NetEq: Fix a bug in expand_rate and speech_expand_rate calculation

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 428 matching lines @@
 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(
       "09fa7646e2ad032a0b156177b95f09012430f81f",
       "1c64eb8b55ce8878676c6a1e6ddd78f48de0668b",
       "09fa7646e2ad032a0b156177b95f09012430f81f",
       "759fef89a5de52bd17e733dc255c671ce86be909");
 
-  const std::string network_stats_checksum = PlatformChecksum(
-      "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
-      "c8b2a93842e48d014f7e6efe10ae96cb3892b129",
-      "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f",
-      "f59b3dfdb9b1b8bbb61abedd7c8cf3fc47c21f5f");
+  const std::string network_stats_checksum =
+      PlatformChecksum("f7c2158761a531dd2804d13da0480033faa7be12",
+                       "8b5e3c8247dce48cb33923eaa1a502ca91429d5e",
+                       "f7c2158761a531dd2804d13da0480033faa7be12",
+                       "f7c2158761a531dd2804d13da0480033faa7be12");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
@@ skipping 12 matching lines @@
 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(
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e",
       "6237dd113ad80d7764fe4c90b55b2ec035eae64e");
 
-  const std::string network_stats_checksum = PlatformChecksum(
-      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
-      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
-      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8",
-      "d8379381d5a619f0616bb3c0a8a9eea1704a8ab8");
+  const std::string network_stats_checksum =
+      PlatformChecksum("0869a450a819b14bf2a91eb6f3629a3421d17606",
+                       "0869a450a819b14bf2a91eb6f3629a3421d17606",
+                       "0869a450a819b14bf2a91eb6f3629a3421d17606",
+                       "0869a450a819b14bf2a91eb6f3629a3421d17606");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
@@ skipping 1140 matching lines @@
   // Pull out data once.
   AudioFrame output;
   bool muted;
   ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
 
   EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
             neteq_->LastDecodedTimestamps());
 }
 
 }  // namespace webrtc