mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-11 16:49:52 +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.
67 lines
1.7 KiB
C++
67 lines
1.7 KiB
C++
/*
|
|
* Copyright (c) 2014 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 "common_audio/vad/include/vad.h"
|
|
|
|
#include <memory>
|
|
|
|
#include "common_audio/vad/include/webrtc_vad.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
|
|
class VadImpl final : public Vad {
|
|
public:
|
|
explicit VadImpl(Aggressiveness aggressiveness)
|
|
: handle_(nullptr), aggressiveness_(aggressiveness) {
|
|
Reset();
|
|
}
|
|
|
|
~VadImpl() override { WebRtcVad_Free(handle_); }
|
|
|
|
Activity VoiceActivity(const int16_t* audio,
|
|
size_t num_samples,
|
|
int sample_rate_hz) override {
|
|
int ret = WebRtcVad_Process(handle_, sample_rate_hz, audio, num_samples);
|
|
switch (ret) {
|
|
case 0:
|
|
return kPassive;
|
|
case 1:
|
|
return kActive;
|
|
default:
|
|
RTC_NOTREACHED() << "WebRtcVad_Process returned an error.";
|
|
return kError;
|
|
}
|
|
}
|
|
|
|
void Reset() override {
|
|
if (handle_)
|
|
WebRtcVad_Free(handle_);
|
|
handle_ = WebRtcVad_Create();
|
|
RTC_CHECK(handle_);
|
|
RTC_CHECK_EQ(WebRtcVad_Init(handle_), 0);
|
|
RTC_CHECK_EQ(WebRtcVad_set_mode(handle_, aggressiveness_), 0);
|
|
}
|
|
|
|
private:
|
|
VadInst* handle_;
|
|
Aggressiveness aggressiveness_;
|
|
};
|
|
|
|
} // namespace
|
|
|
|
std::unique_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness) {
|
|
return std::unique_ptr<Vad>(new VadImpl(aggressiveness));
|
|
}
|
|
|
|
} // namespace webrtc
|