AecDump interface

webrtc/modules/audio_processing/BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("//build/config/arm.gni")
 import("//third_party/protobuf/proto_library.gni")
@@ skipping 215 matching lines @@
     "voice_detection_impl.cc",
     "voice_detection_impl.h",
   ]
 
   defines = []
   deps = [
     "../..:webrtc_common",
     "../../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",
   ]
 
   if (apm_debug_dump) {
     defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
   } else {
     defines += [ "WEBRTC_APM_DEBUG_DUMP=0" ]
   }
 
   if (aec_untrusted_delay_for_testing) {
     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" ]
   }
 
   if (rtc_enable_intelligibility_enhancer) {
     defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
     sources += [
       "intelligibility/intelligibility_enhancer.cc",
       "intelligibility/intelligibility_enhancer.h",
@@ skipping 262 matching lines @@
       ":audio_processing",
       ":audioproc_test_utils",
       "../..:webrtc_common",
       "../../base:gtest_prod",
       "../../base:rtc_base",
       "../../base:rtc_base_approved",
       "../../common_audio:common_audio",
       "../../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",
     ]
 
     defines = []
 
     if (apm_debug_dump) {
       defines += [ "WEBRTC_APM_DEBUG_DUMP=1" ]
     } else {
@@ skipping 153 matching lines @@
         "test/wav_based_simulator.cc",
         "test/wav_based_simulator.h",
       ]
 
       deps = [
         ":audio_processing",
         ":audioproc_debug_proto",
         ":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",
         "../../test:test_support",
         "//testing/gtest",
         "//third_party/gflags:gflags",
       ]
     }  # audioproc_f
   }
 
@@ skipping 99 matching lines @@
       ]
 
       deps = [
         ":audioproc_debug_proto",
         "../..:webrtc_common",
         "../../base:rtc_base_approved",
       ]
     }
   }
 }