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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
71 lines
2.3 KiB
C++
71 lines
2.3 KiB
C++
/*
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* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/agc2/vad_with_level.h"
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#include <algorithm>
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#include <array>
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#include <cmath>
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#include "api/array_view.h"
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/agc2/rnn_vad/common.h"
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namespace webrtc {
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namespace {
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float ProcessForPeak(AudioFrameView<const float> frame) {
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float current_max = 0;
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for (const auto& x : frame.channel(0)) {
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current_max = std::max(std::fabs(x), current_max);
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}
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return current_max;
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}
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float ProcessForRms(AudioFrameView<const float> frame) {
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float rms = 0;
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for (const auto& x : frame.channel(0)) {
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rms += x * x;
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}
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return sqrt(rms / frame.samples_per_channel());
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}
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} // namespace
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VadWithLevel::VadWithLevel() = default;
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VadWithLevel::~VadWithLevel() = default;
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VadWithLevel::LevelAndProbability VadWithLevel::AnalyzeFrame(
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AudioFrameView<const float> frame) {
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SetSampleRate(static_cast<int>(frame.samples_per_channel() * 100));
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std::array<float, rnn_vad::kFrameSize10ms24kHz> work_frame;
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// Feed the 1st channel to the resampler.
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resampler_.Resample(frame.channel(0).data(), frame.samples_per_channel(),
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work_frame.data(), rnn_vad::kFrameSize10ms24kHz);
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std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
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const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
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work_frame, feature_vector);
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const float vad_probability =
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rnn_vad_.ComputeVadProbability(feature_vector, is_silence);
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return LevelAndProbability(vad_probability,
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FloatS16ToDbfs(ProcessForRms(frame)),
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FloatS16ToDbfs(ProcessForPeak(frame)));
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}
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void VadWithLevel::SetSampleRate(int sample_rate_hz) {
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// The source number of channels in 1, because we always use the 1st
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// channel.
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resampler_.InitializeIfNeeded(sample_rate_hz, rnn_vad::kSampleRate24kHz,
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1 /* num_channels */);
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}
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} // namespace webrtc
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