common_audio refactoring: Removed macro WEBRTC_SPL_LSHIFT_U32
The macro is a trivial shift operator including a cast before shift. There is no guard against negative shifts. Replaced with << at place and added casts when necessary.
Affects both fixed and float point versions of iSAC
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/27369004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7320 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 21c5663..ba306ed 100644
--- a/webrtc/common_audio/signal_processing/include/signal_processing_library.h
+++ b/webrtc/common_audio/signal_processing/include/signal_processing_library.h
@@ -95,7 +95,6 @@
#define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c))
#define WEBRTC_SPL_RSHIFT_U32(x, c) ((uint32_t)(x) >> (c))
-#define WEBRTC_SPL_LSHIFT_U32(x, c) ((uint32_t)(x) << (c))
#define WEBRTC_SPL_RAND(a) \
((int16_t)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
diff --git a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
index 3fa3d2e..d5cc5f0 100644
--- a/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
+++ b/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
@@ -72,7 +72,6 @@
EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
- EXPECT_EQ(32766u, WEBRTC_SPL_LSHIFT_U32(a, 1));
EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
index 5311b39..2cc020d 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
@@ -310,7 +310,7 @@
if (streamData->stream_index == 0)
{
/* read first word from bytestream */
- streamval = WEBRTC_SPL_LSHIFT_U32(*streamPtr++, 16);
+ streamval = (uint32_t)(*streamPtr++) << 16;
streamval |= *streamPtr++;
} else {
streamval = streamData->streamval;
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 4efdecc..5133128 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
@@ -184,21 +184,20 @@
* W_upper < 2^24 */
while ( !(W_upper & 0xFF000000) )
{
- W_upper = WEBRTC_SPL_LSHIFT_U32(W_upper, 8);
+ W_upper <<= 8;
if (streamData->full == 0) {
*streamPtr++ += (uint16_t) WEBRTC_SPL_RSHIFT_U32(
streamData->streamval, 24);
streamData->full = 1;
} else {
- *streamPtr = (uint16_t) WEBRTC_SPL_LSHIFT_U32(
- WEBRTC_SPL_RSHIFT_U32(streamData->streamval, 24), 8);
+ *streamPtr = (uint16_t)((streamData->streamval >> 24) << 8);
streamData->full = 0;
}
if( streamPtr > maxStreamPtr )
return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
- streamData->streamval = WEBRTC_SPL_LSHIFT_U32(streamData->streamval, 8);
+ streamData->streamval <<= 8;
}
}
@@ -257,7 +256,7 @@
if (streamData->stream_index == 0)
{
/* read first word from bytestream */
- streamVal = WEBRTC_SPL_LSHIFT_U32(*streamPtr++, 16);
+ streamVal = (uint32_t)(*streamPtr++) << 16;
streamVal |= *streamPtr++;
} else {
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 d28a6f7..e69435f 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
@@ -523,9 +523,9 @@
bweStr->recBw = (int32_t) MIN_ISAC_BW;
}
- bweStr->recBwAvg = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw + bweStr->recHeaderRate, 5);
+ bweStr->recBwAvg = (bweStr->recBw + bweStr->recHeaderRate) << 5;
- bweStr->recBwAvgQ = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw, 7);
+ bweStr->recBwAvgQ = bweStr->recBw << 7;
bweStr->recJitterShortTerm = 0;
@@ -573,8 +573,8 @@
/* compute the BN estimate as decoded on the other side */
/* sendBwAvg = 0.9 * sendBwAvg + 0.1 * kQRateTable[RateInd]; */
- bweStr->sendBwAvg = WEBRTC_SPL_UMUL(461, bweStr->sendBwAvg) +
- WEBRTC_SPL_UMUL(51, WEBRTC_SPL_LSHIFT_U32(kQRateTable[RateInd], 7));
+ bweStr->sendBwAvg = 461 * bweStr->sendBwAvg +
+ 51 * ((uint32_t)kQRateTable[RateInd] << 7);
bweStr->sendBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 9);
@@ -625,8 +625,8 @@
/* Compute the averaged BN estimate on this side */
/* recBwAvg = 0.9 * recBwAvg + 0.1 * (rate + bweStr->recHeaderRate), 0.9 and 0.1 in Q9 */
- bweStr->recBwAvg = WEBRTC_SPL_UMUL(922, bweStr->recBwAvg) +
- WEBRTC_SPL_UMUL(102, WEBRTC_SPL_LSHIFT_U32((uint32_t)rate + bweStr->recHeaderRate, 5));
+ bweStr->recBwAvg = 922 * bweStr->recBwAvg +
+ 102 * (((uint32_t)rate + bweStr->recHeaderRate) << 5);
bweStr->recBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 10);
/* Find quantization index that gives the closest rate after averaging.
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
index 1d36ff0..06c15cb 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
@@ -102,7 +102,7 @@
for (byte_cntr = 0; byte_cntr < len_bitstream_in_bytes; byte_cntr++) {
crc_tbl_indx = (WEBRTC_SPL_RSHIFT_U32(crc_state, 24) ^
bitstream_ptr_uw8[byte_cntr]) & 0xFF;
- crc_state = WEBRTC_SPL_LSHIFT_U32(crc_state, 8) ^ kCrcTable[crc_tbl_indx];
+ crc_state = (crc_state << 8) ^ kCrcTable[crc_tbl_indx];
}
*crc = ~crc_state;
