2d2a1f9f05
This required to move the AGC tools source files into webrtc/tools and create a new agc_test_utils target. Since audio_codec_speed_tests.gypi referenced sources above, the best approach I could come up with was to add an audio_coding.gypi file at a higher level and move the targets in there (+ the includes from modules.gyp which is an improvement IMO). I also added a PRESUBMIT.py check to prevent new source entries being added with <(webrtc_root) in the path. BUG=4185 R=andrew@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37859004 Cr-Commit-Position: refs/heads/master@{#8197} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8197 4adac7df-926f-26a2-2b94-8c16560cd09d
29 lines
1.0 KiB
C++
29 lines
1.0 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_TOOLS_AGC_TEST_UTILS_H_
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#define WEBRTC_TOOLS_AGC_TEST_UTILS_H_
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namespace webrtc {
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class AudioFrame;
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float MicLevel2Gain(int gain_range_db, int level);
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float Db2Linear(float db);
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void ApplyGainLinear(float gain, float last_gain, AudioFrame* frame);
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void ApplyGain(float gain_db, float last_gain_db, AudioFrame* frame);
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void SimulateMic(int gain_range_db, int mic_level, int last_mic_level,
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AudioFrame* frame);
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void SimulateMic(int gain_map[255], int mic_level, int last_mic_level,
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AudioFrame* frame);
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} // namespace webrtc
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#endif // WEBRTC_TOOLS_AGC_TEST_UTILS_H_
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