GN: Fix test_support_unittests and MIPS compile issue.

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 348 matching lines @@
       'type': '<(gtest_target_type)',
       'sources': [
         'audio/audio_receive_stream_unittest.cc',
         'audio/audio_send_stream_unittest.cc',
         'audio/audio_state_unittest.cc',
         'call/bitrate_allocator_unittest.cc',
         'call/bitrate_estimator_tests.cc',
         'call/call_unittest.cc',
         'call/packet_injection_tests.cc',
         'call/ringbuffer_unittest.cc',
-        'test/common_unittest.cc',
-        'test/testsupport/metrics/video_metrics_unittest.cc',
         'video/call_stats_unittest.cc',
         'video/encoder_state_feedback_unittest.cc',
         'video/end_to_end_tests.cc',
         'video/overuse_frame_detector_unittest.cc',
         'video/payload_router_unittest.cc',
         'video/report_block_stats_unittest.cc',
         'video/send_delay_stats_unittest.cc',
         'video/send_statistics_proxy_unittest.cc',
         'video/stats_counter_unittest.cc',
         'video/stream_synchronization_unittest.cc',
         'video/video_capture_input_unittest.cc',
         'video/video_decoder_unittest.cc',
         'video/video_encoder_unittest.cc',
         'video/video_send_stream_tests.cc',
         'video/vie_remb_unittest.cc',
       ],
       'dependencies': [
         '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/common.gyp:webrtc_common',
         '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
         '<(webrtc_root)/modules/modules.gyp:video_capture',
         '<(webrtc_root)/test/test.gyp:channel_transport',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-        'test/metrics.gyp:metrics',
         'test/test.gyp:test_common',
         'test/test.gyp:test_main',
+        'test/test.gyp:test_support',
         'webrtc',
       ],
       'conditions': [
         ['rtc_use_h264==1', {
           'defines': [
             'WEBRTC_END_TO_END_H264_TESTS',
           ],
         }],
         ['OS=="android"', {
           'dependencies': [
@@ skipping 254 matching lines @@
             'build/isolate.gypi',
           ],
           'sources': [
             'webrtc_perf_tests.isolate',
           ],
         },
       ],
     }],
   ],
 }