Adding debug dump to audio network adaptor.

webrtc/modules/audio_coding/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/webrtc.gni")
 import("audio_coding.gni")
@@ skipping 691 matching lines @@
     "audio_network_adaptor/audio_network_adaptor_impl.cc",
     "audio_network_adaptor/audio_network_adaptor_impl.h",
     "audio_network_adaptor/bitrate_controller.cc",
     "audio_network_adaptor/bitrate_controller.h",
     "audio_network_adaptor/channel_controller.cc",
     "audio_network_adaptor/channel_controller.h",
     "audio_network_adaptor/controller.cc",
     "audio_network_adaptor/controller.h",
     "audio_network_adaptor/controller_manager.cc",
     "audio_network_adaptor/controller_manager.h",
+    "audio_network_adaptor/debug_dump_writer.cc",
+    "audio_network_adaptor/debug_dump_writer.h",
     "audio_network_adaptor/dtx_controller.cc",
     "audio_network_adaptor/dtx_controller.h",
     "audio_network_adaptor/frame_length_controller.cc",
     "audio_network_adaptor/frame_length_controller.h",
     "audio_network_adaptor/include/audio_network_adaptor.h",
     "audio_network_adaptor/smoothing_filter.cc",
     "audio_network_adaptor/smoothing_filter.h",
   ]
   configs += [ "../..:common_config" ]
   public_configs = [ "../..:common_inherited_config" ]
+
+  if (rtc_enable_protobuf) {
+    deps = [
+      ":ana_debug_dump_proto",
+    ]
+    defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
+  }
 }
 
 config("neteq_config") {
   include_dirs = [
     # Need Opus header files for the audio classifier.
     "//third_party/opus/src/celt",
     "//third_party/opus/src/src",
   ]
 }
 
@@ skipping 236 matching lines @@
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
     }
     if (is_ios) {
       deps += [ ":audio_decoder_unittests_bundle_data" ]
     }
   }  # audio_decoder_unittests
 
   if (rtc_enable_protobuf) {
+    proto_library("ana_debug_dump_proto") {
+      sources = [
+        "audio_network_adaptor/debug_dump.proto",
+      ]
+      proto_out_dir = "webrtc/modules/audio_coding/audio_network_adaptor"
+    }
+
     proto_library("neteq_unittest_proto") {
       sources = [
         "neteq/neteq_unittest.proto",
       ]
       proto_out_dir = "webrtc/audio_coding/neteq"
     }
 
     rtc_source_set("rtc_event_log_source") {
       testonly = true
       sources = [
@@ skipping 583 matching lines @@
       "//testing/gtest",
     ]
 
     if (is_clang) {
       # Suppress warnings from Chrome's Clang plugins.
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }