session-android/jni/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

88 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/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.
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, int 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_) {
int 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) {
int 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