AecDump interface

webrtc/modules/audio_processing/BUILD.gn

Index: webrtc/modules/audio_processing/BUILD.gn
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index 36f55756f825fe0b719d6a6c0d0ca1563ade5c65..11ea508eaeca89a2eb6634fef3e41e66aa96e177 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -233,7 +233,14 @@ rtc_static_library("audio_processing") {
     "../../audio/utility:audio_frame_operations",
     "../../base:gtest_prod",
     "../audio_coding:isac",
+    "aec_dumper",
   ]
+  if (rtc_enable_protobuf) {
+    # Will change when AEC-dumper is fully implemented.
+    deps += [ "aec_dumper:aec_dumper_no_pb" ]
+  } else {
+    deps += [ "aec_dumper:aec_dumper_no_pb" ]
+  }
   public_deps = [
     ":audio_processing_c",
   ]
@@ -248,6 +255,7 @@ rtc_static_library("audio_processing") {
     defines += [ "WEBRTC_UNTRUSTED_DELAY" ]
   }
 
+  # Will be removed when AEC-dumper is fully implemented.
   if (rtc_enable_protobuf) {
     defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
     deps += [ ":audioproc_debug_proto" ]
@@ -530,6 +538,7 @@ if (rtc_include_tests) {
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
       "../audio_coding:neteq_unittest_tools",
+      "aec_dumper:aec_dumper_unittests",
       "test/conversational_speech:unittest",
       "//testing/gmock",
       "//testing/gtest",
@@ -703,6 +712,7 @@ if (rtc_include_tests) {
         ":audioproc_protobuf_utils",
         ":audioproc_test_utils",
         "../../base:rtc_base_approved",
+        "../../base:rtc_task_queue",
         "../../common_audio:common_audio",
         "../../system_wrappers",
         "../../system_wrappers:system_wrappers_default",