Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific t…

webrtc/modules/audio_coding/BUILD.gn

Index: webrtc/modules/audio_coding/BUILD.gn
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 8a27900edf8bf05db90d64a262a3885f177bcd62..4b7e417a924dd9f80956fb8ef09c0f109798ff7d 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -7,6 +7,7 @@
 # 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")
 import("../../build/webrtc.gni")
 
 config("audio_coding_config") {
@@ -79,6 +80,35 @@ source_set("audio_coding") {
   }
 }
 
+proto_library("acm_dump_proto") {
+  sources = [
+    "main/acm2/dump.proto",
+  ]
+  proto_out_dir = "webrtc/audio_coding"
+}
+
+source_set("acm_dump") {
+  sources = [
+    "main/acm2/acm_dump.cc",
+    "main/acm2/acm_dump.h",
+  ]
+
+  defines = []
+
+  configs += [ "../..:common_config" ]
+
+  public_configs = [ "../..:common_inherited_config" ]
+
+  deps = [
+    ":acm_dump_proto",
+    "../..:webrtc_common",
+  ]
+
+  if (rtc_enable_protobuf) {
+    defines += [ "RTC_AUDIOCODING_DEBUG_DUMP" ]
+  }
+}
+
 source_set("audio_decoder_interface") {
   sources = [
     "codecs/audio_decoder.cc",