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In spl, introduced function WebRtcSpl_Sat32To16(), and changed file resample_by_2.c, both for optimization in ARMv7.

Browse Source 
Review URL: http://webrtc-codereview.appspot.com/140010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@649 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
kma@google.com 2011-09-26 16:35:25 +00:00
parent e185e9f68a
commit 961885a8bb
7 changed files with 179 additions and 165 deletions
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11
src/common_audio/signal_processing_library/main/interface/signal_processing_library.h
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@ -1659,6 +1659,17 @@ void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
// - out_data : Super-wideband speech signal, 0-16 kHz
//
// WebRtc_Word16 WebRtcSpl_SatW32ToW16(...)
//
// This function saturates a 32-bit word into a 16-bit word.
//
// Input:
// - value32 : The value of a 32-bit word.
//
// Output:
// - out16 : the saturated 16-bit word.
//
// WebRtc_Word16 WebRtcSpl_get_version(...)
//
// This function gives the version string of the Signal Processing Library.

39
src/common_audio/signal_processing_library/main/interface/spl_inl.h
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@ -19,16 +19,20 @@
#include "spl_inl_armv7.h"
#else
static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
WebRtc_Word16 out16 = (WebRtc_Word16) value32;
if (value32 > 32767)
out16 = 32767;
else if (value32 < -32768)
out16 = -32768;
return out16;
}
static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
WebRtc_Word16 b) {
WebRtc_Word32 s_sum = (WebRtc_Word32) a + (WebRtc_Word32) b;
if (s_sum > WEBRTC_SPL_WORD16_MAX)
s_sum = WEBRTC_SPL_WORD16_MAX;
else if (s_sum < WEBRTC_SPL_WORD16_MIN)
s_sum = WEBRTC_SPL_WORD16_MIN;
return (WebRtc_Word16)s_sum;
return WebRtcSpl_SatW32ToW16((WebRtc_Word32) a + (WebRtc_Word32) b);
}
static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
@ -54,24 +58,7 @@ static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
WebRtc_Word16 var2) {
WebRtc_Word32 l_diff;
WebRtc_Word16 s_diff;
// perform subtraction
l_diff = (WebRtc_Word32)var1 - (WebRtc_Word32)var2;
// default setting
s_diff = (WebRtc_Word16) l_diff;
// check for overflow
if (l_diff > (WebRtc_Word32)32767)
s_diff = (WebRtc_Word16)32767;
// check for underflow
if (l_diff < (WebRtc_Word32)-32768)
s_diff = (WebRtc_Word16)-32768;
return s_diff;
return WebRtcSpl_SatW32ToW16((WebRtc_Word32) var1 - (WebRtc_Word32) var2);
}
static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,

7
src/common_audio/signal_processing_library/main/interface/spl_inl_armv7.h
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@ -119,4 +119,11 @@ static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
return tmp - 17;
}
static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
WebRtc_Word16 out16;
__asm__("ssat %r0, #16, %r1" : "=r"(out16) : "r"(value32));
return out16;
}
#endif // WEBRTC_SPL_SPL_INL_ARMV7_H_

2
src/common_audio/signal_processing_library/main/source/downsample_fast.c
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@ -52,7 +52,7 @@ int WebRtcSpl_DownsampleFast(WebRtc_Word16 *in_ptr, WebRtc_Word16 in_length,
// If output is higher than 32768, saturate it. Same with negative side
*downsampled_ptr++ = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, o, -32768);
*downsampled_ptr++ = WebRtcSpl_SatW32ToW16(o);
}
return 0;

