mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-15 23:01:21 +01:00
db6af36979
* Move 'VadWithLevel' to AGC2 where it belongs. * Remove the vectors from VadWithLevel. They were there to make it work with modules/audio_processing/vad, which we don't need any longer. * Remove the vector handling from AGC2. It was spread out across AdaptiveDigitalGainApplier, AdaptiveAGC and their unit tests. * Hack the RNN VAD into VadWithLevel. The main issue is the resampling. Bug: webrtc:9076 Change-Id: I13056c985d0ec41269735150caf4aaeb6ff9281e Reviewed-on: https://webrtc-review.googlesource.com/77364 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23688}
68 lines
2.3 KiB
C++
68 lines
2.3 KiB
C++
/*
|
|
* Copyright (c) 2018 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/agc2/vad_with_level.h"
|
|
|
|
#include <algorithm>
|
|
|
|
#include "common_audio/include/audio_util.h"
|
|
#include "modules/audio_processing/agc2/rnn_vad/common.h"
|
|
#include "rtc_base/checks.h"
|
|
|
|
namespace webrtc {
|
|
|
|
namespace {
|
|
float ProcessForPeak(AudioFrameView<const float> frame) {
|
|
float current_max = 0;
|
|
for (const auto& x : frame.channel(0)) {
|
|
current_max = std::max(std::fabs(x), current_max);
|
|
}
|
|
return current_max;
|
|
}
|
|
|
|
float ProcessForRms(AudioFrameView<const float> frame) {
|
|
float rms = 0;
|
|
for (const auto& x : frame.channel(0)) {
|
|
rms += x * x;
|
|
}
|
|
return sqrt(rms / frame.samples_per_channel());
|
|
}
|
|
} // namespace
|
|
|
|
VadWithLevel::VadWithLevel() = default;
|
|
VadWithLevel::~VadWithLevel() = default;
|
|
|
|
VadWithLevel::LevelAndProbability VadWithLevel::AnalyzeFrame(
|
|
AudioFrameView<const float> frame) {
|
|
SetSampleRate(static_cast<int>(frame.samples_per_channel() * 100));
|
|
std::array<float, rnn_vad::kFrameSize10ms24kHz> work_frame;
|
|
// Feed the 1st channel to the resampler.
|
|
resampler_.Resample(frame.channel(0).data(), frame.samples_per_channel(),
|
|
work_frame.data(), rnn_vad::kFrameSize10ms24kHz);
|
|
|
|
std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
|
|
|
|
const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
|
|
work_frame, feature_vector);
|
|
const float vad_probability =
|
|
rnn_vad_.ComputeVadProbability(feature_vector, is_silence);
|
|
return LevelAndProbability(vad_probability,
|
|
FloatS16ToDbfs(ProcessForRms(frame)),
|
|
FloatS16ToDbfs(ProcessForPeak(frame)));
|
|
}
|
|
|
|
void VadWithLevel::SetSampleRate(int sample_rate_hz) {
|
|
// The source number of channels in 1, because we always use the 1st
|
|
// channel.
|
|
resampler_.InitializeIfNeeded(sample_rate_hz, rnn_vad::kSampleRate24kHz,
|
|
1 /* num_channels */);
|
|
}
|
|
|
|
} // namespace webrtc
|