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