mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-02 17:51:06 +01:00
5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
92 lines
2.4 KiB
C++
92 lines
2.4 KiB
C++
/*
|
|
* Copyright (c) 2012 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/standalone_vad.h"
|
|
|
|
#include <string.h>
|
|
|
|
#include "common_audio/vad/include/webrtc_vad.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
static const int kDefaultStandaloneVadMode = 3;
|
|
|
|
StandaloneVad::StandaloneVad(VadInst* vad)
|
|
: vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {}
|
|
|
|
StandaloneVad::~StandaloneVad() {
|
|
WebRtcVad_Free(vad_);
|
|
}
|
|
|
|
StandaloneVad* StandaloneVad::Create() {
|
|
VadInst* vad = WebRtcVad_Create();
|
|
if (!vad)
|
|
return nullptr;
|
|
|
|
int err = WebRtcVad_Init(vad);
|
|
err |= WebRtcVad_set_mode(vad, kDefaultStandaloneVadMode);
|
|
if (err != 0) {
|
|
WebRtcVad_Free(vad);
|
|
return nullptr;
|
|
}
|
|
return new StandaloneVad(vad);
|
|
}
|
|
|
|
int StandaloneVad::AddAudio(const int16_t* data, size_t length) {
|
|
if (length != kLength10Ms)
|
|
return -1;
|
|
|
|
if (index_ + length > kLength10Ms * kMaxNum10msFrames)
|
|
// Reset the buffer if it's full.
|
|
// TODO(ajm): Instead, consider just processing every 10 ms frame. Then we
|
|
// can forgo the buffering.
|
|
index_ = 0;
|
|
|
|
memcpy(&buffer_[index_], data, sizeof(int16_t) * length);
|
|
index_ += length;
|
|
return 0;
|
|
}
|
|
|
|
int StandaloneVad::GetActivity(double* p, size_t length_p) {
|
|
if (index_ == 0)
|
|
return -1;
|
|
|
|
const size_t num_frames = index_ / kLength10Ms;
|
|
if (num_frames > length_p)
|
|
return -1;
|
|
RTC_DCHECK_EQ(0, WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_));
|
|
|
|
int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_);
|
|
if (activity < 0)
|
|
return -1;
|
|
else if (activity == 0)
|
|
p[0] = 0.01; // Arbitrary but small and non-zero.
|
|
else
|
|
p[0] = 0.5; // 0.5 is neutral values when combinned by other probabilities.
|
|
for (size_t n = 1; n < num_frames; n++)
|
|
p[n] = p[0];
|
|
// Reset the buffer to start from the beginning.
|
|
index_ = 0;
|
|
return activity;
|
|
}
|
|
|
|
int StandaloneVad::set_mode(int mode) {
|
|
if (mode < 0 || mode > 3)
|
|
return -1;
|
|
if (WebRtcVad_set_mode(vad_, mode) != 0)
|
|
return -1;
|
|
|
|
mode_ = mode;
|
|
return 0;
|
|
}
|
|
|
|
} // namespace webrtc
|