common_audio: Removed macro WEBRTC_SPL_DIV
The macro has no built-in divide by zero check. The only thing that is done is casting to int32_t.
In addition a bug was discovered where it was supposed to do a division with rounding, but instead did a division with truncation + addition by 2. This is corrected in this CL.
BUG=3348,3353
TESTED=locally on Linux
R=kwiberg@webrtc.org, tina.legrand@webrtc.org, turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/19129004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6998 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
index a9cf384..56bbbe6 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -46,8 +46,6 @@
((uint32_t) ((uint32_t)(a) * (uint16_t)(b)))
#define WEBRTC_SPL_MUL_16_U16(a, b) \
((int32_t)(int16_t)(a) * (uint16_t)(b))
-#define WEBRTC_SPL_DIV(a, b) \
- ((int32_t) ((int32_t)(a) / (int32_t)(b)))
#ifndef WEBRTC_ARCH_ARM_V7
// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index 5d07f16..6a70a02 100644
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -47,7 +47,6 @@
a = b;
b = -3;
- EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
index e574165..4efdecc 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
@@ -282,11 +282,11 @@
if (inSqrt < 0)
inSqrt=-inSqrt;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
+ newRes = (inSqrt / res + res) >> 1;
do
{
res = newRes;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
+ newRes = (inSqrt / res + res) >> 1;
} while (newRes != res && i-- > 0);
tmpARSpecQ8 = (uint16_t)newRes;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index 2ac1535..d28a6f7 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@@ -296,9 +296,9 @@
bweStr->recBwInv = WEBRTC_SPL_RSHIFT_W32((int32_t)bweStr->recBwInv, 13);
} else {
- /* recBwInv = 1 / (INIT_BN_EST + INIT_HDR_RATE) in Q26 (Q30??)*/
- bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 +
- WEBRTC_SPL_LSHIFT_W32(((int32_t)INIT_BN_EST + INIT_HDR_RATE), 1)), INIT_BN_EST + INIT_HDR_RATE);
+ static const uint32_t kInitRate = INIT_BN_EST + INIT_HDR_RATE;
+ /* recBwInv = 1 / kInitRate in Q26 (Q30??)*/
+ bweStr->recBwInv = (1073741824 + kInitRate / 2) / kInitRate;
}
/* reset time-since-update counter */
@@ -854,13 +854,14 @@
} else {
/* handle burst */
if (State->BurstCounter) {
- if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_LEN)), DelayBuildUp), 9)) {
+ if (State->StillBuffered <
+ (((512 - 512 / BURST_LEN) * DelayBuildUp) >> 9)) {
/* max bps derived from BottleNeck and DelayBuildUp values */
- inv_Q12 = WEBRTC_SPL_DIV(4096, WEBRTC_SPL_MUL(BURST_LEN, FrameSamples));
+ inv_Q12 = 4096 / (BURST_LEN * FrameSamples);
MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp, inv_Q12), 3)), BottleNeck);
} else {
/* max bps derived from StillBuffered and DelayBuildUp values */
- inv_Q12 = WEBRTC_SPL_DIV(4096, FrameSamples);
+ inv_Q12 = 4096 / FrameSamples;
if (DelayBuildUp > State->StillBuffered) {
MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp - State->StillBuffered, inv_Q12), 3)), BottleNeck);
} else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) {
@@ -895,10 +896,10 @@
/* keep track of when bottle neck was last exceeded by at least 1% */
//517/512 ~ 1.01
- if (WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, FS8), FrameSamples) > (WEBRTC_SPL_MUL(517, BottleNeck) >> 9)) {
+ if ((StreamSize * (int32_t)FS8) / FrameSamples > (517 * BottleNeck) >> 9) {
if (State->PrevExceed) {
/* bottle_neck exceded twice in a row, decrease ExceedAgo */
- State->ExceedAgo -= WEBRTC_SPL_DIV(BURST_INTERVAL, BURST_LEN - 1);
+ State->ExceedAgo -= BURST_INTERVAL / (BURST_LEN - 1);
if (State->ExceedAgo < 0) {
State->ExceedAgo = 0;
}
@@ -922,7 +923,7 @@
/* Update buffer delay */
- TransmissionTime = (int16_t)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck); /* ms */
+ TransmissionTime = (StreamSize * 8000) / BottleNeck; /* ms */
State->StillBuffered += TransmissionTime;
State->StillBuffered -= (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); //>>4 = SAMPLES_PER_MSEC /* ms */
if (State->StillBuffered < 0) {
@@ -945,13 +946,12 @@
const int16_t FrameSamples, /* samples per frame */
const int16_t BottleNeck) /* bottle neck rate; excl headers (bps) */
{
- int16_t TransmissionTime;
+ const int16_t TransmissionTime = (StreamSize * 8000) / BottleNeck; /* ms */
/* avoid the initial "high-rate" burst */
State->InitCounter = 0;
/* Update buffer delay */
- TransmissionTime = (int16_t)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 1000), BottleNeck); /* ms */
State->StillBuffered += TransmissionTime;
State->StillBuffered -= (int16_t)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
if (State->StillBuffered < 0) {
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
index 0f5c819..263f88a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
@@ -59,8 +59,8 @@
int16_t frame_nb; /* counter */
