AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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22
webrtc/common_audio/signal_processing/spl_sqrt.c
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@ -15,9 +15,8 @@
*
*/
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include <assert.h>
#include "rtc_base/checks.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
int32_t WebRtcSpl_SqrtLocal(int32_t in);
@ -139,8 +138,19 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
A = value;
if (A == 0)
return (int32_t)0; // sqrt(0) = 0
// The convention in this function is to calculate sqrt(abs(A)). Negate the
// input if it is negative.
if (A < 0) {
if (A == WEBRTC_SPL_WORD32_MIN) {
// This number cannot be held in an int32_t after negating.
// Map it to the maximum positive value.
A = WEBRTC_SPL_WORD32_MAX;
} else {
A = -A;
}
} else if (A == 0) {
return 0; // sqrt(0) = 0
}
sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
@ -155,7 +165,7 @@ int32_t WebRtcSpl_Sqrt(int32_t value)
x_norm = (int16_t)(A >> 16); // x_norm = AH
nshift = (sh / 2);
assert(nshift >= 0);
RTC_DCHECK_GE(nshift, 0);
A = (int32_t)WEBRTC_SPL_LSHIFT_W32((int32_t)x_norm, 16);
A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)