Remove unused agc_unittest.cc and agc_harness.cc

webrtc/tools/agc/test_utils.cc

-/*
- *  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
-