Alex Luebs | eeb2765 | 2017-11-20 11:13:56 -0800 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2017 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_decoder_opus.h" |
| 12 | #include "api/audio_codecs/opus/audio_encoder_opus.h" |
| 13 | #include "common_audio/include/audio_util.h" |
Alex Luebs | eeb2765 | 2017-11-20 11:13:56 -0800 | [diff] [blame] | 14 | #include "common_audio/window_generator.h" |
Alessio Bazzica | d4161a3 | 2018-08-31 10:41:37 +0200 | [diff] [blame] | 15 | #include "modules/audio_coding/codecs/opus/test/lapped_transform.h" |
Alex Luebs | eeb2765 | 2017-11-20 11:13:56 -0800 | [diff] [blame] | 16 | #include "modules/audio_coding/neteq/tools/audio_loop.h" |
| 17 | #include "test/field_trial.h" |
| 18 | #include "test/gtest.h" |
Steve Anton | 10542f2 | 2019-01-11 09:11:00 -0800 | [diff] [blame] | 19 | #include "test/testsupport/file_utils.h" |
Alex Luebs | eeb2765 | 2017-11-20 11:13:56 -0800 | [diff] [blame] | 20 | |
| 21 | namespace webrtc { |
| 22 | namespace { |
| 23 | |
| 24 | constexpr size_t kNumChannels = 1u; |
| 25 | constexpr int kSampleRateHz = 48000; |
| 26 | constexpr size_t kMaxLoopLengthSamples = kSampleRateHz * 50; // 50 seconds. |
| 27 | constexpr size_t kInputBlockSizeSamples = 10 * kSampleRateHz / 1000; // 10 ms |
| 28 | constexpr size_t kOutputBlockSizeSamples = 20 * kSampleRateHz / 1000; // 20 ms |
| 29 | constexpr size_t kFftSize = 1024; |
| 30 | constexpr size_t kNarrowbandSize = 4000 * kFftSize / kSampleRateHz; |
| 31 | constexpr float kKbdAlpha = 1.5f; |
| 32 | |
| 33 | class PowerRatioEstimator : public LappedTransform::Callback { |
| 34 | public: |
| 35 | PowerRatioEstimator() : low_pow_(0.f), high_pow_(0.f) { |
| 36 | WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_); |
| 37 | transform_.reset(new LappedTransform(kNumChannels, 0u, |
| 38 | kInputBlockSizeSamples, window_, |
| 39 | kFftSize, kFftSize / 2, this)); |
| 40 | } |
| 41 | |
| 42 | void ProcessBlock(float* data) { transform_->ProcessChunk(&data, nullptr); } |
| 43 | |
| 44 | float PowerRatio() { return high_pow_ / low_pow_; } |
| 45 | |
| 46 | protected: |
| 47 | void ProcessAudioBlock(const std::complex<float>* const* input, |
| 48 | size_t num_input_channels, |
| 49 | size_t num_freq_bins, |
| 50 | size_t num_output_channels, |
| 51 | std::complex<float>* const* output) override { |
| 52 | float low_pow = 0.f; |
| 53 | float high_pow = 0.f; |
| 54 | for (size_t i = 0u; i < num_input_channels; ++i) { |
| 55 | for (size_t j = 0u; j < kNarrowbandSize; ++j) { |
| 56 | float low_mag = std::abs(input[i][j]); |
| 57 | low_pow += low_mag * low_mag; |
| 58 | float high_mag = std::abs(input[i][j + kNarrowbandSize]); |
| 59 | high_pow += high_mag * high_mag; |
| 60 | } |
| 61 | } |
| 62 | low_pow_ += low_pow / (num_input_channels * kFftSize); |
| 63 | high_pow_ += high_pow / (num_input_channels * kFftSize); |
| 64 | } |
| 65 | |
| 66 | private: |
| 67 | std::unique_ptr<LappedTransform> transform_; |
| 68 | float window_[kFftSize]; |
| 69 | float low_pow_; |
| 70 | float high_pow_; |
| 71 | }; |
| 72 | |
| 73 | float EncodedPowerRatio(AudioEncoder* encoder, |
| 74 | AudioDecoder* decoder, |
| 75 | test::AudioLoop* audio_loop) { |
| 76 | // Encode and decode. |
| 77 | uint32_t rtp_timestamp = 0u; |
| 78 | constexpr size_t kBufferSize = 500; |
| 79 | rtc::Buffer encoded(kBufferSize); |
