| 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" ]
|
| }
|
| }
|
| }
|
|
|