Split webrtc/video into webrtc/{audio,call,video}.

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 128 matching lines @@
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         'test/webrtc_test_common.gyp:webrtc_test_common',
         'test/webrtc_test_common.gyp:webrtc_test_renderer',
         '<(webrtc_root)/modules/modules.gyp:video_capture',
         '<(webrtc_root)/modules/modules.gyp:video_render',
         '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
         'webrtc',
       ],
     },
     {
-      # TODO(pbos): Rename target to webrtc_tests or rtc_tests, this target is
-      # not meant to only include video.
+      # TODO(pbos): Add separate target webrtc_audio_tests and move files there.
       'target_name': 'video_engine_tests',
       'type': '<(gtest_target_type)',
       'sources': [
+        'audio/audio_receive_stream_unittest.cc',
+        'call/bitrate_estimator_tests.cc',
+        'call/packet_injection_tests.cc',
         'test/common_unittest.cc',
         'test/testsupport/metrics/video_metrics_unittest.cc',
         'tools/agc/agc_manager_unittest.cc',
-        'video/audio_receive_stream_unittest.cc',
-        'video/bitrate_estimator_tests.cc',
         'video/end_to_end_tests.cc',
-        'video/packet_injection_tests.cc',
         'video/send_statistics_proxy_unittest.cc',
         'video/video_capture_input_unittest.cc',
         'video/video_decoder_unittest.cc',
         'video/video_encoder_unittest.cc',
         'video/video_send_stream_tests.cc',
       ],
       'dependencies': [
         '<(DEPTH)/testing/gmock.gyp:gmock',
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(webrtc_root)/common.gyp:webrtc_common',
@@ skipping 16 matching lines @@
         }],
         ['enable_protobuf==1', {
           'defines': [
             'ENABLE_RTC_EVENT_LOG',
           ],
           'dependencies': [
             'webrtc.gyp:rtc_event_log',
             'webrtc.gyp:rtc_event_log_proto',
           ],
           'sources': [
-            'video/rtc_event_log_unittest.cc',
+            'call/rtc_event_log_unittest.cc',
           ],
         }],
       ],
     },
     {
       'target_name': 'webrtc_perf_tests',
       'type': '<(gtest_target_type)',
       'sources': [
+        'call/call_perf_tests.cc',
         'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
         'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
-
         'tools/agc/agc_manager_integrationtest.cc',
-        'video/call_perf_tests.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:video_capture',
         '<(webrtc_root)/test/test.gyp:channel_transport',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
@@ skipping 80 matching lines @@
             'build/isolate.gypi',
           ],
           'sources': [
             'webrtc_perf_tests.isolate',
           ],
         },
       ],
     }],
   ],
 }