1ece0cbbec
And the follow-up fix in r8198 that was not sufficient. Reason: breaks Chromium bots runhooks (GYP). I will have to try some more to make sure I don't include test code, since include_tests==0 in Chromium. TBR=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37039004 Cr-Commit-Position: refs/heads/master@{#8200} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8200 4adac7df-926f-26a2-2b94-8c16560cd09d
29 lines
1.1 KiB
C++
29 lines
1.1 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.
|
|
*/
|
|
|
|
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
|
|
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
|
|
namespace webrtc {
|
|
|
|
class AudioFrame;
|
|
|
|
float MicLevel2Gain(int gain_range_db, int level);
|
|
float Db2Linear(float db);
|
|
void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame);
|
|
void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame);
|
|
void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
|
|
AudioFrame* frame);
|
|
void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
|
|
AudioFrame* frame);
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
|