rhubarb-lip-sync/lib/webrtc-8d2248ff/webrtc/tools/agc/test_utils.cc

65 lines
2.1 KiB
C++

/*
* 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/tools/agc/test_utils.h"
#include <cmath>
#include <algorithm>
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
float MicLevel2Gain(int gain_range_db, int level) {
return (level - 127.0f) / 128.0f * gain_range_db / 2;
}
float Db2Linear(float db) {
return powf(10.0f, db / 20.0f);
}
void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame) {
const size_t frame_length =
frame->samples_per_channel_ * frame->num_channels_;
// Smooth the transition between gain levels across the frame.
float smoothed_gain = last_gain;
float gain_step = (gain - last_gain) / (frame_length - 1);
for (size_t i = 0; i < frame_length; ++i) {
smoothed_gain += gain_step;
float sample = std::floor(frame->data_[i] * smoothed_gain + 0.5);
sample = std::max(std::min(32767.0f, sample), -32768.0f);
frame->data_[i] = static_cast<int16_t>(sample);
}
}
void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame) {
ApplyGainLinear(Db2Linear(gain_db), Db2Linear(last_gain_db), frame);
}
void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
AudioFrame* frame) {
assert(mic_level >= 0 && mic_level <= 255);
assert(last_mic_level >= 0 && last_mic_level <= 255);
ApplyGain(MicLevel2Gain(gain_range_db, mic_level),
MicLevel2Gain(gain_range_db, last_mic_level),
frame);
}
void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
AudioFrame* frame) {
assert(mic_level >= 0 && mic_level <= 255);
assert(last_mic_level >= 0 && last_mic_level <= 255);
ApplyGain(gain_map[mic_level], gain_map[last_mic_level], frame);
}
} // namespace webrtc