Update tests and DTX check for Opus 1.1.3.

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 474 matching lines @@
                    FLAGS_gen_ref);
 }
 
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     defined(WEBRTC_CODEC_OPUS)
 #define MAYBE_TestOpusBitExactness TestOpusBitExactness
 #else
 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
 #endif
-TEST_F(NetEqDecodingTest, DISABLED_TestOpusBitExactness) {
+TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
 
   const std::string output_checksum = PlatformChecksum(
-      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
-      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
-      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
-      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4");
+      "9d7d52bc94e941d106aa518f324f16a58d231586",
+      "9d7d52bc94e941d106aa518f324f16a58d231586",
+      "9d7d52bc94e941d106aa518f324f16a58d231586",
+      "9d7d52bc94e941d106aa518f324f16a58d231586");
 
   const std::string network_stats_checksum = PlatformChecksum(
-      "6eab76efbde753d4dde38983445ca16b4ce59b39",
-      "6eab76efbde753d4dde38983445ca16b4ce59b39",
-      "6eab76efbde753d4dde38983445ca16b4ce59b39",
-      "6eab76efbde753d4dde38983445ca16b4ce59b39");
+      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef",
+      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef",
+      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef",
+      "191af29ed3b8b6dd4c4cc94dc3f33bdf48f055ef");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
@@ skipping 1259 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc