c1eb560a5c
* The old resampler was found to have a wraparound bug. * Remove support for the old resampler from PushResampler. * Use PushResampler in AudioCodingModule. * The old resampler must still be removed from the file utility. BUG=webrtc:1867,webrtc:827 TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio R=henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1590004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
37 lines
1.1 KiB
C++
37 lines
1.1 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 "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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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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int16_t Resample10Msec(const int16_t* in_audio,
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const int32_t in_freq_hz,
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int16_t* out_audio,
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const int32_t out_freq_hz,
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uint8_t num_audio_channels);
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private:
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PushResampler resampler_;
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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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