OLD | NEW |
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 |
(...skipping 31 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
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 | |
60 // This test encodes an audio file using Opus twice with different bitrates | 52 // This test encodes an audio file using Opus twice with different bitrates |
61 // (12.5 kbps and 15.5 kbps). The runtime for each is measured, and the ratio | 53 // (12.5 kbps and 15.5 kbps). The runtime for each is measured, and the ratio |
62 // between the two is calculated and tracked. This test explicitly sets the | 54 // between the two is calculated and tracked. This test explicitly sets the |
63 // low_rate_complexity to 9. When running on desktop platforms, this is the same | 55 // low_rate_complexity to 9. When running on desktop platforms, this is the same |
64 // as the regular complexity, and the expectation is that the resulting ratio | 56 // as the regular complexity, and the expectation is that the resulting ratio |
65 // should be less than 100% (since the encoder runs faster at lower bitrates, | 57 // should be less than 100% (since the encoder runs faster at lower bitrates, |
66 // given a fixed complexity setting). On the other hand, when running on | 58 // given a fixed complexity setting). On the other hand, when running on |
67 // mobiles, the regular complexity is 5, and we expect the resulting ratio to | 59 // mobiles, the regular complexity is 5, and we expect the resulting ratio to |
68 // be higher, since we have explicitly asked for a higher complexity setting at | 60 // be higher, since we have explicitly asked for a higher complexity setting at |
69 // the lower rate. | 61 // the lower rate. |
70 TEST(MAYBE_AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) { | 62 TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOn) { |
71 // Create config. | 63 // Create config. |
72 AudioEncoderOpus::Config config; | 64 AudioEncoderOpus::Config config; |
73 config.bitrate_bps = rtc::Optional<int>(12500); | 65 config.bitrate_bps = rtc::Optional<int>(12500); |
74 config.low_rate_complexity = 9; | 66 config.low_rate_complexity = 9; |
75 int64_t runtime_12500bps = RunComplexityTest(config); | 67 int64_t runtime_12500bps = RunComplexityTest(config); |
76 | 68 |
77 config.bitrate_bps = rtc::Optional<int>(15500); | 69 config.bitrate_bps = rtc::Optional<int>(15500); |
78 int64_t runtime_15500bps = RunComplexityTest(config); | 70 int64_t runtime_15500bps = RunComplexityTest(config); |
79 | 71 |
80 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on", | 72 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_on", |
81 100.0 * runtime_12500bps / runtime_15500bps, "percent", | 73 100.0 * runtime_12500bps / runtime_15500bps, "percent", |
82 true); | 74 true); |
83 } | 75 } |
84 | 76 |
85 // This test is identical to the one above, but without the complexity | 77 // This test is identical to the one above, but without the complexity |
86 // adaptation enabled (neither on desktop, nor on mobile). The expectation is | 78 // adaptation enabled (neither on desktop, nor on mobile). The expectation is |
87 // that the resulting ratio is less than 100% at all times. | 79 // that the resulting ratio is less than 100% at all times. |
88 TEST(MAYBE_AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) { | 80 TEST(AudioEncoderOpusComplexityAdaptationTest, AdaptationOff) { |
89 // Create config. | 81 // Create config. |
90 AudioEncoderOpus::Config config; | 82 AudioEncoderOpus::Config config; |
91 config.bitrate_bps = rtc::Optional<int>(12500); | 83 config.bitrate_bps = rtc::Optional<int>(12500); |
92 int64_t runtime_12500bps = RunComplexityTest(config); | 84 int64_t runtime_12500bps = RunComplexityTest(config); |
93 | 85 |
94 config.bitrate_bps = rtc::Optional<int>(15500); | 86 config.bitrate_bps = rtc::Optional<int>(15500); |
95 int64_t runtime_15500bps = RunComplexityTest(config); | 87 int64_t runtime_15500bps = RunComplexityTest(config); |
96 | 88 |
97 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off", | 89 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off", |
98 100.0 * runtime_12500bps / runtime_15500bps, "", true); | 90 100.0 * runtime_12500bps / runtime_15500bps, "", true); |
99 } | 91 } |
100 } // namespace webrtc | 92 } // namespace webrtc |
OLD | NEW |