diff --git a/webrtc/modules/modules.gyp b/webrtc/modules/modules.gyp
index b0c5bb5aa6..12200fe8f7 100644
--- a/webrtc/modules/modules.gyp
+++ b/webrtc/modules/modules.gyp
@@ -98,9 +98,9 @@
             '<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
             '<(webrtc_root)/modules/video_coding/codecs/vp8/vp8.gyp:webrtc_vp8',
             '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
-            '<(webrtc_root)/test/test.gyp:test_support_main',
             '<(webrtc_root)/test/test.gyp:frame_generator',
-            '<(webrtc_root)/test/test.gyp:rtcp_packet_parser',
+            '<(webrtc_root)/test/test.gyp:rtp_test_utils',
+            '<(webrtc_root)/test/test.gyp:test_support_main',
           ],
           'sources': [
             'audio_coding/main/acm2/acm_opus_unittest.cc',
diff --git a/webrtc/test/test.gyp b/webrtc/test/test.gyp
index 92f007bdfc..51864331ee 100644
--- a/webrtc/test/test.gyp
+++ b/webrtc/test/test.gyp
@@ -53,11 +53,13 @@
       ],
     },
     {
-      'target_name': 'rtcp_packet_parser',
+      'target_name': 'rtp_test_utils',
       'type': 'static_library',
       'sources': [
         'rtcp_packet_parser.cc',
         'rtcp_packet_parser.h',
+        'rtp_file_reader.cc',
+        'rtp_file_reader.h',
       ],
       'dependencies': [
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
diff --git a/webrtc/test/webrtc_test_common.gyp b/webrtc/test/webrtc_test_common.gyp
index 1f153ae391..cd076639c4 100644
--- a/webrtc/test/webrtc_test_common.gyp
+++ b/webrtc/test/webrtc_test_common.gyp
@@ -35,8 +35,6 @@
         'mock_transport.h',
         'null_transport.cc',
         'null_transport.h',
-        'rtp_file_reader.cc',
-        'rtp_file_reader.h',
         'rtp_rtcp_observer.h',
         'run_loop.cc',
         'run_loop.h',
@@ -63,6 +61,7 @@
         '<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
         '<(webrtc_root)/test/test.gyp:frame_generator',
         '<(webrtc_root)/test/test.gyp:test_support',
+        '<(webrtc_root)/test/test.gyp:rtp_test_utils',
         '<(webrtc_root)/webrtc.gyp:webrtc',
       ],
     },