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

Issue 1484343003: NetEq: Add codec name and RTP timestamp rate to DecoderInfo (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase Created 5 years 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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859 return others; 859 return others;
860 #endif 860 #endif
861 } 861 }
862 862
863 protected: 863 protected:
864 struct ExternalDecoder { 864 struct ExternalDecoder {
865 int rtp_payload_type; 865 int rtp_payload_type;
866 AudioDecoder* external_decoder; 866 AudioDecoder* external_decoder;
867 int sample_rate_hz; 867 int sample_rate_hz;
868 int num_channels; 868 int num_channels;
869 std::string name;
869 }; 870 };
870 871
871 void Run(int output_freq_hz, 872 void Run(int output_freq_hz,
872 const std::string& checksum_ref, 873 const std::string& checksum_ref,
873 const std::vector<ExternalDecoder>& external_decoders) { 874 const std::vector<ExternalDecoder>& external_decoders) {
874 const std::string input_file_name = 875 const std::string input_file_name =
875 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"); 876 webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
876 rtc::scoped_ptr<test::RtpFileSource> packet_source( 877 rtc::scoped_ptr<test::RtpFileSource> packet_source(
877 test::RtpFileSource::Create(input_file_name)); 878 test::RtpFileSource::Create(input_file_name));
878 #ifdef WEBRTC_ANDROID 879 #ifdef WEBRTC_ANDROID
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894 895
895 test::AcmReceiveTestOldApi test( 896 test::AcmReceiveTestOldApi test(
896 packet_source.get(), 897 packet_source.get(),
897 &output, 898 &output,
898 output_freq_hz, 899 output_freq_hz,
899 test::AcmReceiveTestOldApi::kArbitraryChannels); 900 test::AcmReceiveTestOldApi::kArbitraryChannels);
900 ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs()); 901 ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
901 for (const auto& ed : external_decoders) { 902 for (const auto& ed : external_decoders) {
902 ASSERT_EQ(0, test.RegisterExternalReceiveCodec( 903 ASSERT_EQ(0, test.RegisterExternalReceiveCodec(
903 ed.rtp_payload_type, ed.external_decoder, 904 ed.rtp_payload_type, ed.external_decoder,
904 ed.sample_rate_hz, ed.num_channels)); 905 ed.sample_rate_hz, ed.num_channels, ed.name));
905 } 906 }
906 test.Run(); 907 test.Run();
907 908
908 std::string checksum_string = checksum.Finish(); 909 std::string checksum_string = checksum.Finish();
909 EXPECT_EQ(checksum_ref, checksum_string); 910 EXPECT_EQ(checksum_ref, checksum_string);
910 911
911 // Delete the output file. 912 // Delete the output file.
912 remove(output_file_name.c_str()); 913 remove(output_file_name.c_str());
913 } 914 }
914 }; 915 };
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1019 .Times(AtLeast(1)) 1020 .Times(AtLeast(1))
1020 .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc)); 1021 .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
1021 EXPECT_CALL(mock_decoder, PacketDuration(_, _)) 1022 EXPECT_CALL(mock_decoder, PacketDuration(_, _))
1022 .Times(AtLeast(1)) 1023 .Times(AtLeast(1))
1023 .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration)); 1024 .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration));
1024 ExternalDecoder ed; 1025 ExternalDecoder ed;
1025 ed.rtp_payload_type = 0; 1026 ed.rtp_payload_type = 0;
1026 ed.external_decoder = &mock_decoder; 1027 ed.external_decoder = &mock_decoder;
1027 ed.sample_rate_hz = 8000; 1028 ed.sample_rate_hz = 8000;
1028 ed.num_channels = 1; 1029 ed.num_channels = 1;
1030 ed.name = "MockPCMU";
1029 std::vector<ExternalDecoder> external_decoders; 1031 std::vector<ExternalDecoder> external_decoders;
1030 external_decoders.push_back(ed); 1032 external_decoders.push_back(ed);
1031 1033
1032 Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0", 1034 Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
1033 "0155665e93067c4e89256b944dd11999", 1035 "0155665e93067c4e89256b944dd11999",
1034 "4ee2730fa1daae755e8a8fd3abd779ec"), 1036 "4ee2730fa1daae755e8a8fd3abd779ec"),
1035 external_decoders); 1037 external_decoders);
1036 1038
1037 EXPECT_CALL(mock_decoder, Die()); 1039 EXPECT_CALL(mock_decoder, Die());
1038 } 1040 }
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1794 Run(16000, 8000, 1000); 1796 Run(16000, 8000, 1000);
1795 } 1797 }
1796 1798
1797 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) { 1799 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
1798 Run(8000, 16000, 1000); 1800 Run(8000, 16000, 1000);
1799 } 1801 }
1800 1802
1801 #endif 1803 #endif
1802 1804
1803 } // namespace webrtc 1805 } // namespace webrtc
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