modules/audio_processing/vad/voice_activity_detector.cc

/*
 *  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