Update sha1 files and unittests for Opus v1.1.2

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 550 matching lines @@
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     defined(WEBRTC_CODEC_OPUS)
 #define MAYBE_TestOpusBitExactness TestOpusBitExactness
 #else
 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
 #endif
 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
   const std::string input_ref_file =
+      // The pcm files were generated by using Opus v1.1.2 to decode the RTC
+      // file generated by Opus v1.1
       webrtc::test::ResourcePath("audio_coding/neteq4_opus_ref", "pcm");
   const std::string network_stat_ref_file =
+      // The network stats file was generated when using Opus v1.1.2 to decode
+      // the RTC file generated by Opus v1.1
       webrtc::test::ResourcePath("audio_coding/neteq4_opus_network_stats",
                                  "dat");
   const std::string rtcp_stat_ref_file =
       webrtc::test::ResourcePath("audio_coding/neteq4_opus_rtcp_stats", "dat");
 
   if (FLAGS_gen_ref) {
     DecodeAndCompare(input_rtp_file, "", "", "");
   } else {
     DecodeAndCompare(input_rtp_file,
                      input_ref_file,
@@ skipping 1070 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