6955870806
IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
40 lines
1.1 KiB
C++
40 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_ACM2_ACM_RESAMPLER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_ACM2_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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namespace acm2 {
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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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int Resample10Msec(const int16_t* in_audio,
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int in_freq_hz,
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int out_freq_hz,
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size_t num_audio_channels,
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size_t out_capacity_samples,
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int16_t* out_audio);
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private:
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PushResampler<int16_t> resampler_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
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