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/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "api/audio_codecs/opus/audio_decoder_opus.h"
#include "api/audio_codecs/opus/audio_encoder_opus.h"
#include "common_audio/include/audio_util.h"
#include "common_audio/window_generator.h"
#include "modules/audio_coding/codecs/opus/test/lapped_transform.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
namespace webrtc {
namespace {
constexpr size_t kNumChannels = 1u;
constexpr int kSampleRateHz = 48000;
constexpr size_t kMaxLoopLengthSamples = kSampleRateHz * 50; // 50 seconds.
constexpr size_t kInputBlockSizeSamples = 10 * kSampleRateHz / 1000; // 10 ms
constexpr size_t kOutputBlockSizeSamples = 20 * kSampleRateHz / 1000; // 20 ms
constexpr size_t kFftSize = 1024;
constexpr size_t kNarrowbandSize = 4000 * kFftSize / kSampleRateHz;
constexpr float kKbdAlpha = 1.5f;
class PowerRatioEstimator : public LappedTransform::Callback {
public:
PowerRatioEstimator() : low_pow_(0.f), high_pow_(0.f) {
WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_);
transform_.reset(new LappedTransform(kNumChannels, 0u,
kInputBlockSizeSamples, window_,
kFftSize, kFftSize / 2, this));
}
void ProcessBlock(float* data) { transform_->ProcessChunk(&data, nullptr); }
float PowerRatio() { return high_pow_ / low_pow_; }
protected:
void ProcessAudioBlock(const std::complex<float>* const* input,
size_t num_input_channels,
size_t num_freq_bins,
size_t num_output_channels,
std::complex<float>* const* output) override {
float low_pow = 0.f;
float high_pow = 0.f;
for (size_t i = 0u; i < num_input_channels; ++i) {
for (size_t j = 0u; j < kNarrowbandSize; ++j) {
float low_mag = std::abs(input[i][j]);
low_pow += low_mag * low_mag;
float high_mag = std::abs(input[i][j + kNarrowbandSize]);
high_pow += high_mag * high_mag;
}
}
low_pow_ += low_pow / (num_input_channels * kFftSize);
high_pow_ += high_pow / (num_input_channels * kFftSize);
}
private:
std::unique_ptr<LappedTransform> transform_;
float window_[kFftSize];
float low_pow_;
float high_pow_;
};
float EncodedPowerRatio(AudioEncoder* encoder,
AudioDecoder* decoder,
test::AudioLoop* audio_loop) {
// Encode and decode.
uint32_t rtp_timestamp = 0u;
constexpr size_t kBufferSize = 500;
rtc::Buffer encoded(kBufferSize);
std::vector<int16_t> decoded(kOutputBlockSizeSamples);
std::vector<float> decoded_float(kOutputBlockSizeSamples);
AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
PowerRatioEstimator power_ratio_estimator;
for (size_t i = 0; i < 1000; ++i) {
encoded.Clear();
AudioEncoder::EncodedInfo encoder_info =
encoder->Encode(rtp_timestamp, audio_loop->GetNextBlock(), &encoded);
rtp_timestamp += kInputBlockSizeSamples;
if (encoded.size() > 0) {
int decoder_info = decoder->Decode(
encoded.data(), encoded.size(), kSampleRateHz,
decoded.size() * sizeof(decoded[0]), decoded.data(), &speech_type);
if (decoder_info > 0) {
S16ToFloat(decoded.data(), decoded.size(), decoded_float.data());
power_ratio_estimator.ProcessBlock(decoded_float.data());
}
}
}
return power_ratio_estimator.PowerRatio();
}
} // namespace
// TODO(ivoc): Remove this test, WebRTC-AdjustOpusBandwidth is obsolete.
TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-AdjustOpusBandwidth/Enabled/");
constexpr float kMaxNarrowbandRatio = 0.0035f;
constexpr float kMinWidebandRatio = 0.01f;
// Create encoder.
AudioEncoderOpusConfig enc_config;
enc_config.bitrate_bps = absl::optional<int>(7999);
enc_config.num_channels = kNumChannels;
constexpr int payload_type = 17;
auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type);
// Create decoder.
AudioDecoderOpus::Config dec_config;
dec_config.num_channels = kNumChannels;
auto decoder = AudioDecoderOpus::MakeAudioDecoder(dec_config);
// Open speech file.
const std::string kInputFileName =
webrtc::test::ResourcePath("audio_coding/speech_mono_32_48kHz", "pcm");
test::AudioLoop audio_loop;
EXPECT_EQ(kSampleRateHz, encoder->SampleRateHz());
ASSERT_TRUE(audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
kInputBlockSizeSamples));
EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
kMaxNarrowbandRatio);
encoder->OnReceivedTargetAudioBitrate(9000);
EXPECT_LT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
kMaxNarrowbandRatio);
encoder->OnReceivedTargetAudioBitrate(9001);
EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
kMinWidebandRatio);
encoder->OnReceivedTargetAudioBitrate(8000);
EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
kMinWidebandRatio);
encoder->OnReceivedTargetAudioBitrate(12001);
EXPECT_GT(EncodedPowerRatio(encoder.get(), decoder.get(), &audio_loop),
kMinWidebandRatio);
}
} // namespace webrtc
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