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Side by Side Diff: webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc

Issue 2389263003: Fix lint errors in audio_coding_module_unittest.cc (Closed)
Patch Set: Created 4 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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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1044 return format.name == "MockPCMu" ? std::move(mock_decoder_) 1044 return format.name == "MockPCMu" ? std::move(mock_decoder_)
1045 : fact_->MakeAudioDecoder(format); 1045 : fact_->MakeAudioDecoder(format);
1046 } 1046 }
1047 1047
1048 private: 1048 private:
1049 // Class intended to forward a call from a mock DecodeInternal to Decode on 1049 // Class intended to forward a call from a mock DecodeInternal to Decode on
1050 // the real decoder's Decode. DecodeInternal for the real decoder isn't 1050 // the real decoder's Decode. DecodeInternal for the real decoder isn't
1051 // public. 1051 // public.
1052 class DecodeForwarder { 1052 class DecodeForwarder {
1053 public: 1053 public:
1054 DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {} 1054 explicit DecodeForwarder(AudioDecoder* decoder) : decoder_(decoder) {}
1055 int Decode(const uint8_t* encoded, 1055 int Decode(const uint8_t* encoded,
1056 size_t encoded_len, 1056 size_t encoded_len,
1057 int sample_rate_hz, 1057 int sample_rate_hz,
1058 int16_t* decoded, 1058 int16_t* decoded,
1059 AudioDecoder::SpeechType* speech_type) { 1059 AudioDecoder::SpeechType* speech_type) {
1060 return decoder_->Decode(encoded, encoded_len, sample_rate_hz, 1060 return decoder_->Decode(encoded, encoded_len, sample_rate_hz,
1061 decoder_->PacketDuration(encoded, encoded_len) * 1061 decoder_->PacketDuration(encoded, encoded_len) *
1062 decoder_->Channels() * sizeof(int16_t), 1062 decoder_->Channels() * sizeof(int16_t),
1063 decoded, speech_type); 1063 decoded, speech_type);
1064 } 1064 }
(...skipping 478 matching lines...) Expand 10 before | Expand all | Expand 10 after
1543 std::unique_ptr<test::AcmSendTestOldApi> send_test_; 1543 std::unique_ptr<test::AcmSendTestOldApi> send_test_;
1544 std::unique_ptr<test::InputAudioFile> audio_source_; 1544 std::unique_ptr<test::InputAudioFile> audio_source_;
1545 }; 1545 };
1546 1546
1547 TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) { 1547 TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
1548 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); 1548 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
1549 #if defined(WEBRTC_ANDROID) 1549 #if defined(WEBRTC_ANDROID)
1550 Run(10000, 9288); 1550 Run(10000, 9288);
1551 #else 1551 #else
1552 Run(10000, 9024); 1552 Run(10000, 9024);
1553 #endif // WEBRTC_ANDROID 1553 #endif // WEBRTC_ANDROID
1554
1555 } 1554 }
1556 1555
1557 TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) { 1556 TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
1558 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); 1557 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
1559 #if defined(WEBRTC_ANDROID) 1558 #if defined(WEBRTC_ANDROID)
1560 Run(50000, 47960); 1559 Run(50000, 47960);
1561 #else 1560 #else
1562 Run(50000, 49544); 1561 Run(50000, 49544);
1563 #endif // WEBRTC_ANDROID 1562 #endif // WEBRTC_ANDROID
1564 } 1563 }
1565 1564
1566 // The result on the Android platforms is inconsistent for this test case. 1565 // The result on the Android platforms is inconsistent for this test case.
1567 // On android_rel the result is different from android and android arm64 rel. 1566 // On android_rel the result is different from android and android arm64 rel.
1568 #if defined(WEBRTC_ANDROID) 1567 #if defined(WEBRTC_ANDROID)
1569 #define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps 1568 #define MAYBE_Opus_48khz_20ms_100kbps DISABLED_Opus_48khz_20ms_100kbps
1570 #else 1569 #else
1571 #define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps 1570 #define MAYBE_Opus_48khz_20ms_100kbps Opus_48khz_20ms_100kbps
1572 #endif 1571 #endif
1573 TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) { 1572 TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
(...skipping 61 matching lines...) Expand 10 before | Expand all | Expand 10 after
1635 uint32_t sampling_freq_hz_; 1634 uint32_t sampling_freq_hz_;
1636 uint32_t frame_size_samples_; 1635 uint32_t frame_size_samples_;
1637 }; 1636 };
1638 1637
1639 TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) { 1638 TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps) {
1640 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); 1639 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
1641 #if defined(WEBRTC_ANDROID) 1640 #if defined(WEBRTC_ANDROID)
1642 Run(10000, 32200, 5176); 1641 Run(10000, 32200, 5176);
1643 #else 1642 #else
1644 Run(10000, 32200, 5456); 1643 Run(10000, 32200, 5456);
1645 #endif // WEBRTC_ANDROID 1644 #endif // WEBRTC_ANDROID
1646 } 1645 }
1647 1646
1648 TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) { 1647 TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps) {
1649 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); 1648 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
1650 #if defined(WEBRTC_ANDROID) 1649 #if defined(WEBRTC_ANDROID)
1651 Run(50000, 32200, 24768); 1650 Run(50000, 32200, 24768);
1652 #else 1651 #else
1653 Run(50000, 32200, 24848); 1652 Run(50000, 32200, 24848);
1654 #endif // WEBRTC_ANDROID 1653 #endif // WEBRTC_ANDROID
1655 } 1654 }
1656 1655
1657 TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) { 1656 TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps) {
1658 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960)); 1657 ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
1659 #if defined(WEBRTC_ANDROID) 1658 #if defined(WEBRTC_ANDROID)
1660 #if defined(WEBRTC_ARCH_ARM64) 1659 #if defined(WEBRTC_ARCH_ARM64)
1661 Run(100000, 32200, 51152); 1660 Run(100000, 32200, 51152);
1662 #else 1661 #else
1663 Run(100000, 32200, 51248); 1662 Run(100000, 32200, 51248);
1664 #endif // WEBRTC_ARCH_ARM64 1663 #endif // WEBRTC_ARCH_ARM64
1665 #else 1664 #else
1666 Run(100000, 32200, 50584); 1665 Run(100000, 32200, 50584);
1667 #endif // WEBRTC_ANDROID 1666 #endif // WEBRTC_ANDROID
1668 } 1667 }
1669 1668
1670 // These next 2 tests ensure that the SetBitRate function has no effect on PCM 1669 // These next 2 tests ensure that the SetBitRate function has no effect on PCM
1671 TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_8kbps) { 1670 TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
1672 ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); 1671 ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
1673 Run(8000, 64000, 64000); 1672 Run(8000, 64000, 64000);
1674 } 1673 }
1675 1674
1676 TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_32kbps) { 1675 TEST_F(AcmChangeBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
1677 ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80)); 1676 ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
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1834 Run(16000, 8000, 1000); 1833 Run(16000, 8000, 1000);
1835 } 1834 }
1836 1835
1837 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) { 1836 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
1838 Run(8000, 16000, 1000); 1837 Run(8000, 16000, 1000);
1839 } 1838 }
1840 1839
1841 #endif 1840 #endif
1842 1841
1843 } // namespace webrtc 1842 } // namespace webrtc
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