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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
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 "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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