blob: b7c64373fc8a00a8291046e71741c133f8739e1d [file] [log] [blame]
* 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/adaptive_mode_level_estimator_agc.h"
#include <cmath>
#include <vector>
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
ApmDataDumper* apm_data_dumper)
: level_estimator_(apm_data_dumper) {
// |audio| must be mono; in a multi-channel stream, provide the first (usually
// left) channel.
void AdaptiveModeLevelEstimatorAgc::Process(const int16_t* audio,
size_t length,
int sample_rate_hz) {
std::vector<float> float_audio_frame(audio, audio + length);
const float* const first_channel = &float_audio_frame[0];
AudioFrameView<const float> frame_view(&first_channel, 1 /* num channels */,
const auto vad_prob = agc2_vad_.AnalyzeFrame(frame_view);
latest_voice_probability_ = vad_prob.speech_probability;
if (latest_voice_probability_ > kVadConfidenceThreshold) {
time_in_ms_since_last_estimate_ += kFrameDurationMs;
// Retrieves the difference between the target RMS level and the current
// signal RMS level in dB. Returns true if an update is available and false
// otherwise, in which case |error| should be ignored and no action taken.
bool AdaptiveModeLevelEstimatorAgc::GetRmsErrorDb(int* error) {
if (time_in_ms_since_last_estimate_ <= kTimeUntilConfidentMs) {
return false;
*error = std::floor(target_level_dbfs() -
level_estimator_.LatestLevelEstimate() + 0.5f);
time_in_ms_since_last_estimate_ = 0;
return true;
void AdaptiveModeLevelEstimatorAgc::Reset() {
float AdaptiveModeLevelEstimatorAgc::voice_probability() const {
return latest_voice_probability_;
} // namespace webrtc