Step 1 to prepare call_test.* for combined audio/video tests.

webrtc/webrtc_tests.gypi

 # Copyright (c) 2013 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.
 {
   'targets': [
     {
@@ skipping 201 matching lines @@
             'call/rtc_event_log_unittest.cc',
           ],
         }],
       ],
     },
     {
       'target_name': 'webrtc_perf_tests',
       'type': '<(gtest_target_type)',
       'sources': [
         'call/call_perf_tests.cc',
+        'call/rampup_tests.cc',
+        'call/rampup_tests.h',
         'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
         'modules/audio_processing/audio_processing_performance_unittest.cc',
         'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
         'video/full_stack.cc',
-        'video/rampup_tests.cc',
-        'video/rampup_tests.h',
       ],
       'dependencies': [
         '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/modules/modules.gyp:audio_processing',
         '<(webrtc_root)/modules/modules.gyp:audioproc_test_utils',
         '<(webrtc_root)/modules/modules.gyp:video_capture',
         '<(webrtc_root)/test/test.gyp:channel_transport',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
         'video_quality_test',
@@ skipping 149 matching lines @@
             'build/isolate.gypi',
           ],
           'sources': [
             'webrtc_perf_tests.isolate',
           ],
         },
       ],
     }],
   ],
 }