|  | /* | 
|  | *  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/interface/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 int 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 (int 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 | 
|  |  |