| Index: webrtc/modules/audio_processing/BUILD.gn
|
| diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
|
| index 93ca85b3034034744d5aed736033725fd151c519..4e00328e48ed0f81ce82f2921c267f77dd7bfe3a 100644
|
| --- a/webrtc/modules/audio_processing/BUILD.gn
|
| +++ b/webrtc/modules/audio_processing/BUILD.gn
|
| @@ -230,6 +230,7 @@ rtc_static_library("audio_processing") {
|
|
|
| defines = []
|
| deps = [
|
| + ":aec_dump_interface",
|
| "..:module_api",
|
| "../..:webrtc_common",
|
| "../../audio/utility:audio_frame_operations",
|
| @@ -304,6 +305,17 @@ rtc_static_library("audio_processing") {
|
| ]
|
| }
|
|
|
| +rtc_source_set("aec_dump_interface") {
|
| + sources = [
|
| + "include/aec_dump.cc",
|
| + "include/aec_dump.h",
|
| + ]
|
| +
|
| + deps = [
|
| + "../../base:rtc_base_approved",
|
| + ]
|
| +}
|
| +
|
| rtc_source_set("audio_processing_c") {
|
| visibility = [ ":*" ] # Only targets in this file can depend on this.
|
| sources = [
|
|
|