mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-03 10:07:45 +01:00
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.
47 lines
1.3 KiB
C
47 lines
1.3 KiB
C
/*
|
|
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
|
|
/*
|
|
* This file contains the function WebRtcSpl_GetScalingSquare().
|
|
* The description header can be found in signal_processing_library.h
|
|
*
|
|
*/
|
|
|
|
#include "common_audio/signal_processing/include/signal_processing_library.h"
|
|
|
|
int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector,
|
|
size_t in_vector_length,
|
|
size_t times)
|
|
{
|
|
int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
|
|
size_t i;
|
|
int16_t smax = -1;
|
|
int16_t sabs;
|
|
int16_t *sptr = in_vector;
|
|
int16_t t;
|
|
size_t looptimes = in_vector_length;
|
|
|
|
for (i = looptimes; i > 0; i--)
|
|
{
|
|
sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
|
|
smax = (sabs > smax ? sabs : smax);
|
|
}
|
|
t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
|
|
|
|
if (smax == 0)
|
|
{
|
|
return 0; // Since norm(0) returns 0
|
|
} else
|
|
{
|
|
return (t > nbits) ? 0 : nbits - t;
|
|
}
|
|
}
|