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Merge pull request #15788 from kencooke/audio-simd-UBSan-warnings

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Fix UBSan warnings due to SIMD intrinsics
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Ken Cooke 2019-06-18 13:38:57 -07:00 committed by GitHub
commit 5c837406ac
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2 changed files with 42 additions and 16 deletions
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38
libraries/audio/src/AudioReverb.cpp
View file

@ -35,16 +35,6 @@ static const double SQRT2 = 1.41421356237309504880;
static const double FIXQ31 = 2147483648.0;
static const double FIXQ32 = 4294967296.0;
// Round an integer to the next power-of-two, at compile time.
// VS2013 does not support constexpr so macros are used instead.
#define SETBITS0(x) (x)
#define SETBITS1(x) (SETBITS0(x) | (SETBITS0(x) >> 1))
#define SETBITS2(x) (SETBITS1(x) | (SETBITS1(x) >> 2))
#define SETBITS3(x) (SETBITS2(x) | (SETBITS2(x) >> 4))
#define SETBITS4(x) (SETBITS3(x) | (SETBITS3(x) >> 8))
#define SETBITS5(x) (SETBITS4(x) | (SETBITS4(x) >> 16))
#define NEXTPOW2(x) (SETBITS5((x) - 1) + 1)
//
// Allpass delay modulation
//
@ -111,6 +101,18 @@ static const int M_AP19 = 113;
static const int M_AP20 = 107;
static const int M_AP21 = 127;
// Round an integer to the next power-of-two, at compile time
constexpr uint32_t NEXTPOW2(uint32_t n) {
n -= 1;
n |= (n >> 1);
n |= (n >> 2);
n |= (n >> 4);
n |= (n >> 8);
n |= (n >> 16);
n += 1;
return n;
}
//
// Filter design tools using analog-matched response.
// All filter types approximate the s-plane response, including cutoff > Nyquist.
@ -1796,6 +1798,18 @@ void AudioReverb::render(float** inputs, float** outputs, int numFrames) {
#include <emmintrin.h>
// unaligned load/store without undefined behavior
static inline __m128i mm_loadu_si32(void const* mem_addr) {
int32_t temp;
memcpy(&temp, mem_addr, sizeof(int32_t));
return _mm_cvtsi32_si128(temp);
}
static inline void mm_storeu_si32(void* mem_addr, __m128i a) {
int32_t temp = _mm_cvtsi128_si32(a);
memcpy(mem_addr, &temp, sizeof(int32_t));
}
// convert int16_t to float, deinterleave stereo
void AudioReverb::convertInput(const int16_t* input, float** outputs, int numFrames) {
__m128 scale = _mm_set1_ps(1/32768.0f);
@ -1816,7 +1830,7 @@ void AudioReverb::convertInput(const int16_t* input, float** outputs, int numFra
_mm_storeu_ps(&outputs[1][i], f1);
}
for (; i < numFrames; i++) {
__m128i a0 = _mm_cvtsi32_si128(*(int32_t*)&input[2*i]);
__m128i a0 = mm_loadu_si32((__m128i*)&input[2*i]);
__m128i a1 = a0;
// deinterleave and sign-extend
@ -1887,7 +1901,7 @@ void AudioReverb::convertOutput(float** inputs, int16_t* output, int numFrames)
// interleave
a0 = _mm_unpacklo_epi16(a0, a1);
*(int32_t*)&output[2*i] = _mm_cvtsi128_si32(a0);
mm_storeu_si32((__m128i*)&output[2*i], a0);
}
}

20
libraries/audio/src/AudioSRC.cpp
View file

@ -793,6 +793,18 @@ int AudioSRC::multirateFilter4(const float* input0, const float* input1, const f
#include <emmintrin.h> // SSE2
// unaligned load/store without undefined behavior
static inline __m128i mm_loadu_si32(void const* mem_addr) {
int32_t temp;
memcpy(&temp, mem_addr, sizeof(int32_t));
return _mm_cvtsi32_si128(temp);
}
static inline void mm_storeu_si32(void* mem_addr, __m128i a) {
int32_t temp = _mm_cvtsi128_si32(a);
memcpy(mem_addr, &temp, sizeof(int32_t));
}
// convert int16_t to float, deinterleave stereo
void AudioSRC::convertInput(const int16_t* input, float** outputs, int numFrames) {
__m128 scale = _mm_set1_ps(1/32768.0f);
@ -839,7 +851,7 @@ void AudioSRC::convertInput(const int16_t* input, float** outputs, int numFrames
_mm_storeu_ps(&outputs[1][i], f1);
}
for (; i < numFrames; i++) {
__m128i a0 = _mm_cvtsi32_si128(*(int32_t*)&input[2*i]);
__m128i a0 = mm_loadu_si32((__m128i*)&input[2*i]);
__m128i a1 = a0;
// deinterleave and sign-extend
@ -878,9 +890,9 @@ void AudioSRC::convertInput(const int16_t* input, float** outputs, int numFrames
_mm_storeu_ps(&outputs[3][i], _mm_shuffle_ps(f1, f3, _MM_SHUFFLE(3,1,3,1)));
}
for (; i < numFrames; i++) {
__m128i a0 = _mm_cvtsi32_si128(*(int32_t*)&input[4*i+0]);
__m128i a0 = mm_loadu_si32((__m128i*)&input[4*i+0]);
__m128i a1 = a0;
__m128i a2 = _mm_cvtsi32_si128(*(int32_t*)&input[4*i+2]);
__m128i a2 = mm_loadu_si32((__m128i*)&input[4*i+2]);
__m128i a3 = a2;
// deinterleave and sign-extend
@ -986,7 +998,7 @@ void AudioSRC::convertOutput(float** inputs, int16_t* output, int numFrames) {
// interleave
a0 = _mm_unpacklo_epi16(a0, a1);
*(int32_t*)&output[2*i] = _mm_cvtsi128_si32(a0);
mm_storeu_si32((__m128i*)&output[2*i], a0);
}
} else if (_numChannels == 4) {