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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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507 size_t* payload_len) { | 507 size_t* payload_len) { |
508 rtp_info->header.sequenceNumber = frame_index; | 508 rtp_info->header.sequenceNumber = frame_index; |
509 rtp_info->header.timestamp = timestamp; | 509 rtp_info->header.timestamp = timestamp; |
510 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC. | 510 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC. |
511 rtp_info->header.payloadType = 98; // WB CNG. | 511 rtp_info->header.payloadType = 98; // WB CNG. |
512 rtp_info->header.markerBit = 0; | 512 rtp_info->header.markerBit = 0; |
513 payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen. | 513 payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen. |
514 *payload_len = 1; // Only noise level, no spectral parameters. | 514 *payload_len = 1; // Only noise level, no spectral parameters. |
515 } | 515 } |
516 | 516 |
517 // Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820 | |
518 #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ | 517 #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ |
519 (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ | 518 (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ |
520 defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) && \ | 519 defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) && \ |
521 !defined(WEBRTC_ARCH_ARM64) && !defined(UNDEFINED_SANITIZER) | 520 !defined(WEBRTC_ARCH_ARM64) |
522 #define MAYBE_TestBitExactness TestBitExactness | 521 #define MAYBE_TestBitExactness TestBitExactness |
523 #else | 522 #else |
524 #define MAYBE_TestBitExactness DISABLED_TestBitExactness | 523 #define MAYBE_TestBitExactness DISABLED_TestBitExactness |
525 #endif | 524 #endif |
526 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { | 525 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { |
527 const std::string input_rtp_file = | 526 const std::string input_rtp_file = |
528 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); | 527 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); |
529 // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm | 528 // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm |
530 // are identical. The latter could have been removed, but if clients still | 529 // are identical. The latter could have been removed, but if clients still |
531 // have a copy of the file, the test will fail. | 530 // have a copy of the file, the test will fail. |
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546 if (FLAGS_gen_ref) { | 545 if (FLAGS_gen_ref) { |
547 DecodeAndCompare(input_rtp_file, "", "", ""); | 546 DecodeAndCompare(input_rtp_file, "", "", ""); |
548 } else { | 547 } else { |
549 DecodeAndCompare(input_rtp_file, | 548 DecodeAndCompare(input_rtp_file, |
550 input_ref_file, | 549 input_ref_file, |
551 network_stat_ref_file, | 550 network_stat_ref_file, |
552 rtcp_stat_ref_file); | 551 rtcp_stat_ref_file); |
553 } | 552 } |
554 } | 553 } |
555 | 554 |
556 // Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820 | |
557 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \ | 555 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \ |
558 defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ | 556 defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ |
559 defined(WEBRTC_CODEC_OPUS) && !defined(UNDEFINED_SANITIZER) | 557 defined(WEBRTC_CODEC_OPUS) |
560 #define MAYBE_TestOpusBitExactness TestOpusBitExactness | 558 #define MAYBE_TestOpusBitExactness TestOpusBitExactness |
561 #else | 559 #else |
562 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness | 560 #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness |
563 #endif | 561 #endif |
564 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { | 562 TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { |
565 const std::string input_rtp_file = | 563 const std::string input_rtp_file = |
566 webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); | 564 webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); |
567 const std::string input_ref_file = | 565 const std::string input_ref_file = |
568 // The pcm files were generated by using Opus v1.1.2 to decode the RTC | 566 // The pcm files were generated by using Opus v1.1.2 to decode the RTC |
569 // file generated by Opus v1.1 | 567 // file generated by Opus v1.1 |
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1575 timestamp += kSamples; | 1573 timestamp += kSamples; |
1576 | 1574 |
1577 // Pull audio once. | 1575 // Pull audio once. |
1578 ASSERT_EQ(0, neteq_->GetAudio(&out_frame_)); | 1576 ASSERT_EQ(0, neteq_->GetAudio(&out_frame_)); |
1579 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); | 1577 ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_); |
1580 } | 1578 } |
1581 // Verify speech output. | 1579 // Verify speech output. |
1582 EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_); | 1580 EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_); |
1583 } | 1581 } |
1584 } // namespace webrtc | 1582 } // namespace webrtc |
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