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Commit 7e71fffb authored by Arun Raghavan's avatar Arun Raghavan
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Update code to upstream revision r766 

Removes matlab tests, adds delay estimation logging, and some other
minor fixes/improvements.
parent 139f0b6d
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configure.ac
View file @ 7e71fffb
# Revision changelog (version - date, svn rev. from upstream that was merged)
# 0.1 - 15 Sep 2011, r597
# 0.1 - 19 Oct 2011, r766
AC_INIT([webrtc-audio-processing], [0.1])
AM_INIT_AUTOMAKE([tar-ustar])
......
src/common_audio/signal_processing_library/OWNERS deleted 100644 → 0
View file @ 139f0b6d
bjornv@webrtc.org
tina.legrand@webrtc.org
jan.skoglund@webrtc.org
src/common_audio/signal_processing_library/main/interface/signal_processing_library.h
View file @ 7e71fffb
......@@ -1659,6 +1659,30 @@ 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.
//
// int32_t WebRtc_MulAccumW16(...)
//
// This function multiply a 16-bit word by a 16-bit word, and accumulate this
// value to a 32-bit integer.
//
// Input:
// - a : The value of the first 16-bit word.
// - b : The value of the second 16-bit word.
// - c : The value of an 32-bit integer.
//
// Return Value: The value of a * b + c.
//
// WebRtc_Word16 WebRtcSpl_get_version(...)
//
// This function gives the version string of the Signal Processing Library.
......
src/common_audio/signal_processing_library/main/interface/spl_inl.h
View file @ 7e71fffb
......@@ -19,16 +19,20 @@
#include "spl_inl_armv7.h"
#else
static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
WebRtc_Word16 b) {
WebRtc_Word32 s_sum = (WebRtc_Word32) a + (WebRtc_Word32) b;
static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
WebRtc_Word16 out16 = (WebRtc_Word16) value32;
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;
if (value32 > 32767)
out16 = 32767;
else if (value32 < -32768)
out16 = -32768;
return (WebRtc_Word16)s_sum;
return out16;
}
static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
WebRtc_Word16 b) {
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,
......@@ -161,6 +148,12 @@ static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
return zeros;
}
static __inline int32_t WebRtc_MulAccumW16(int16_t a,
int16_t b,
int32_t c) {
return (a * b + c);
}
#endif // WEBRTC_ARCH_ARM_V7A
#endif // WEBRTC_SPL_SPL_INL_H_
src/common_audio/signal_processing_library/main/interface/spl_inl_armv7.h
View file @ 7e71fffb
......@@ -45,6 +45,14 @@ static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
return tmp;
}
static __inline int32_t WebRtc_MulAccumW16(int16_t a,
int16_t b,
int32_t c) {
int32_t tmp = 0;
__asm__("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
return tmp;
}
static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
WebRtc_Word16 b) {
WebRtc_Word32 s_sum;
......@@ -119,4 +127,11 @@ 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_
src/common_audio/signal_processing_library/main/source/downsample_fast.c
View file @ 7e71fffb
......@@ -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;
......
src/common_audio/signal_processing_library/main/source/resample_by_2.c
View file @ 7e71fffb
......@@ -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;
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;
// 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;
// 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
}
filtState[0] = state0;
filtState[1] = state1;
filtState[2] = state2;
filtState[3] = state3;
filtState[4] = state4;
filtState[5] = state5;
filtState[6] = state6;
filtState[7] = state7;
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];
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 = 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;
// 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;
}
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];
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;
// 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
// 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 = (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;
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];
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;
// round; limit amplitude to prevent wrap-around; write to output array
out32 = (state3 + 512) >> 10;
*out++ = WebRtcSpl_SatW32ToW16(out32);
// 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;
}
src/common_audio/signal_processing_library/main/source/splitting_filter.c
View file @ 7e71fffb
......@@ -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);
}
}
src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c
View file @ 7e71fffb
......@@ -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);
}
}
......
src/common_audio/vad/Makefile.am
View file @ 7e71fffb
......@@ -2,11 +2,9 @@ noinst_LTLIBRARIES = libvad.la
libvad_la_SOURCES = main/interface/webrtc_vad.h \
main/source/webrtc_vad.c \
main/source/vad_const.c \
main/source/vad_const.h \
main/source/vad_defines.h \
main/source/vad_core.c \
main/source/vad_core.h \
main/source/vad_defines.h \
main/source/vad_filterbank.c \
main/source/vad_filterbank.h \
main/source/vad_gmm.c \
......
src/common_audio/vad/OWNERS deleted 100644 → 0
View file @ 139f0b6d
bjornv@webrtc.org
jan.skoglund@webrtc.org
src/common_audio/vad/main/source/vad.gypi
View file @ 7e71fffb
......@@ -25,11 +25,9 @@
'sources': [
'../interface/webrtc_vad.h',
'webrtc_vad.c',
'vad_const.c',
'vad_const.h',
'vad_defines.h',
'vad_core.c',
'vad_core.h',
'vad_defines.h',
'vad_filterbank.c',
'vad_filterbank.h',
'vad_gmm.c',
......
src/common_audio/vad/main/source/vad_const.c deleted 100644 → 0
View file @ 139f0b6d
/*
* 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 includes the constant values used internally in VAD.
*/
#include "vad_const.h"
// Spectrum Weighting
const WebRtc_Word16 kSpectrumWeight[6] = {6, 8, 10, 12, 14, 16};
const WebRtc_Word16 kCompVar = 22005;
// Constant 160*log10(2) in Q9
const WebRtc_Word16 kLogConst = 24660;
// Constant log2(exp(1)) in Q12
const WebRtc_Word16 kLog10Const = 5909;
// Q15
const WebRtc_Word16 kNoiseUpdateConst = 655;
const WebRtc_Word16 kSpeechUpdateConst = 6554;
// Q8
const WebRtc_Word16 kBackEta = 154;
// Coefficients used by WebRtcVad_HpOutput, Q14
const WebRtc_Word16 kHpZeroCoefs[3] = {6631, -13262, 6631};
const WebRtc_Word16 kHpPoleCoefs[3] = {16384, -7756, 5620};
// Allpass filter coefficients, upper and lower, in Q15
// Upper: 0.64, Lower: 0.17
const WebRtc_Word16 kAllPassCoefsQ15[2] = {20972, 5571};
const WebRtc_Word16 kAllPassCoefsQ13[2] = {5243, 1392}; // Q13
// Minimum difference between the two models, Q5
const WebRtc_Word16 kMinimumDifference[6] = {544, 544, 576, 576, 576, 576};
// Upper limit of mean value for speech model, Q7
const WebRtc_Word16 kMaximumSpeech[6] = {11392, 11392, 11520, 11520, 11520, 11520};