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1 /* | 1 /* |
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2016 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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42 uint32_t rtp_timestamp = 0u; | 42 uint32_t rtp_timestamp = 0u; |
43 for (size_t i = 0; i < 10000; ++i) { | 43 for (size_t i = 0; i < 10000; ++i) { |
44 encoded.Clear(); | 44 encoded.Clear(); |
45 info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded); | 45 info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded); |
46 rtp_timestamp += kInputBlockSizeSamples; | 46 rtp_timestamp += kInputBlockSizeSamples; |
47 } | 47 } |
48 return clock->TimeInMilliseconds() - start_time_ms; | 48 return clock->TimeInMilliseconds() - start_time_ms; |
49 } | 49 } |
50 } // namespace | 50 } // namespace |
51 | 51 |
| 52 #if defined(WEBRTC_ANDROID) |
| 53 #define MAYBE_AudioEncoderOpusComplexityAdaptationTest \ |
| 54 DISABLED_AudioEncoderOpusComplexityAdaptationTest |
| 55 #else |
| 56 #define MAYBE_AudioEncoderOpusComplexityAdaptationTest \ |
| 57 AudioEncoderOpusComplexityAdaptationTest |
| 58 #endif |
| 59 |
52 // This test encodes an audio file using Opus twice with different bitrates | 60 // This test encodes an audio file using Opus twice with different bitrates |
53 // (12.5 kbps and 15.5 kbps). The runtime for each is measured, and the ratio | 61 // (12.5 kbps and 15.5 kbps). The runtime for each is measured, and the ratio |
54 // between the two is calculated and tracked. This test explicitly sets the | 62 // between the two is calculated and tracked. This test explicitly sets the |
55 // low_rate_complexity to 9. When running on desktop platforms, this is the same | 63 // low_rate_complexity to 9. When running on desktop platforms, this is the same |
56 // as the regular complexity, and the expectation is that the resulting ratio | 64 // as the regular complexity, and the expectation is that the resulting ratio |
57 // should be less than 100% (since the encoder runs faster at lower bitrates, | 65 // should be less than 100% (since the encoder runs faster at lower bitrates, |
58 // given a fixed complexity setting). On the other hand, when running on | 66 // given a fixed complexity setting). On the other hand, when running on |
59 // mobiles, the regular complexity is 5, and we expect the resulting ratio to | 67 // mobiles, the regular complexity is 5, and we expect the resulting ratio to |
60 // be higher, since we have explicitly asked for a higher complexity setting at | 68 // be higher, since we have explicitly asked for a higher complexity setting at |
61 // the lower rate. | 69 // the lower rate. |
62 TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) { | 70 TEST(MAYBE_AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) { |
63 // Create config. | 71 // Create config. |
64 AudioEncoderOpus::Config config; | 72 AudioEncoderOpus::Config config; |
65 config.bitrate_bps = rtc::Optional<int>(12500); | 73 config.bitrate_bps = rtc::Optional<int>(12500); |
66 config.low_rate_complexity = 9; | 74 config.low_rate_complexity = 9; |
67 int64_t runtime_12500bps = RunComplexityTest(config); | 75 int64_t runtime_12500bps = RunComplexityTest(config); |
68 | 76 |
69 config.bitrate_bps = rtc::Optional<int>(15500); | 77 config.bitrate_bps = rtc::Optional<int>(15500); |
70 int64_t runtime_15500bps = RunComplexityTest(config); | 78 int64_t runtime_15500bps = RunComplexityTest(config); |
71 | 79 |
72 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on", | 80 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on", |
73 100.0 * runtime_12500bps / runtime_15500bps, "percent", | 81 100.0 * runtime_12500bps / runtime_15500bps, "percent", |
74 true); | 82 true); |
75 } | 83 } |
76 | 84 |
77 // This test is identical to the one above, but without the complexity | 85 // This test is identical to the one above, but without the complexity |
78 // adaptation enabled (neither on desktop, nor on mobile). The expectation is | 86 // adaptation enabled (neither on desktop, nor on mobile). The expectation is |
79 // that the resulting ratio is less than 100% at all times. | 87 // that the resulting ratio is less than 100% at all times. |
80 TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) { | 88 TEST(MAYBE_AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) { |
81 // Create config. | 89 // Create config. |
82 AudioEncoderOpus::Config config; | 90 AudioEncoderOpus::Config config; |
83 config.bitrate_bps = rtc::Optional<int>(12500); | 91 config.bitrate_bps = rtc::Optional<int>(12500); |
84 int64_t runtime_12500bps = RunComplexityTest(config); | 92 int64_t runtime_12500bps = RunComplexityTest(config); |
85 | 93 |
86 config.bitrate_bps = rtc::Optional<int>(15500); | 94 config.bitrate_bps = rtc::Optional<int>(15500); |
87 int64_t runtime_15500bps = RunComplexityTest(config); | 95 int64_t runtime_15500bps = RunComplexityTest(config); |
88 | 96 |
89 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off", | 97 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off", |
90 100.0 * runtime_12500bps / runtime_15500bps, "", true); | 98 100.0 * runtime_12500bps / runtime_15500bps, "", true); |
91 } | 99 } |
92 } // namespace webrtc | 100 } // namespace webrtc |
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