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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 |
11 #include "webrtc/base/format_macros.h" | 11 #include "webrtc/base/format_macros.h" |
| 12 #include "webrtc/base/timeutils.h" |
12 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" | 13 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h" |
13 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" | 14 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" |
14 #include "webrtc/test/gtest.h" | 15 #include "webrtc/test/gtest.h" |
15 #include "webrtc/test/testsupport/fileutils.h" | 16 #include "webrtc/test/testsupport/fileutils.h" |
16 #include "webrtc/test/testsupport/perf_test.h" | 17 #include "webrtc/test/testsupport/perf_test.h" |
17 #include "webrtc/system_wrappers/include/clock.h" | |
18 | 18 |
19 namespace webrtc { | 19 namespace webrtc { |
20 | 20 |
21 namespace { | 21 namespace { |
22 int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) { | 22 int64_t RunComplexityTest(const AudioEncoderOpus::Config& config) { |
23 // Create encoder. | 23 // Create encoder. |
24 AudioEncoderOpus encoder(config); | 24 AudioEncoderOpus encoder(config); |
25 // Open speech file. | 25 // Open speech file. |
26 const std::string kInputFileName = | 26 const std::string kInputFileName = |
27 webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); | 27 webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); |
28 test::AudioLoop audio_loop; | 28 test::AudioLoop audio_loop; |
29 constexpr int kSampleRateHz = 48000; | 29 constexpr int kSampleRateHz = 48000; |
30 EXPECT_EQ(kSampleRateHz, encoder.SampleRateHz()); | 30 EXPECT_EQ(kSampleRateHz, encoder.SampleRateHz()); |
31 constexpr size_t kMaxLoopLengthSamples = | 31 constexpr size_t kMaxLoopLengthSamples = |
32 kSampleRateHz * 10; // 10 second loop. | 32 kSampleRateHz * 10; // 10 second loop. |
33 constexpr size_t kInputBlockSizeSamples = | 33 constexpr size_t kInputBlockSizeSamples = |
34 10 * kSampleRateHz / 1000; // 60 ms. | 34 10 * kSampleRateHz / 1000; // 60 ms. |
35 EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, | 35 EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, |
36 kInputBlockSizeSamples)); | 36 kInputBlockSizeSamples)); |
37 // Encode. | 37 // Encode. |
38 webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock(); | 38 const int64_t start_time_ms = rtc::TimeMillis(); |
39 const int64_t start_time_ms = clock->TimeInMilliseconds(); | |
40 AudioEncoder::EncodedInfo info; | 39 AudioEncoder::EncodedInfo info; |
41 rtc::Buffer encoded(500); | 40 rtc::Buffer encoded(500); |
42 uint32_t rtp_timestamp = 0u; | 41 uint32_t rtp_timestamp = 0u; |
43 for (size_t i = 0; i < 10000; ++i) { | 42 for (size_t i = 0; i < 10000; ++i) { |
44 encoded.Clear(); | 43 encoded.Clear(); |
45 info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded); | 44 info = encoder.Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded); |
46 rtp_timestamp += kInputBlockSizeSamples; | 45 rtp_timestamp += kInputBlockSizeSamples; |
47 } | 46 } |
48 return clock->TimeInMilliseconds() - start_time_ms; | 47 return rtc::TimeMillis() - start_time_ms; |
49 } | 48 } |
50 } // namespace | 49 } // namespace |
51 | 50 |
52 // This test encodes an audio file using Opus twice with different bitrates | 51 // This test encodes an audio file using Opus twice with different bitrates |
53 // (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio | 52 // (~11 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 | 53 // 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 | 54 // 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 | 55 // 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, | 56 // 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 | 57 // given a fixed complexity setting). On the other hand, when running on |
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89 int64_t runtime_10999bps = RunComplexityTest(config); | 88 int64_t runtime_10999bps = RunComplexityTest(config); |
90 | 89 |
91 config.bitrate_bps = rtc::Optional<int>(15500); | 90 config.bitrate_bps = rtc::Optional<int>(15500); |
92 int64_t runtime_15500bps = RunComplexityTest(config); | 91 int64_t runtime_15500bps = RunComplexityTest(config); |
93 | 92 |
94 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off", | 93 test::PrintResult("opus_encoding_complexity_ratio", "", "adaptation_off", |
95 100.0 * runtime_10999bps / runtime_15500bps, "percent", | 94 100.0 * runtime_10999bps / runtime_15500bps, "percent", |
96 true); | 95 true); |
97 } | 96 } |
98 } // namespace webrtc | 97 } // namespace webrtc |
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