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Side by Side Diff: webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc

Issue 1172163004: Reformat existing code. There should be no functional effects. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 5 years, 6 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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73 value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer], 73 value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
74 bit_stream); 74 bit_stream);
75 } 75 }
76 clocks = clock() - clocks; 76 clocks = clock() - clocks;
77 EXPECT_GT(value, 0); 77 EXPECT_GT(value, 0);
78 assert(value <= max_bytes); 78 assert(value <= max_bytes);
79 *encoded_bytes = value; 79 *encoded_bytes = value;
80 return 1000.0 * clocks / CLOCKS_PER_SEC; 80 return 1000.0 * clocks / CLOCKS_PER_SEC;
81 } 81 }
82 82
83 float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream, int encoded_bytes, 83 float IsacSpeedTest::DecodeABlock(const uint8_t* bit_stream,
84 int encoded_bytes,
84 int16_t* out_data) { 85 int16_t* out_data) {
85 int value; 86 int value;
86 int16_t audio_type; 87 int16_t audio_type;
87 clock_t clocks = clock(); 88 clock_t clocks = clock();
88 value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, 89 value = WebRtcIsacfix_Decode(ISACFIX_main_inst_, bit_stream, encoded_bytes,
89 bit_stream, 90 out_data, &audio_type);
90 encoded_bytes, out_data, &audio_type);
91 clocks = clock() - clocks; 91 clocks = clock() - clocks;
92 EXPECT_EQ(output_length_sample_, value); 92 EXPECT_EQ(output_length_sample_, value);
93 return 1000.0 * clocks / CLOCKS_PER_SEC; 93 return 1000.0 * clocks / CLOCKS_PER_SEC;
94 } 94 }
95 95
96 TEST_P(IsacSpeedTest, IsacEncodeDecodeTest) { 96 TEST_P(IsacSpeedTest, IsacEncodeDecodeTest) {
97 size_t kDurationSec = 400; // Test audio length in second. 97 size_t kDurationSec = 400; // Test audio length in second.
98 EncodeDecode(kDurationSec); 98 EncodeDecode(kDurationSec);
99 } 99 }
100 100
101 const coding_param param_set[] = 101 const coding_param param_set[] =
102 {::std::tr1::make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"), 102 {::std::tr1::make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"),
103 string("pcm"), true)}; 103 string("pcm"), true)};
104 104
105 INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest, 105 INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest,
106 ::testing::ValuesIn(param_set)); 106 ::testing::ValuesIn(param_set));
107 107
108 } // namespace webrtc 108 } // namespace webrtc
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