- int16_t frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
- int16_t processed_samples;
+ int16_t frame_mode; /* 0 for 30ms, 1 for 60ms */
+ static const int16_t kProcessedSamples = 480; /* 480 (for both 30, 60 ms) */
/* PLC */
int16_t overlapWin[ 240 ];
@@ -76,14 +76,14 @@
if (err<0) // error check
return err;
- frame_mode = (int16_t)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
- processed_samples = (int16_t)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
+ frame_mode = *current_framesamples / MAX_FRAMESAMPLES; /* 0, or 1 */
err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno);
if (err<0) // error check
return err;
- /* one loop if it's one frame (20 or 30ms), 2 loops if 2 frames bundled together (60ms) */
+ /* one loop if it's one frame (30ms), two loops if two frames bundled together
+ * (60ms) */
for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
/* decode & dequantize pitch parameters */
@@ -210,7 +210,10 @@
Vector_Word16_2[k] = tmp_2;
}
- WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1, Vector_Word16_2, signal_out16 + frame_nb * processed_samples, &ISACdec_obj->postfiltbankstr_obj);
+ WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1,
+ Vector_Word16_2,
+ signal_out16 + frame_nb * kProcessedSamples,
+ &ISACdec_obj->postfiltbankstr_obj);
}
return len;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 435f572..27d1c1f 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -350,11 +350,11 @@
if(in_sqrt<0)
in_sqrt=-in_sqrt;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ newRes = (in_sqrt / res + res) >> 1;
do
{
res = newRes;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ newRes = (in_sqrt / res + res) >> 1;
} while (newRes != res && i-- > 0);
CurveQ8[k] = (int16_t)newRes;
@@ -368,11 +368,11 @@
if(in_sqrt<0)
in_sqrt=-in_sqrt;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ newRes = (in_sqrt / res + res) >> 1;
do
{
res = newRes;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
+ newRes = (in_sqrt / res + res) >> 1;
} while (newRes != res && i-- > 0);
CurveQ8[k] = (int16_t)newRes;
diff --git a/webrtc/modules/audio_processing/agc/analog_agc.c b/webrtc/modules/audio_processing/agc/analog_agc.c
index 4f110cc..0376dae 100644
--- a/webrtc/modules/audio_processing/agc/analog_agc.c
+++ b/webrtc/modules/audio_processing/agc/analog_agc.c
@@ -208,7 +208,7 @@
tmp16 = (int16_t)(stt->micVol - stt->maxAnalog);
tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
tmp16 = (int16_t)(stt->maxLevel - stt->maxAnalog);
- targetGainIdx = (uint16_t)WEBRTC_SPL_DIV(tmp32, tmp16);
+ targetGainIdx = tmp32 / tmp16;
assert(targetGainIdx < GAIN_TBL_LEN);
/* Increment through the table towards the target gain.
@@ -1078,8 +1078,7 @@
tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
if (stt->maxInit != stt->minLevel)
{
- volNormFIX = (int16_t)WEBRTC_SPL_DIV(tmp32,
- (stt->maxInit - stt->minLevel));
+ volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
}
/* Find correct curve */
@@ -1138,8 +1137,7 @@
tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
if (stt->maxInit != stt->minLevel)
{
- volNormFIX = (int16_t)WEBRTC_SPL_DIV(tmp32,
- (stt->maxInit - stt->minLevel));
+ volNormFIX = tmp32 / (stt->maxInit - stt->minLevel);
}
/* Find correct curve */
diff --git a/webrtc/modules/audio_processing/agc/digital_agc.c b/webrtc/modules/audio_processing/agc/digital_agc.c
index 7b515a5..b15b6e3 100644
--- a/webrtc/modules/audio_processing/agc/digital_agc.c
+++ b/webrtc/modules/audio_processing/agc/digital_agc.c
@@ -210,7 +210,7 @@
{
numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
}
- y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
+ y32 = numFIX / tmp32no1; // in Q14
if (limiterEnable && (i < limiterIdx))
{
tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
diff --git a/webrtc/modules/audio_processing/ns/nsx_core.c b/webrtc/modules/audio_processing/ns/nsx_core.c
index 41244d4..5a88c12 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core.c
@@ -1493,8 +1493,7 @@
}
assert(inst->energyIn > 0);
- energyRatio = (int16_t)WEBRTC_SPL_DIV(energyOut
- + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
+ energyRatio = (energyOut + inst->energyIn / 2) / inst->energyIn; // Q8
// Limit the ratio to [0, 1] in Q8, i.e., [0, 256]
energyRatio = WEBRTC_SPL_SAT(256, energyRatio, 0);
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_c.c b/webrtc/modules/audio_processing/ns/nsx_core_c.c
index de92441..b50d4f8 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_c.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_c.c
@@ -258,8 +258,8 @@
tmp32no1 = WEBRTC_SPL_LSHIFT_W32((int32_t)inst->priorNonSpeechProb,
8); // Q22
- nonSpeechProbFinal[i] = (uint16_t)WEBRTC_SPL_DIV(tmp32no1,
- (int32_t)inst->priorNonSpeechProb + invLrtFX); // Q8
+ nonSpeechProbFinal[i] = tmp32no1 /
+ (inst->priorNonSpeechProb + invLrtFX); // Q8
}
}
}