273
src/common_audio/signal_processing_library/main/source/resample_by_2.c
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@ -17,154 +17,165 @@
#include "signal_processing_library.h"
#ifdef WEBRTC_ARCH_ARM_V7A
// allpass filter coefficients.
static const WebRtc_UWord32 kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
static const WebRtc_UWord32 kResampleAllpass2[3] =
{12199, 37471 << 15, 60255 << 15};
// Multiply two 32-bit values and accumulate to another input value.
// Return: state + ((diff * tbl_value) >> 16)
static __inline WebRtc_Word32 MUL_ACCUM_1(WebRtc_Word32 tbl_value,
WebRtc_Word32 diff,
WebRtc_Word32 state) {
WebRtc_Word32 result;
__asm__("smlawb %r0, %r1, %r2, %r3": "=r"(result): "r"(diff),
"r"(tbl_value), "r"(state));
return result;
}
// Multiply two 32-bit values and accumulate to another input value.
// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
//
// The reason to introduce this function is that, in case we can't use smlawb
// instruction (in MUL_ACCUM_1) due to input value range, we can still use
// smmla to save some cycles.
static __inline WebRtc_Word32 MUL_ACCUM_2(WebRtc_Word32 tbl_value,
WebRtc_Word32 diff,
WebRtc_Word32 state) {
WebRtc_Word32 result;
__asm__("smmla %r0, %r1, %r2, %r3": "=r"(result): "r"(diff << 1),
"r"(tbl_value), "r"(state));
return result;
}
#else
// allpass filter coefficients.
static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
#endif // WEBRTC_ARCH_ARM_V7A
// decimator
void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
WebRtc_Word16* out, WebRtc_Word32* filtState)
{
WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
WebRtc_Word16 i;
WebRtc_Word16* out, WebRtc_Word32* filtState) {
WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
WebRtc_Word16 i;
register WebRtc_Word32 state0 = filtState[0];
register WebRtc_Word32 state1 = filtState[1];
register WebRtc_Word32 state2 = filtState[2];
register WebRtc_Word32 state3 = filtState[3];
register WebRtc_Word32 state4 = filtState[4];
register WebRtc_Word32 state5 = filtState[5];
register WebRtc_Word32 state6 = filtState[6];
register WebRtc_Word32 state7 = filtState[7];
register WebRtc_Word32 state0 = filtState[0];
register WebRtc_Word32 state1 = filtState[1];
register WebRtc_Word32 state2 = filtState[2];
register WebRtc_Word32 state3 = filtState[3];
register WebRtc_Word32 state4 = filtState[4];
register WebRtc_Word32 state5 = filtState[5];
register WebRtc_Word32 state6 = filtState[6];
register WebRtc_Word32 state7 = filtState[7];
for (i = (len >> 1); i > 0; i--)
{
// lower allpass filter
in32 = (WebRtc_Word32)(*in++) << 10;
diff = in32 - state1;
tmp1 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass2[0], diff, state0);
state0 = in32;
diff = tmp1 - state2;
tmp2 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass2[1], diff, state1);
state1 = tmp1;
diff = tmp2 - state3;
state3 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass2[2], diff, state2);
state2 = tmp2;
for (i = (len >> 1); i > 0; i--) {
// lower allpass filter
in32 = (WebRtc_Word32)(*in++) << 10;
diff = in32 - state1;
tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
state0 = in32;
diff = tmp1 - state2;
tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
state1 = tmp1;
diff = tmp2 - state3;
state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
state2 = tmp2;
// upper allpass filter
in32 = (WebRtc_Word32)(*in++) << 10;
diff = in32 - state5;
tmp1 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass1[0], diff, state4);
state4 = in32;
diff = tmp1 - state6;
tmp2 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass1[1], diff, state5);
state5 = tmp1;
diff = tmp2 - state7;
state7 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass1[2], diff, state6);
state6 = tmp2;
// upper allpass filter
in32 = (WebRtc_Word32)(*in++) << 10;
diff = in32 - state5;
tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
state4 = in32;
diff = tmp1 - state6;
tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
state5 = tmp1;
diff = tmp2 - state7;
state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
state6 = tmp2;
// add two allpass outputs, divide by two and round
out32 = (state3 + state7 + 1024) >> 11;
// add two allpass outputs, divide by two and round
out32 = (state3 + state7 + 1024) >> 11;
// limit amplitude to prevent wrap-around, and write to output array
#ifdef WEBRTC_ARCH_ARM_V7A
__asm__("ssat %r0, #16, %r1" : "=r"(*out) : "r"(out32));
out++;
#else
if (out32 > 32767)
*out++ = 32767;
else if (out32 < -32768)
*out++ = -32768;
else
*out++ = (WebRtc_Word16)out32;
#endif
}
// limit amplitude to prevent wrap-around, and write to output array
*out++ = WebRtcSpl_SatW32ToW16(out32);
}
filtState[0] = state0;
filtState[1] = state1;
filtState[2] = state2;
filtState[3] = state3;
filtState[4] = state4;
filtState[5] = state5;
filtState[6] = state6;
filtState[7] = state7;
filtState[0] = state0;
filtState[1] = state1;
filtState[2] = state2;
filtState[3] = state3;
filtState[4] = state4;
filtState[5] = state5;
filtState[6] = state6;
filtState[7] = state7;
}
void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
WebRtc_Word32* filtState)
{
WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
WebRtc_Word16 i;
register WebRtc_Word32 state0 = filtState[0];
register WebRtc_Word32 state1 = filtState[1];
