Cleaning neteq_unittest resource files.

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  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.
  */
 
@@ skipping 506 matching lines @@
 
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
 #define MAYBE_TestBitExactness TestBitExactness
 #else
 #define MAYBE_TestBitExactness DISABLED_TestBitExactness
 #endif
 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
-  const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
-      "resources/audio_coding/neteq_universal_new.rtp";
+  const std::string input_rtp_file =
+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
   const std::string input_ref_file =
       webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
 #if defined(_MSC_VER) && (_MSC_VER >= 1700)
   // For Visual Studio 2012 and later, we will have to use the generic reference
   // file, rather than the windows-specific one.
   const std::string network_stat_ref_file = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq4_network_stats.dat";
@@ skipping 1100 matching lines @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc