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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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405 void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) { | 405 void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) { |
406 rtp_source_.reset(test::RtpFileSource::Create(rtp_file)); | 406 rtp_source_.reset(test::RtpFileSource::Create(rtp_file)); |
407 } | 407 } |
408 | 408 |
409 void NetEqDecodingTest::Process(size_t* out_len) { | 409 void NetEqDecodingTest::Process(size_t* out_len) { |
410 // Check if time to receive. | 410 // Check if time to receive. |
411 while (packet_ && sim_clock_ >= packet_->time_ms()) { | 411 while (packet_ && sim_clock_ >= packet_->time_ms()) { |
412 if (packet_->payload_length_bytes() > 0) { | 412 if (packet_->payload_length_bytes() > 0) { |
413 WebRtcRTPHeader rtp_header; | 413 WebRtcRTPHeader rtp_header; |
414 packet_->ConvertHeader(&rtp_header); | 414 packet_->ConvertHeader(&rtp_header); |
| 415 #ifndef WEBRTC_CODEC_ISAC |
| 416 // Ignore payload type 104 (iSAC-swb) if ISAC is not supported. |
| 417 if (rtp_header.header.payloadType != 104) |
| 418 #endif |
415 ASSERT_EQ(0, neteq_->InsertPacket( | 419 ASSERT_EQ(0, neteq_->InsertPacket( |
416 rtp_header, | 420 rtp_header, |
417 rtc::ArrayView<const uint8_t>( | 421 rtc::ArrayView<const uint8_t>( |
418 packet_->payload(), packet_->payload_length_bytes()), | 422 packet_->payload(), packet_->payload_length_bytes()), |
419 static_cast<uint32_t>(packet_->time_ms() * | 423 static_cast<uint32_t>(packet_->time_ms() * |
420 (output_sample_rate_ / 1000)))); | 424 (output_sample_rate_ / 1000)))); |
421 } | 425 } |
422 // Get next packet. | 426 // Get next packet. |
423 packet_.reset(rtp_source_->NextPacket()); | 427 packet_.reset(rtp_source_->NextPacket()); |
424 } | 428 } |
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508 size_t* payload_len) { | 512 size_t* payload_len) { |
509 rtp_info->header.sequenceNumber = frame_index; | 513 rtp_info->header.sequenceNumber = frame_index; |
510 rtp_info->header.timestamp = timestamp; | 514 rtp_info->header.timestamp = timestamp; |
511 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC. | 515 rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC. |
512 rtp_info->header.payloadType = 98; // WB CNG. | 516 rtp_info->header.payloadType = 98; // WB CNG. |
513 rtp_info->header.markerBit = 0; | 517 rtp_info->header.markerBit = 0; |
514 payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen. | 518 payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen. |
515 *payload_len = 1; // Only noise level, no spectral parameters. | 519 *payload_len = 1; // Only noise level, no spectral parameters. |
516 } | 520 } |
517 | 521 |
518 #if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \ | 522 #if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ |
519 defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ | 523 (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ |
520 (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ | 524 defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) && \ |
521 defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) | 525 !defined(WEBRTC_ARCH_ARM64) |
522 #define MAYBE_TestBitExactness TestBitExactness | 526 #define MAYBE_TestBitExactness TestBitExactness |
523 #else | 527 #else |
524 #define MAYBE_TestBitExactness DISABLED_TestBitExactness | 528 #define MAYBE_TestBitExactness DISABLED_TestBitExactness |
525 #endif | 529 #endif |
526 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { | 530 TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { |
527 const std::string input_rtp_file = | 531 const std::string input_rtp_file = |
528 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); | 532 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); |
529 // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm | 533 // 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 | 534 // are identical. The latter could have been removed, but if clients still |
531 // have a copy of the file, the test will fail. | 535 // have a copy of the file, the test will fail. |
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922 TEST_F(NetEqDecodingTest, UnknownPayloadType) { | 926 TEST_F(NetEqDecodingTest, UnknownPayloadType) { |
923 const size_t kPayloadBytes = 100; | 927 const size_t kPayloadBytes = 100; |
924 uint8_t payload[kPayloadBytes] = {0}; | 928 uint8_t payload[kPayloadBytes] = {0}; |
925 WebRtcRTPHeader rtp_info; | 929 WebRtcRTPHeader rtp_info; |
926 PopulateRtpInfo(0, 0, &rtp_info); | 930 PopulateRtpInfo(0, 0, &rtp_info); |
927 rtp_info.header.payloadType = 1; // Not registered as a decoder. | 931 rtp_info.header.payloadType = 1; // Not registered as a decoder. |
928 EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0)); | 932 EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0)); |
929 EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError()); | 933 EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError()); |
930 } | 934 } |
931 | 935 |
932 #if defined(WEBRTC_ANDROID) | 936 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) |
| 937 #define MAYBE_DecoderError DecoderError |
| 938 #else |
933 #define MAYBE_DecoderError DISABLED_DecoderError | 939 #define MAYBE_DecoderError DISABLED_DecoderError |
934 #else | |
935 #define MAYBE_DecoderError DecoderError | |
936 #endif | 940 #endif |
937 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) | 941 |
938 TEST_F(NetEqDecodingTest, MAYBE_DecoderError) { | 942 TEST_F(NetEqDecodingTest, MAYBE_DecoderError) { |
939 const size_t kPayloadBytes = 100; | 943 const size_t kPayloadBytes = 100; |
940 uint8_t payload[kPayloadBytes] = {0}; | 944 uint8_t payload[kPayloadBytes] = {0}; |
941 WebRtcRTPHeader rtp_info; | 945 WebRtcRTPHeader rtp_info; |
942 PopulateRtpInfo(0, 0, &rtp_info); | 946 PopulateRtpInfo(0, 0, &rtp_info); |
943 rtp_info.header.payloadType = 103; // iSAC, but the payload is invalid. | 947 rtp_info.header.payloadType = 103; // iSAC, but the payload is invalid. |
944 EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0)); | 948 EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0)); |
945 NetEqOutputType type; | 949 NetEqOutputType type; |
946 // Set all of |out_data_| to 1, and verify that it was set to 0 by the call | 950 // Set all of |out_data_| to 1, and verify that it was set to 0 by the call |
947 // to GetAudio. | 951 // to GetAudio. |
948 for (size_t i = 0; i < kMaxBlockSize; ++i) { | 952 for (size_t i = 0; i < kMaxBlockSize; ++i) { |
949 out_data_[i] = 1; | 953 out_data_[i] = 1; |
950 } | 954 } |
951 size_t num_channels; | 955 size_t num_channels; |
952 size_t samples_per_channel; | 956 size_t samples_per_channel; |
953 EXPECT_EQ(NetEq::kFail, | 957 EXPECT_EQ(NetEq::kFail, |
954 neteq_->GetAudio(kMaxBlockSize, out_data_, | 958 neteq_->GetAudio(kMaxBlockSize, out_data_, |
955 &samples_per_channel, &num_channels, &type)); | 959 &samples_per_channel, &num_channels, &type)); |
956 // Verify that there is a decoder error to check. | 960 // Verify that there is a decoder error to check. |
957 EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError()); | 961 EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError()); |
958 // Code 6730 is an iSAC error code. | 962 |
959 EXPECT_EQ(6730, neteq_->LastDecoderError()); | 963 enum NetEqDecoderError { |
| 964 ISAC_LENGTH_MISMATCH = 6730, |
| 965 ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH = 6640 |
| 966 }; |
| 967 #if defined(WEBRTC_CODEC_ISAC) |
| 968 EXPECT_EQ(ISAC_LENGTH_MISMATCH, neteq_->LastDecoderError()); |
| 969 #elif defined(WEBRTC_CODEC_ISACFX) |
| 970 EXPECT_EQ(ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH, neteq_->LastDecoderError()); |
| 971 #endif |
960 // Verify that the first 160 samples are set to 0, and that the remaining | 972 // Verify that the first 160 samples are set to 0, and that the remaining |
961 // samples are left unmodified. | 973 // samples are left unmodified. |
962 static const int kExpectedOutputLength = 160; // 10 ms at 16 kHz sample rate. | 974 static const int kExpectedOutputLength = 160; // 10 ms at 16 kHz sample rate. |
963 for (int i = 0; i < kExpectedOutputLength; ++i) { | 975 for (int i = 0; i < kExpectedOutputLength; ++i) { |
964 std::ostringstream ss; | 976 std::ostringstream ss; |
965 ss << "i = " << i; | 977 ss << "i = " << i; |
966 SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. | 978 SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. |
967 EXPECT_EQ(0, out_data_[i]); | 979 EXPECT_EQ(0, out_data_[i]); |
968 } | 980 } |
969 for (size_t i = kExpectedOutputLength; i < kMaxBlockSize; ++i) { | 981 for (size_t i = kExpectedOutputLength; i < kMaxBlockSize; ++i) { |
970 std::ostringstream ss; | 982 std::ostringstream ss; |
971 ss << "i = " << i; | 983 ss << "i = " << i; |
972 SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. | 984 SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. |
973 EXPECT_EQ(1, out_data_[i]); | 985 EXPECT_EQ(1, out_data_[i]); |
974 } | 986 } |
975 } | 987 } |
976 #endif | |
977 | 988 |
978 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) { | 989 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) { |
979 NetEqOutputType type; | 990 NetEqOutputType type; |
980 // Set all of |out_data_| to 1, and verify that it was set to 0 by the call | 991 // Set all of |out_data_| to 1, and verify that it was set to 0 by the call |
