14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
41 lines
1.2 KiB
C++
41 lines
1.2 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_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
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#include "resampler.h"
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#include "typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class ACMResampler {
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public:
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ACMResampler();
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~ACMResampler();
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WebRtc_Word16 Resample10Msec(const WebRtc_Word16* inAudio,
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const WebRtc_Word32 inFreqHz,
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WebRtc_Word16* outAudio,
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const WebRtc_Word32 outFreqHz,
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WebRtc_UWord8 numAudioChannels);
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private:
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// Use the Resampler class.
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Resampler _resampler;
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CriticalSectionWrapper* _resamplerCritSect;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
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