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In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
46 lines
1.3 KiB
C
46 lines
1.3 KiB
C
/*
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* Copyright (c) 2011 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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/*
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* This file contains the function WebRtcSpl_GetScalingSquare().
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* The description header can be found in signal_processing_library.h
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*
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*/
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#include "common_audio/signal_processing/include/signal_processing_library.h"
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int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
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size_t in_vector_length,
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size_t times)
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{
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int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
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size_t i;
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int16_t smax = -1;
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int16_t sabs;
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int16_t *sptr = in_vector;
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int16_t t;
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size_t looptimes = in_vector_length;
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for (i = looptimes; i > 0; i--)
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{
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sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
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smax = (sabs > smax ? sabs : smax);
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}
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t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
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if (smax == 0)
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{
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return 0; // Since norm(0) returns 0
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} else
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{
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return (t > nbits) ? 0 : nbits - t;
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}
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}
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