Index: webrtc/modules/audio_processing/BUILD.gn |
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn |
index 2626b7ebff874043c46ca4e37549e18a369e37e6..fcbb44a831dab0cfeecfcfcc4b2a0a52ac8125eb 100644 |
--- a/webrtc/modules/audio_processing/BUILD.gn |
+++ b/webrtc/modules/audio_processing/BUILD.gn |
@@ -11,9 +11,6 @@ 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. |
@@ -163,10 +160,10 @@ source_set("audio_processing") { |
"../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) { |
@@ -270,10 +267,10 @@ 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" ] |
+ if (apm_debug_dump) { |
+ defines = [ "WEBRTC_APM_DEBUG_DUMP=1" ] |
} else { |
- defines = [ "WEBRTC_AEC_DEBUG_DUMP=0" ] |
+ defines = [ "WEBRTC_APM_DEBUG_DUMP=0" ] |
} |
} |
} |
@@ -311,10 +308,10 @@ if (rtc_build_with_neon) { |
"../../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" ] |
} |
} |
} |