Index: webrtc/modules/audio_processing/BUILD.gn |
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn |
index a3434836d84d9068b739d4c089d90051a80f493a..c18c1d80cfe054063253707ad24d3dede9b647bd 100644 |
--- a/webrtc/modules/audio_processing/BUILD.gn |
+++ b/webrtc/modules/audio_processing/BUILD.gn |
@@ -81,9 +81,8 @@ source_set("audio_processing") { |
"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", |
@@ -149,7 +148,9 @@ source_set("audio_processing") { |
] |
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) { |
@@ -250,6 +251,12 @@ if (current_cpu == "x86" || current_cpu == "x64") { |
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" ] |
+ } |
} |
} |
@@ -285,5 +292,11 @@ if (rtc_build_with_neon) { |
deps = [ |
"../../common_audio", |
] |
+ |
+ if (aec_debug_dump) { |
+ defines = [ "WEBRTC_AEC_DEBUG_DUMP=1" ] |
+ } else { |
+ defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ] |
+ } |
} |
} |