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common_audio: Removed macro WEBRTC_SPL_DIV

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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
This commit is contained in:
bjornv@webrtc.org
2014-08-28 12:57:32 +00:00
parent 1f8a23757a
commit df9fef6638
10 changed files with 33 additions and 36 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
View File

@@ -282,11 +282,11 @@ int16_t WebRtcIsacfix_DecLogisticMulti2(int16_t *dataQ7,
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;

22
webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
View File

@@ -296,9 +296,9 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
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 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
} 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 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
/* 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 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
/* 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 @@ void WebRtcIsacfix_UpdateRateModel(RateModel *State,
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) {

15
webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
View File

@@ -59,8 +59,8 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
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 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
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 @@ int16_t WebRtcIsacfix_DecodeImpl(int16_t *signal_out16,
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;

8
webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
View File

@@ -350,11 +350,11 @@ static void CalcRootInvArSpec(const int16_t *ARCoefQ12,
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 @@ static void CalcRootInvArSpec(const int16_t *ARCoefQ12,
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;