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1 /*
2  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3  *
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
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #include "api/audio_codecs/opus/audio_encoder_opus.h"
12 #include "api/test/metrics/global_metrics_logger_and_exporter.h"
13 #include "api/test/metrics/metric.h"
14 #include "modules/audio_coding/neteq/tools/audio_loop.h"
15 #include "rtc_base/time_utils.h"
16 #include "test/gtest.h"
17 #include "test/testsupport/file_utils.h"
18 
19 namespace webrtc {
20 namespace {
21 
22 using ::webrtc::test::GetGlobalMetricsLogger;
23 using ::webrtc::test::ImprovementDirection;
24 using ::webrtc::test::Unit;
25 
RunComplexityTest(const AudioEncoderOpusConfig & config)26 int64_t RunComplexityTest(const AudioEncoderOpusConfig& config) {
27   // Create encoder.
28   constexpr int payload_type = 17;
29   const auto encoder = AudioEncoderOpus::MakeAudioEncoder(config, payload_type);
30   // Open speech file.
31   const std::string kInputFileName =
32       webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
33   test::AudioLoop audio_loop;
34   constexpr int kSampleRateHz = 48000;
35   EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
36   constexpr size_t kMaxLoopLengthSamples =
37       kSampleRateHz * 10;  // 10 second loop.
38   constexpr size_t kInputBlockSizeSamples =
39       10 * kSampleRateHz / 1000;  // 60 ms.
40   EXPECT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
41                               kInputBlockSizeSamples));
42   // Encode.
43   const int64_t start_time_ms = rtc::TimeMillis();
44   AudioEncoder::EncodedInfo info;
45   rtc::Buffer encoded(500);
46   uint32_t rtp_timestamp = 0u;
47   for (size_t i = 0; i < 10000; ++i) {
48     encoded.Clear();
49     info = encoder->Encode(rtp_timestamp, audio_loop.GetNextBlock(), &encoded);
50     rtp_timestamp += kInputBlockSizeSamples;
51   }
52   return rtc::TimeMillis() - start_time_ms;
53 }
54 
55 // This test encodes an audio file using Opus twice with different bitrates
56 // (~11 kbps and 15.5 kbps). The runtime for each is measured, and the ratio
57 // between the two is calculated and tracked. This test explicitly sets the
58 // low_rate_complexity to 9. When running on desktop platforms, this is the same
59 // as the regular complexity, and the expectation is that the resulting ratio
60 // should be less than 100% (since the encoder runs faster at lower bitrates,
61 // given a fixed complexity setting). On the other hand, when running on
62 // mobiles, the regular complexity is 5, and we expect the resulting ratio to
63 // be higher, since we have explicitly asked for a higher complexity setting at
64 // the lower rate.
TEST(AudioEncoderOpusComplexityAdaptationTest,Adaptation_On)65 TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_On) {
66   // Create config.
67   AudioEncoderOpusConfig config;
68   // The limit -- including the hysteresis window -- at which the complexity
69   // shuold be increased.
70   config.bitrate_bps = 11000 - 1;
71   config.low_rate_complexity = 9;
72   int64_t runtime_10999bps = RunComplexityTest(config);
73 
74   config.bitrate_bps = 15500;
75   int64_t runtime_15500bps = RunComplexityTest(config);
76 
77   GetGlobalMetricsLogger()->LogSingleValueMetric(
78       "opus_encoding_complexity_ratio", "adaptation_on",
79       100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
80       ImprovementDirection::kNeitherIsBetter);
81 }
82 
83 // This test is identical to the one above, but without the complexity
84 // adaptation enabled (neither on desktop, nor on mobile). The expectation is
85 // that the resulting ratio is less than 100% at all times.
TEST(AudioEncoderOpusComplexityAdaptationTest,Adaptation_Off)86 TEST(AudioEncoderOpusComplexityAdaptationTest, Adaptation_Off) {
87   // Create config.
88   AudioEncoderOpusConfig config;
89   // The limit -- including the hysteresis window -- at which the complexity
90   // shuold be increased (but not in this test since complexity adaptation is
91   // disabled).
92   config.bitrate_bps = 11000 - 1;
93   int64_t runtime_10999bps = RunComplexityTest(config);
94 
95   config.bitrate_bps = 15500;
96   int64_t runtime_15500bps = RunComplexityTest(config);
97 
98   GetGlobalMetricsLogger()->LogSingleValueMetric(
99       "opus_encoding_complexity_ratio", "adaptation_off",
100       100.0 * runtime_10999bps / runtime_15500bps, Unit::kPercent,
101       ImprovementDirection::kNeitherIsBetter);
102 }
103 
104 }  // namespace
105 }  // namespace webrtc
106