register WebRtc_Word32 state2 = filtState[2];
register WebRtc_Word32 state3 = filtState[3];
register WebRtc_Word32 state4 = filtState[4];
register WebRtc_Word32 state5 = filtState[5];
register WebRtc_Word32 state6 = filtState[6];
register WebRtc_Word32 state7 = filtState[7];
void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len,
WebRtc_Word16* out, WebRtc_Word32* filtState) {
WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
WebRtc_Word16 i;
for (i = len; i > 0; i--)
{
// lower allpass filter
in32 = (WebRtc_Word32)(*in++) << 10;
diff = in32 - state1;
tmp1 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass1[0], diff, state0);
state0 = in32;
diff = tmp1 - state2;
tmp2 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass1[1], diff, state1);
state1 = tmp1;
diff = tmp2 - state3;
state3 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass1[2], diff, state2);
state2 = tmp2;
register WebRtc_Word32 state0 = filtState[0];
register WebRtc_Word32 state1 = filtState[1];
register WebRtc_Word32 state2 = filtState[2];
register WebRtc_Word32 state3 = filtState[3];
register WebRtc_Word32 state4 = filtState[4];
register WebRtc_Word32 state5 = filtState[5];
register WebRtc_Word32 state6 = filtState[6];
register WebRtc_Word32 state7 = filtState[7];
// round; limit amplitude to prevent wrap-around; write to output array
out32 = (state3 + 512) >> 10;
#ifdef WEBRTC_ARCH_ARM_V7A
__asm__("ssat %r0, #16, %r1":"=r"(*out): "r"(out32));
out++;
#else
if (out32 > 32767)
*out++ = 32767;
else if (out32 < -32768)
*out++ = -32768;
else
*out++ = (WebRtc_Word16)out32;
#endif
for (i = len; i > 0; i--) {
// lower allpass filter
in32 = (WebRtc_Word32)(*in++) << 10;
diff = in32 - state1;
tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
state0 = in32;
diff = tmp1 - state2;
tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
state1 = tmp1;
diff = tmp2 - state3;
state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
state2 = tmp2;
// upper allpass filter
diff = in32 - state5;
tmp1 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass2[0], diff, state4);
state4 = in32;
diff = tmp1 - state6;
tmp2 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass2[1], diff, state5);
state5 = tmp1;
diff = tmp2 - state7;
state7 = WEBRTC_SPL_SCALEDIFF32(kResampleAllpass2[2], diff, state6);
state6 = tmp2;
// round; limit amplitude to prevent wrap-around; write to output array
out32 = (state3 + 512) >> 10;
*out++ = WebRtcSpl_SatW32ToW16(out32);
// round; limit amplitude to prevent wrap-around; write to output array
out32 = (state7 + 512) >> 10;
#ifdef WEBRTC_ARCH_ARM_V7A
__asm__("ssat %r0, #16, %r1":"=r"(*out): "r"(out32));
out++;
#else
if (out32 > 32767)
*out++ = 32767;
else if (out32 < -32768)
*out++ = -32768;
else
*out++ = (WebRtc_Word16)out32;
#endif
}
filtState[0] = state0;
filtState[1] = state1;
filtState[2] = state2;
filtState[3] = state3;
filtState[4] = state4;
filtState[5] = state5;
filtState[6] = state6;
filtState[7] = state7;
// upper allpass filter
diff = in32 - state5;
tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
state4 = in32;
diff = tmp1 - state6;
tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
state5 = tmp1;
diff = tmp2 - state7;
state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
state6 = tmp2;
// round; limit amplitude to prevent wrap-around; write to output array
out32 = (state7 + 512) >> 10;
*out++ = WebRtcSpl_SatW32ToW16(out32);
}
filtState[0] = state0;
filtState[1] = state1;
filtState[2] = state2;
filtState[3] = state3;
filtState[4] = state4;
filtState[5] = state5;
filtState[6] = state6;
filtState[7] = state7;
}

10
src/common_audio/signal_processing_library/main/source/splitting_filter.c
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@ -147,13 +147,11 @@ void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data, WebRtc_Word16* low_band
{
tmp = filter1[i] + filter2[i] + 1024;
tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
low_band[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
tmp, WEBRTC_SPL_WORD16_MIN);
low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
tmp = filter1[i] - filter2[i] + 1024;
tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
high_band[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
tmp, WEBRTC_SPL_WORD16_MIN);
high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
}
}
@ -191,10 +189,10 @@ void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band, const WebRtc_Word16*
for (i = 0, k = 0; i < kBandFrameLength; i++)
{
tmp = WEBRTC_SPL_RSHIFT_W32(filter2[i] + 512, 10);
out_data[k++] = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmp, -32768);
out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
tmp = WEBRTC_SPL_RSHIFT_W32(filter1[i] + 512, 10);
out_data[k++] = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmp, -32768);
out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
}
}

2
src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c
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@ -125,7 +125,7 @@ void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word1
for (i = 0; i < in_vector_length; i++)
{
tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
( *outptr++) = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmpW32, -32768);
(*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
}
}