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/*
* Copyright (c) 2015 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 "modules/audio_processing/vad/voice_activity_detector.h"
#include <algorithm>
#include "rtc_base/checks.h"
namespace webrtc {
namespace {
const size_t kNumChannels = 1;
const double kDefaultVoiceValue = 1.0;
const double kNeutralProbability = 0.5;
const double kLowProbability = 0.01;
} // namespace
VoiceActivityDetector::VoiceActivityDetector()
: last_voice_probability_(kDefaultVoiceValue),
standalone_vad_(StandaloneVad::Create()) {}
VoiceActivityDetector::~VoiceActivityDetector() = default;
// Because ISAC has a different chunk length, it updates
// `chunkwise_voice_probabilities_` and `chunkwise_rms_` when there is new data.
// Otherwise it clears them.
void VoiceActivityDetector::ProcessChunk(const int16_t* audio,
size_t length,
int sample_rate_hz) {
RTC_DCHECK_EQ(length, sample_rate_hz / 100);
// TODO(bugs.webrtc.org/7494): Remove resampling and force 16 kHz audio.
// Resample to the required rate.
const int16_t* resampled_ptr = audio;
if (sample_rate_hz != kSampleRateHz) {
RTC_CHECK_EQ(
resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
0);
resampler_.Push(audio, length, resampled_, kLength10Ms, length);
resampled_ptr = resampled_;
}
RTC_DCHECK_EQ(length, kLength10Ms);
// Each chunk needs to be passed into `standalone_vad_`, because internally it
// buffers the audio and processes it all at once when GetActivity() is
// called.
RTC_CHECK_EQ(standalone_vad_->AddAudio(resampled_ptr, length), 0);
audio_processing_.ExtractFeatures(resampled_ptr, length, &features_);
chunkwise_voice_probabilities_.resize(features_.num_frames);
chunkwise_rms_.resize(features_.num_frames);
std::copy(features_.rms, features_.rms + chunkwise_rms_.size(),
chunkwise_rms_.begin());
if (features_.num_frames > 0) {
if (features_.silence) {
// The other features are invalid, so set the voice probabilities to an
// arbitrary low value.
std::fill(chunkwise_voice_probabilities_.begin(),
chunkwise_voice_probabilities_.end(), kLowProbability);
} else {
std::fill(chunkwise_voice_probabilities_.begin(),
chunkwise_voice_probabilities_.end(), kNeutralProbability);
RTC_CHECK_GE(
standalone_vad_->GetActivity(&chunkwise_voice_probabilities_[0],
chunkwise_voice_probabilities_.size()),
0);
RTC_CHECK_GE(pitch_based_vad_.VoicingProbability(
features_, &chunkwise_voice_probabilities_[0]),
0);
}
last_voice_probability_ = chunkwise_voice_probabilities_.back();
}
}
} // namespace webrtc
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