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|  }
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|  
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| @@ -285,5 +292,11 @@ if (rtc_build_with_neon) {
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|      deps = [
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|        "../../common_audio",
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|      ]
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| +
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| +    if (aec_debug_dump) {
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| +      defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ]
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| +    } else {
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| +      defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ]
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| +    }
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|    }
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|  }
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| 
 |