
The macro is defined as #define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \ (WEBRTC_SPL_MUL_16_16(a, b) >> (c)) where the latter macro is in C defined as (For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h) The replacement consists of - avoiding casts to int16_t if inputs already are int16_t - adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t) - minor cleanups like remove of unnecessary parantheses and style changes BUG=3348, 3353 TESTED=locally on Mac and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39139004 Cr-Commit-Position: refs/heads/master@{#8544} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8544 4adac7df-926f-26a2-2b94-8c16560cd09d
82 lines
2.6 KiB
C
82 lines
2.6 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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iLBC Speech Coder ANSI-C Source Code
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WebRtcIlbcfix_Chebyshev.c
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******************************************************************/
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#include "defines.h"
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#include "constants.h"
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/*------------------------------------------------------------------*
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* Calculate the Chevyshev polynomial series
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* F(w) = 2*exp(-j5w)*C(x)
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* C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
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* T_i(x) is the i:th order Chebyshev polynomial
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*------------------------------------------------------------------*/
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int16_t WebRtcIlbcfix_Chebyshev(
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/* (o) Result of C(x) */
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int16_t x, /* (i) Value to the Chevyshev polynomial */
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int16_t *f /* (i) The coefficients in the polynomial */
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) {
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int16_t b1_high, b1_low; /* Use the high, low format to increase the accuracy */
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int32_t b2;
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int32_t tmp1W32;
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int32_t tmp2W32;
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int i;
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b2 = (int32_t)0x1000000; /* b2 = 1.0 (Q23) */
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/* Calculate b1 = 2*x + f[1] */
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tmp1W32 = WEBRTC_SPL_LSHIFT_W32((int32_t)x, 10);
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tmp1W32 += WEBRTC_SPL_LSHIFT_W32((int32_t)f[1], 14);
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for (i = 2; i < 5; i++) {
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tmp2W32 = tmp1W32;
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/* Split b1 (in tmp1W32) into a high and low part */
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b1_high = (int16_t)(tmp1W32 >> 16);
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b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
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/* Calculate 2*x*b1-b2+f[i] */
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tmp1W32 = WEBRTC_SPL_LSHIFT_W32(b1_high * x + ((b1_low * x) >> 15), 2);
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tmp1W32 -= b2;
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tmp1W32 += WEBRTC_SPL_LSHIFT_W32((int32_t)f[i], 14);
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/* Update b2 for next round */
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b2 = tmp2W32;
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}
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/* Split b1 (in tmp1W32) into a high and low part */
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b1_high = (int16_t)(tmp1W32 >> 16);
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b1_low = (int16_t)((tmp1W32 - ((int32_t)b1_high << 16)) >> 1);
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/* tmp1W32 = x*b1 - b2 + f[i]/2 */
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tmp1W32 = WEBRTC_SPL_LSHIFT_W32(b1_high * x, 1) +
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WEBRTC_SPL_LSHIFT_W32((b1_low * x) >> 15, 1);
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tmp1W32 -= b2;
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tmp1W32 += WEBRTC_SPL_LSHIFT_W32((int32_t)f[i], 13);
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/* Handle overflows and set to maximum or minimum int16_t instead */
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if (tmp1W32>((int32_t)33553408)) {
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return(WEBRTC_SPL_WORD16_MAX);
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} else if (tmp1W32<((int32_t)-33554432)) {
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return(WEBRTC_SPL_WORD16_MIN);
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} else {
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return (int16_t)(tmp1W32 >> 10);
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}
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}
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