Added an empty voice engine unit test binary in order to get correct coverage measurements. This will make the voice engine show up in the coverage measurements. The empty test is necessary to get the coverage tool to pick it up (and it will be easier to start writing unit tests for the voice engine later).

BUG=
TEST=

Review URL: http://webrtc-codereview.appspot.com/334003

git-svn-id: http://webrtc.googlecode.com/svn/trunk@1245 4adac7df-926f-26a2-2b94-8c16560cd09d

src/voice_engine/main/source/channel_unittest.cc

@@ -0,0 +1,15 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+#include "channel.h"
+#include "gtest/gtest.h"
+
+// Empty test just to get coverage metrics.
+TEST(ChannelTest, EmptyTestToGetCodeCoverage) {}

src/voice_engine/main/source/voice_engine_core.gypi

@@ -116,6 +116,39 @@
       ],
     },
   ],
+  'conditions': [
+    ['build_with_chromium==0', {
+      'targets': [
+        {
+          'target_name': 'voice_engine_unittests',
+          'type': 'executable',
+          'dependencies': [
+            'voice_engine_core',
+            '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
+            '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
+            '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
+            '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
+            '<(webrtc_root)/modules/modules.gyp:audio_device',
+            '<(webrtc_root)/modules/modules.gyp:audio_processing',
+            '<(webrtc_root)/modules/modules.gyp:media_file',
+            '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
+            '<(webrtc_root)/modules/modules.gyp:udp_transport',
+            '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
+            '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+            '<(webrtc_root)/../test/test.gyp:test_support_main',
+            '<(webrtc_root)/../testing/gtest.gyp:gtest',
+          ],
+          'include_dirs': [            
+            '../../..',
+            '../interface',
+          ],
+          'sources': [
+            'channel_unittest.cc',
+          ],
+        },
+      ], # targets
+    }], # build_with_chromium
+  ], # conditions
 }
 
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