Upstream version 10.39.225.0

[platform/framework/web/crosswalk.git] / src / third_party / webrtc / modules / audio_coding / codecs / isac / fix / source / bandwidth_estimator.c

diff --git a/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
index 724a900..d28a6f7 100644 (file)
--- a/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
+++ b/src/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
@@ -196,7 +196,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
 
       bweStr->maxBwInv = kInvBandwidth[3];
       bweStr->minBwInv = kInvBandwidth[2];
-      bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
+      bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate);
     }
 
     /* kBitsByteSec is in Q15 */
@@ -211,7 +211,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
 
       bweStr->maxBwInv = kInvBandwidth[1];
       bweStr->minBwInv = kInvBandwidth[0];
-      bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
+      bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate);
     }
 
     /* kBitsByteSec is in Q14 */
@@ -265,7 +265,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
       if ((arrivalTime - bweStr->lastUpdate) > FS3) {
 
         /* Calculate expected number of received packets since last update */
-        numPktsExpected =  WEBRTC_SPL_UDIV(arrivalTime - bweStr->lastUpdate, frameSizeSampl);
+        numPktsExpected = (arrivalTime - bweStr->lastUpdate) / frameSizeSampl;
 
         /* If received number of packets is more than 90% of expected (922 = 0.9 in Q10): */
         /* do the update, else not                                                        */
@@ -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 */
@@ -417,8 +417,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
            and NOT right shifting recBwAvg 5 bits to an integer
            At max 13 bits are used
            shift to Q5 */
-        recBwAvgInv = WEBRTC_SPL_UDIV((uint32_t)(0x80000000 + WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 1)),
-                                      bweStr->recBwAvg);
+        recBwAvgInv = (0x80000000 + bweStr->recBwAvg / 2) / bweStr->recBwAvg;
 
         /* Calculate Projected arrival time difference */
 
@@ -513,7 +512,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
   bweStr->prevSendTime = sendTime;
 
   /* Replace bweStr->recBw by the new value */
-  bweStr->recBw = WEBRTC_SPL_UDIV(1073741824, bweStr->recBwInv) - bweStr->recHeaderRate;
+  bweStr->recBw = 1073741824 / bweStr->recBwInv - bweStr->recHeaderRate;
 
   if (immediateSet) {
     /* delay correction factor is in Q10 */
@@ -530,7 +529,7 @@ int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
 
     bweStr->recJitterShortTerm = 0;
 
-    bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, bweStr->recBw + bweStr->recHeaderRate);
+    bweStr->recBwInv = 1073741824 / (bweStr->recBw + bweStr->recHeaderRate);
 
     immediateSet = 0;
   }
@@ -725,7 +724,7 @@ uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
 
   /* Q18  rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
      2^18 then needs to be shifted 13 bits to 2^31 */
-  rec_jitter_short_term_abs_inv = WEBRTC_SPL_UDIV(0x80000000, bweStr->recJitterShortTermAbs);
+  rec_jitter_short_term_abs_inv = 0x80000000u / bweStr->recJitterShortTermAbs;
 
   /* Q27 = 9 + 18 */
   jitter_sign = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 4), (int32_t)rec_jitter_short_term_abs_inv);
@@ -855,13 +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) {
@@ -887,7 +887,7 @@ uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
   //round and shift before conversion
   MinRate += 256;
   MinRate = WEBRTC_SPL_RSHIFT_W32(MinRate, 9);
-  MinBytes = (uint16_t)WEBRTC_SPL_UDIV(WEBRTC_SPL_MUL(MinRate, FrameSamples), FS8);
+  MinBytes = MinRate * FrameSamples / FS8;
 
   /* StreamSize will be adjusted if less than MinBytes */
   if (StreamSize < MinBytes) {
@@ -896,10 +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;
       }
@@ -923,7 +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) {
@@ -946,13 +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) {