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/*
* 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 "webrtc/modules/audio_processing/voice_detection_impl.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
namespace webrtc {
class VoiceDetectionImpl::Vad {
public:
Vad() {
state_ = WebRtcVad_Create();
RTC_CHECK(state_);
int error = WebRtcVad_Init(state_);
RTC_DCHECK_EQ(0, error);
}
~Vad() {
WebRtcVad_Free(state_);
}
VadInst* state() { return state_; }
private:
VadInst* state_ = nullptr;
RTC_DISALLOW_COPY_AND_ASSIGN(Vad);
};
VoiceDetectionImpl::VoiceDetectionImpl(rtc::CriticalSection* crit)
: crit_(crit) {
RTC_DCHECK(crit);
}
VoiceDetectionImpl::~VoiceDetectionImpl() {}
void VoiceDetectionImpl::Initialize(int sample_rate_hz) {
rtc::CritScope cs(crit_);
sample_rate_hz_ = sample_rate_hz;
rtc::scoped_ptr<Vad> new_vad;
if (enabled_) {
new_vad.reset(new Vad());
}
vad_.swap(new_vad);
using_external_vad_ = false;
frame_size_samples_ =
static_cast<size_t>(frame_size_ms_ * sample_rate_hz_) / 1000;
set_likelihood(likelihood_);
}
void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
rtc::CritScope cs(crit_);
if (!enabled_) {
return;
}
if (using_external_vad_) {
using_external_vad_ = false;
return;
}
RTC_DCHECK_GE(160u, audio->num_frames_per_band());
// TODO(ajm): concatenate data in frame buffer here.
int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
audio->mixed_low_pass_data(),
frame_size_samples_);
if (vad_ret == 0) {
stream_has_voice_ = false;
audio->set_activity(AudioFrame::kVadPassive);
} else if (vad_ret == 1) {
stream_has_voice_ = true;
audio->set_activity(AudioFrame::kVadActive);
} else {
RTC_NOTREACHED();
}
}
int VoiceDetectionImpl::Enable(bool enable) {
rtc::CritScope cs(crit_);
if (enabled_ != enable) {
enabled_ = enable;
Initialize(sample_rate_hz_);
}
return AudioProcessing::kNoError;
}
bool VoiceDetectionImpl::is_enabled() const {
rtc::CritScope cs(crit_);
return enabled_;
}
int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
rtc::CritScope cs(crit_);
using_external_vad_ = true;
stream_has_voice_ = has_voice;
return AudioProcessing::kNoError;
}
bool VoiceDetectionImpl::stream_has_voice() const {
rtc::CritScope cs(crit_);
// TODO(ajm): enable this assertion?
//assert(using_external_vad_ || is_component_enabled());
return stream_has_voice_;
}
int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
rtc::CritScope cs(crit_);
likelihood_ = likelihood;
if (enabled_) {
int mode = 2;
switch (likelihood) {
case VoiceDetection::kVeryLowLikelihood:
mode = 3;
break;
case VoiceDetection::kLowLikelihood:
mode = 2;
break;
case VoiceDetection::kModerateLikelihood:
mode = 1;
break;
case VoiceDetection::kHighLikelihood:
mode = 0;
break;
default:
RTC_NOTREACHED();
break;
}
int error = WebRtcVad_set_mode(vad_->state(), mode);
RTC_DCHECK_EQ(0, error);
}
return AudioProcessing::kNoError;
}
VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
rtc::CritScope cs(crit_);
return likelihood_;
}
int VoiceDetectionImpl::set_frame_size_ms(int size) {
rtc::CritScope cs(crit_);
RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported.
frame_size_ms_ = size;
Initialize(sample_rate_hz_);
return AudioProcessing::kNoError;
}
int VoiceDetectionImpl::frame_size_ms() const {
rtc::CritScope cs(crit_);
return frame_size_ms_;
}
} // namespace webrtc
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