AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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92
webrtc/modules/audio_processing/voice_detection.cc Normal file
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/*
* Copyright (c) 2019 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/voice_detection.h"
#include "common_audio/vad/include/webrtc_vad.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
class VoiceDetection::Vad {
public:
Vad() {
state_ = WebRtcVad_Create();
RTC_CHECK(state_);
int error = WebRtcVad_Init(state_);
RTC_DCHECK_EQ(0, error);
}
~Vad() { WebRtcVad_Free(state_); }
Vad(Vad&) = delete;
Vad& operator=(Vad&) = delete;
VadInst* state() { return state_; }
private:
VadInst* state_ = nullptr;
};
VoiceDetection::VoiceDetection(int sample_rate_hz, Likelihood likelihood)
: sample_rate_hz_(sample_rate_hz),
frame_size_samples_(static_cast<size_t>(sample_rate_hz_ / 100)),
likelihood_(likelihood),
vad_(new Vad()) {
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);
}
VoiceDetection::~VoiceDetection() {}
bool VoiceDetection::ProcessCaptureAudio(AudioBuffer* audio) {
RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
audio->num_frames_per_band());
std::array<int16_t, AudioBuffer::kMaxSplitFrameLength> mixed_low_pass_data;
rtc::ArrayView<const int16_t> mixed_low_pass(mixed_low_pass_data.data(),
audio->num_frames_per_band());
if (audio->num_channels() == 1) {
FloatS16ToS16(audio->split_bands_const(0)[kBand0To8kHz],
audio->num_frames_per_band(), mixed_low_pass_data.data());
} else {
const int num_channels = static_cast<int>(audio->num_channels());
for (size_t i = 0; i < audio->num_frames_per_band(); ++i) {
int32_t value =
FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[0][i]);
for (int j = 1; j < num_channels; ++j) {
value += FloatS16ToS16(audio->split_channels_const(kBand0To8kHz)[j][i]);
}
mixed_low_pass_data[i] = value / num_channels;
}
}
int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
mixed_low_pass.data(), frame_size_samples_);
RTC_DCHECK(vad_ret == 0 || vad_ret == 1);
return vad_ret == 0 ? false : true;
}
} // namespace webrtc