diff --git a/BUILD.gn b/BUILD.gn
index 8243281217..96210fd625 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -249,7 +249,6 @@ config("common_config") {
       # TODO(kjellander): Cleanup unused ones and move defines closer to the
       # source when webrtc:4256 is completed.
       "HAVE_WEBRTC_VIDEO",
-      "HAVE_WEBRTC_VOICE",
       "LOGGING_INSIDE_WEBRTC",
     ]
   } else {
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 87cdfd5e0e..c05a6aec91 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -20,10 +20,7 @@ group("media") {
 }
 
 config("rtc_media_defines_config") {
-  defines = [
-    "HAVE_WEBRTC_VIDEO",
-    "HAVE_WEBRTC_VOICE",
-  ]
+  defines = [ "HAVE_WEBRTC_VIDEO" ]
 }
 
 rtc_source_set("rtc_h264_profile_id") {
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 4dec78e643..691f7d7bee 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -8,8 +8,6 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifdef HAVE_WEBRTC_VOICE
-
 #include "media/engine/webrtc_voice_engine.h"
 
 #include <algorithm>
@@ -2286,5 +2284,3 @@ bool WebRtcVoiceMediaChannel::MaybeDeregisterUnsignaledRecvStream(
   return false;
 }
 }  // namespace cricket
-
-#endif  // HAVE_WEBRTC_VOICE