Changed OLA window for neteq. Old code didnt work well with 48khz

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 424 matching lines @@
     !defined(WEBRTC_ARCH_ARM64)
 #define MAYBE_TestBitExactness TestBitExactness
 #else
 #define MAYBE_TestBitExactness DISABLED_TestBitExactness
 #endif
 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
   const std::string input_rtp_file =
       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
 
   const std::string output_checksum = PlatformChecksum(
-      "5a8184bc60c0d7dddb50af8966360675476a8d8b",
-      "be982d2c5685dd1ca4ea5d352283df50e8e5b46d",
-      "5a8184bc60c0d7dddb50af8966360675476a8d8b",
-      "c86aec95439748f4949de95b50c94be291118615");
+      "09fa7646e2ad032a0b156177b95f09012430f81f",
+      "1c64eb8b55ce8878676c6a1e6ddd78f48de0668b",
+      "09fa7646e2ad032a0b156177b95f09012430f81f",
+      "759fef89a5de52bd17e733dc255c671ce86be909");
 
   const std::string network_stats_checksum = PlatformChecksum(
       "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
       "c8b2a93842e48d014f7e6efe10ae96cb3892b129",
       "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
       "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
@@ skipping 13 matching lines @@
     !WEBRTC_OPUS_SUPPORT_120MS_PTIME
 #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 output_checksum = PlatformChecksum(
-      "9d7d52bc94e941d106aa518f324f16a58d231586",
-      "9d7d52bc94e941d106aa518f324f16a58d231586",
-      "9d7d52bc94e941d106aa518f324f16a58d231586",
-      "9d7d52bc94e941d106aa518f324f16a58d231586");
+      "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
+      "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
+      "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
+      "6237dd113ad80d7764fe4c90b55b2ec035eae64e");
 
   const std::string network_stats_checksum = PlatformChecksum(
       "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
       "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
       "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
       "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
@@ skipping 1086 matching lines @@
     if (muted) {
       EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
     } else {
       EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
     }
   }
   EXPECT_FALSE(muted);
 }
 
 }  // namespace webrtc