981 // to GetAudio. | 992 // to GetAudio. |
982 for (size_t i = 0; i < kMaxBlockSize; ++i) { | 993 for (size_t i = 0; i < kMaxBlockSize; ++i) { |
983 out_data_[i] = 1; | 994 out_data_[i] = 1; |
984 } | 995 } |
985 size_t num_channels; | 996 size_t num_channels; |
986 size_t samples_per_channel; | 997 size_t samples_per_channel; |
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1165 CheckBgn(32000); | 1176 CheckBgn(32000); |
1166 } | 1177 } |
1167 | 1178 |
1168 TEST_F(NetEqBgnTestFade, RunTest) { | 1179 TEST_F(NetEqBgnTestFade, RunTest) { |
1169 CheckBgn(8000); | 1180 CheckBgn(8000); |
1170 CheckBgn(16000); | 1181 CheckBgn(16000); |
1171 CheckBgn(32000); | 1182 CheckBgn(32000); |
1172 } | 1183 } |
1173 | 1184 |
1174 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) | 1185 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX) |
1175 TEST_F(NetEqDecodingTest, SyncPacketInsert) { | 1186 #define MAYBE_SyncPacketInsert SyncPacketInsert |
| 1187 #else |
| 1188 #define MAYBE_SyncPacketInsert DISABLED_SyncPacketInsert |
| 1189 #endif |
| 1190 TEST_F(NetEqDecodingTest, MAYBE_SyncPacketInsert) { |
1176 WebRtcRTPHeader rtp_info; | 1191 WebRtcRTPHeader rtp_info; |
1177 uint32_t receive_timestamp = 0; | 1192 uint32_t receive_timestamp = 0; |
1178 // For the readability use the following payloads instead of the defaults of | 1193 // For the readability use the following payloads instead of the defaults of |
1179 // this test. | 1194 // this test. |
1180 uint8_t kPcm16WbPayloadType = 1; | 1195 uint8_t kPcm16WbPayloadType = 1; |
1181 uint8_t kCngNbPayloadType = 2; | 1196 uint8_t kCngNbPayloadType = 2; |
1182 uint8_t kCngWbPayloadType = 3; | 1197 uint8_t kCngWbPayloadType = 3; |
1183 uint8_t kCngSwb32PayloadType = 4; | 1198 uint8_t kCngSwb32PayloadType = 4; |
1184 uint8_t kCngSwb48PayloadType = 5; | 1199 uint8_t kCngSwb48PayloadType = 5; |
1185 uint8_t kAvtPayloadType = 6; | 1200 uint8_t kAvtPayloadType = 6; |
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1244 EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); | 1259 EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); |
1245 | 1260 |
1246 // Change of SSRC is not allowed with a sync packet. | 1261 // Change of SSRC is not allowed with a sync packet. |
1247 rtp_info.header.payloadType = kPcm16WbPayloadType; | 1262 rtp_info.header.payloadType = kPcm16WbPayloadType; |
1248 ++rtp_info.header.ssrc; | 1263 ++rtp_info.header.ssrc; |
1249 EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); | 1264 EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); |
1250 | 1265 |
1251 --rtp_info.header.ssrc; | 1266 --rtp_info.header.ssrc; |
1252 EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); | 1267 EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp)); |
1253 } | 1268 } |
1254 #endif | |
1255 | 1269 |
1256 // First insert several noise like packets, then sync-packets. Decoding all | 1270 // First insert several noise like packets, then sync-packets. Decoding all |
1257 // packets should not produce error, statistics should not show any packet loss | 1271 // packets should not produce error, statistics should not show any packet loss |
1258 // and sync-packets should decode to zero. | 1272 // and sync-packets should decode to zero. |
1259 // TODO(turajs) we will have a better test if we have a referece NetEq, and | 1273 // TODO(turajs) we will have a better test if we have a referece NetEq, and |
1260 // when Sync packets are inserted in "test" NetEq we insert all-zero payload | 1274 // when Sync packets are inserted in "test" NetEq we insert all-zero payload |
1261 // in reference NetEq and compare the output of those two. | 1275 // in reference NetEq and compare the output of those two. |
1262 TEST_F(NetEqDecodingTest, SyncPacketDecode) { | 1276 TEST_F(NetEqDecodingTest, SyncPacketDecode) { |
1263 WebRtcRTPHeader rtp_info; | 1277 WebRtcRTPHeader rtp_info; |
1264 PopulateRtpInfo(0, 0, &rtp_info); | 1278 PopulateRtpInfo(0, 0, &rtp_info); |
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1639 // Pull audio once. | 1653 // Pull audio once. |
1640 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len, | 1654 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len, |
1641 &num_channels, &type)); | 1655 &num_channels, &type)); |
1642 ASSERT_EQ(kBlockSize16kHz, out_len); | 1656 ASSERT_EQ(kBlockSize16kHz, out_len); |
1643 } | 1657 } |
1644 // Verify speech output. | 1658 // Verify speech output. |
1645 EXPECT_EQ(kOutputNormal, type); | 1659 EXPECT_EQ(kOutputNormal, type); |
1646 } | 1660 } |
1647 | 1661 |
1648 } // namespace webrtc | 1662 } // namespace webrtc |
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