Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq4/post_decode_vad.cc
henrik.lundin@webrtc.org d94659dc27 Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.

It has been through extensive internal review during the course of
the project.

TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1073005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 12:09:21 +00:00

83 lines
2.3 KiB
C++

/*
* 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/neteq4/post_decode_vad.h"
namespace webrtc {
void PostDecodeVad::Enable() {
if (!vad_instance_) {
// Create the instance.
if (WebRtcVad_Create(&vad_instance_) != 0) {
// 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 (size_t vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
vad_frame_size_ms -= 10) {
size_t vad_frame_size_samples = 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