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Side by Side Diff: webrtc/modules/audio_coding/neteq/neteq_unittest.cc

Issue 1931933004: Reland "Avoiding overflow in cross correlation in NetEq." (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fixing bitexactness tests Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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453 !defined(WEBRTC_ARCH_ARM64) && !defined(UNDEFINED_SANITIZER) 453 !defined(WEBRTC_ARCH_ARM64) && !defined(UNDEFINED_SANITIZER)
454 #define MAYBE_TestBitExactness TestBitExactness 454 #define MAYBE_TestBitExactness TestBitExactness
455 #else 455 #else
456 #define MAYBE_TestBitExactness DISABLED_TestBitExactness 456 #define MAYBE_TestBitExactness DISABLED_TestBitExactness
457 #endif 457 #endif
458 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { 458 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
459 const std::string input_rtp_file = 459 const std::string input_rtp_file =
460 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); 460 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
461 461
462 const std::string output_checksum = PlatformChecksum( 462 const std::string output_checksum = PlatformChecksum(
463 "f587883b7c371ee8d87dbf1b0f07525af7d959b8", 463 "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
464 "a349bd71dba548029b05d1d2a6dc7caafab9a856", 464 "d2a6b6ff54b340cf9f961c7f07768d86b3761073",
465 "f587883b7c371ee8d87dbf1b0f07525af7d959b8", 465 "472ebe1126f41fdb6b5c63c87f625a52e7604e49",
466 "08266b198e7686b3cd9330813e0d2cd72fc8fdc2"); 466 "f9749813dbc3fb59dae761de518fec65b8407c5b");
467 467
468 const std::string network_stats_checksum = PlatformChecksum( 468 const std::string network_stats_checksum = PlatformChecksum(
469 "2cf380a05ee07080bd72471e8ec7777a39644ec9", 469 "2cf380a05ee07080bd72471e8ec7777a39644ec9",
470 "2853ab577fe571adfc7b18f77bbe58f1253d2019", 470 "01be67dc4c3b8e74743a45cbd8684c0535dec9ad",
471 "2cf380a05ee07080bd72471e8ec7777a39644ec9", 471 "2cf380a05ee07080bd72471e8ec7777a39644ec9",
472 "2cf380a05ee07080bd72471e8ec7777a39644ec9"); 472 "2cf380a05ee07080bd72471e8ec7777a39644ec9");
473 473
474 const std::string rtcp_stats_checksum = PlatformChecksum( 474 const std::string rtcp_stats_checksum = PlatformChecksum(
475 "b8880bf9fed2487efbddcb8d94b9937a29ae521d", 475 "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
476 "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4", 476 "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
477 "b8880bf9fed2487efbddcb8d94b9937a29ae521d", 477 "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
478 "b8880bf9fed2487efbddcb8d94b9937a29ae521d"); 478 "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
479 479
480 DecodeAndCompare(input_rtp_file, 480 DecodeAndCompare(input_rtp_file,
481 output_checksum, 481 output_checksum,
482 network_stats_checksum, 482 network_stats_checksum,
483 rtcp_stats_checksum, 483 rtcp_stats_checksum,
484 FLAGS_gen_ref); 484 FLAGS_gen_ref);
485 } 485 }
486 486
487 // Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820 487 // Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820
488 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \ 488 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
489 defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ 489 defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
490 defined(WEBRTC_CODEC_OPUS) && !defined(UNDEFINED_SANITIZER) 490 defined(WEBRTC_CODEC_OPUS) && !defined(UNDEFINED_SANITIZER)
491 #define MAYBE_TestOpusBitExactness TestOpusBitExactness 491 #define MAYBE_TestOpusBitExactness TestOpusBitExactness
492 #else 492 #else
493 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness 493 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
494 #endif 494 #endif
495 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { 495 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
496 const std::string input_rtp_file = 496 const std::string input_rtp_file =
497 webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); 497 webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
498 498
499 const std::string output_checksum = PlatformChecksum( 499 const std::string output_checksum = PlatformChecksum(
500 "c23004d91ffbe5e7a1f24620fc89b58c0426040f", 500 "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
501 "c23004d91ffbe5e7a1f24620fc89b58c0426040f", 501 "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
502 "c23004d91ffbe5e7a1f24620fc89b58c0426040f", 502 "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
503 "c23004d91ffbe5e7a1f24620fc89b58c0426040f"); 503 "19ad24b4a1eb7a9620e6da09f98c49aa5792ade4");
504 504
505 const std::string network_stats_checksum = PlatformChecksum( 505 const std::string network_stats_checksum = PlatformChecksum(
506 "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb", 506 "6eab76efbde753d4dde38983445ca16b4ce59b39",
507 "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb", 507 "6eab76efbde753d4dde38983445ca16b4ce59b39",
508 "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb", 508 "6eab76efbde753d4dde38983445ca16b4ce59b39",
509 "dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb"); 509 "6eab76efbde753d4dde38983445ca16b4ce59b39");
510 510
511 const std::string rtcp_stats_checksum = PlatformChecksum( 511 const std::string rtcp_stats_checksum = PlatformChecksum(
512 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0", 512 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
513 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0", 513 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
514 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0", 514 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
515 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0"); 515 "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
516 516
517 DecodeAndCompare(input_rtp_file, 517 DecodeAndCompare(input_rtp_file,
518 output_checksum, 518 output_checksum,
519 network_stats_checksum, 519 network_stats_checksum,
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1510 timestamp += kSamples; 1510 timestamp += kSamples;
1511 1511
1512 // Pull audio once. 1512 // Pull audio once.
1513 ASSERT_EQ(0, neteq_->GetAudio(&out_frame_)); 1513 ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
1514 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); 1514 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
1515 } 1515 }
1516 // Verify speech output. 1516 // Verify speech output.
1517 EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_); 1517 EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
1518 } 1518 }
1519 } // namespace webrtc 1519 } // namespace webrtc
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