Reland "Avoiding overflow in cross correlation in NetEq."

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 442 matching lines @@
     !defined(WEBRTC_ARCH_ARM64) && !defined(UNDEFINED_SANITIZER)
 #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(
-      "f587883b7c371ee8d87dbf1b0f07525af7d959b8",
-      "a349bd71dba548029b05d1d2a6dc7caafab9a856",
-      "f587883b7c371ee8d87dbf1b0f07525af7d959b8",
-      "08266b198e7686b3cd9330813e0d2cd72fc8fdc2");
+      "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
+      "d2a6b6ff54b340cf9f961c7f07768d86b3761073",
+      "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
+      "f9749813dbc3fb59dae761de518fec65b8407c5b");
 
   const std::string network_stats_checksum = PlatformChecksum(
       "2cf380a05ee07080bd72471e8ec7777a39644ec9",
-      "2853ab577fe571adfc7b18f77bbe58f1253d2019",
+      "01be67dc4c3b8e74743a45cbd8684c0535dec9ad",
       "2cf380a05ee07080bd72471e8ec7777a39644ec9",
       "2cf380a05ee07080bd72471e8ec7777a39644ec9");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
       "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
                    rtcp_stats_checksum,
                    FLAGS_gen_ref);
 }
 
 // Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820
 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) &&             \
     defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
     defined(WEBRTC_CODEC_OPUS) && !defined(UNDEFINED_SANITIZER)
 #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(
-      "c23004d91ffbe5e7a1f24620fc89b58c0426040f",
-      "c23004d91ffbe5e7a1f24620fc89b58c0426040f",
-      "c23004d91ffbe5e7a1f24620fc89b58c0426040f",
-      "c23004d91ffbe5e7a1f24620fc89b58c0426040f");
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
+      "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4");
 
   const std::string network_stats_checksum = PlatformChecksum(
-      "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
-      "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
-      "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
-      "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb");
+      "6eab76efbde753d4dde38983445ca16b4ce59b39",
+      "6eab76efbde753d4dde38983445ca16b4ce59b39",
+      "6eab76efbde753d4dde38983445ca16b4ce59b39",
+      "6eab76efbde753d4dde38983445ca16b4ce59b39");
 
   const std::string rtcp_stats_checksum = PlatformChecksum(
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
 
   DecodeAndCompare(input_rtp_file,
                    output_checksum,
                    network_stats_checksum,
@@ skipping 990 matching lines @@
     timestamp += kSamples;
 
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
     ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
   }
   // Verify speech output.
   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
 }
 }  // namespace webrtc