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https://github.com/danog/libtgvoip.git
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333c4a1101
Added simple audio resampler Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
40 lines
1.1 KiB
C
40 lines
1.1 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_Energy().
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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 "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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int32_t WebRtcSpl_Energy(int16_t* vector,
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size_t vector_length,
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int* scale_factor)
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{
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int32_t en = 0;
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size_t i;
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int scaling =
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WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
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size_t looptimes = vector_length;
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int16_t *vectorptr = vector;
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for (i = 0; i < looptimes; i++)
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{
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en += (*vectorptr * *vectorptr) >> scaling;
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vectorptr++;
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}
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*scale_factor = scaling;
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return en;
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}
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