| 80 | std::vector<int16_t> decoded(kOutputBlockSizeSamples); |
| 81 | std::vector<float> decoded_float(kOutputBlockSizeSamples); |
| 82 | AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech; |
| 83 | PowerRatioEstimator power_ratio_estimator; |
| 84 | for (size_t i = 0; i < 1000; ++i) { |
| 85 | encoded.Clear(); |
| 86 | AudioEncoder::EncodedInfo encoder_info = |
| 87 | encoder->Encode(rtp_timestamp, audio_loop->GetNextBlock(), &encoded); |
| 88 | rtp_timestamp += kInputBlockSizeSamples; |
| 89 | if (encoded.size() > 0) { |
| 90 | int decoder_info = decoder->Decode( |
| 91 | encoded.data(), encoded.size(), kSampleRateHz, |
| 92 | decoded.size() * sizeof(decoded[0]), decoded.data(), &speech_type); |
| 93 | if (decoder_info > 0) { |
| 94 | S16ToFloat(decoded.data(), decoded.size(), decoded_float.data()); |
| 95 | power_ratio_estimator.ProcessBlock(decoded_float.data()); |
| 96 | } |
| 97 | } |
| 98 | } |
| 99 | return power_ratio_estimator.PowerRatio(); |
| 100 | } |
| 101 | |
| 102 | } // namespace |
| 103 | |
| 104 | TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) { |
| 105 | test::ScopedFieldTrials override_field_trials( |
| 106 | "WebRTC-AdjustOpusBandwidth/Enabled/"); |
| 107 | |
Gustaf Ullberg | b9fc650 | 2018-05-21 15:50:22 +0200 | [diff] [blame] | 108 | constexpr float kMaxNarrowbandRatio = 0.0035f; |
Alex Luebs | eeb2765 | 2017-11-20 11:13:56 -0800 | [diff] [blame] | 109 | constexpr float kMinWidebandRatio = 0.03f; |
| 110 | |
| 111 | // Create encoder. |
| 112 | AudioEncoderOpusConfig enc_config; |
Danil Chapovalov | b602123 | 2018-06-19 13:26:36 +0200 | [diff] [blame] | 113 | enc_config.bitrate_bps = absl::optional<int>(7999); |
Alex Luebs | eeb2765 | 2017-11-20 11:13:56 -0800 | [diff] [blame] | 114 | enc_config.num_channels = kNumChannels; |
| 115 | constexpr int payload_type = 17; |
| 116 | auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type); |
| 117 | |
| 118 | // Create decoder. |
| 119 | AudioDecoderOpus::Config dec_config; |
| 120 | dec_config.num_channels = kNumChannels; |
| 121 | auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config); |
| 122 | |
| 123 | // Open speech file. |
| 124 | const std::string kInputFileName = |
| 125 | webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm"); |
| 126 | test::AudioLoop audio_loop; |
| 127 | EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz()); |
| 128 | ASSERT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples, |
| 129 | kInputBlockSizeSamples)); |
| 130 | |
| 131 | EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop), |
| 132 | kMaxNarrowbandRatio); |
| 133 | |
| 134 | encoder->OnReceivedTargetAudioBitrate(9000); |
| 135 | EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop), |
| 136 | kMaxNarrowbandRatio); |
| 137 | |
| 138 | encoder->OnReceivedTargetAudioBitrate(9001); |
| 139 | EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop), |
| 140 | kMinWidebandRatio); |
| 141 | |
| 142 | encoder->OnReceivedTargetAudioBitrate(8000); |
| 143 | EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop), |
| 144 | kMinWidebandRatio); |
| 145 | |
| 146 | encoder->OnReceivedTargetAudioBitrate(12001); |
| 147 | EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop), |
| 148 | kMinWidebandRatio); |
| 149 | } |
| 150 | |
| 151 | } // namespace webrtc |