webrtc/modules/audio_coding/neteq/post_decode_vad.cc

/*
 *  Copyright (c) 2013 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 "webrtc/modules/audio_coding/neteq/post_decode_vad.h"

namespace webrtc {

PostDecodeVad::~PostDecodeVad() {
  if (vad_instance_)
    WebRtcVad_Free(vad_instance_);
}

void PostDecodeVad::Enable() {
  if (!vad_instance_) {
    // Create the instance.
    vad_instance_ = WebRtcVad_Create();
    if (vad_instance_ == nullptr) {
      // Failed to create instance.
      Disable();
      return;
    }
  }
  Init();
  enabled_ = true;
}

void PostDecodeVad::Disable() {
  enabled_ = false;
  running_ = false;
}

void PostDecodeVad::Init() {
  running_ = false;
  if (vad_instance_) {
    WebRtcVad_Init(vad_instance_);
    WebRtcVad_set_mode(vad_instance_, kVadMode);
    running_ = true;
  }
}

void PostDecodeVad::Update(int16_t* signal, size_t length,
                           AudioDecoder::SpeechType speech_type,
                           bool sid_frame,
                           int fs_hz) {
  if (!vad_instance_ || !enabled_) {
    return;
  }

  if (speech_type == AudioDecoder::kComfortNoise || sid_frame ||
      fs_hz > 16000) {
    // TODO(hlundin): Remove restriction on fs_hz.
    running_ = false;
    active_speech_ = true;
    sid_interval_counter_ = 0;
  } else if (!running_) {
    ++sid_interval_counter_;
  }

  if (sid_interval_counter_ >= kVadAutoEnable) {
    Init();
  }

  if (length > 0 && running_) {
    size_t vad_sample_index = 0;
    active_speech_ = false;
    // Loop through frame sizes 30, 20, and 10 ms.
    for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
        vad_frame_size_ms -= 10) {
      size_t vad_frame_size_samples =
          static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
      while (length - vad_sample_index >= vad_frame_size_samples) {
        int vad_return = WebRtcVad_Process(
            vad_instance_, fs_hz, &signal[vad_sample_index],
            vad_frame_size_samples);
        active_speech_ |= (vad_return == 1);
        vad_sample_index += vad_frame_size_samples;
      }
    }
  }
}

}  // namespace webrtc