Match existing type usage better.
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.cc b/talk/app/webrtc/test/fakeaudiocapturemodule.cc
index c6339d3..e52b686 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.cc
@@ -45,7 +45,7 @@
// Constants here are derived by running VoE using a real ADM.
// The constants correspond to 10ms of mono audio at 44kHz.
static const int kTimePerFrameMs = 10;
-static const int kNumberOfChannels = 1;
+static const uint8_t kNumberOfChannels = 1;
static const int kSamplesPerSecond = 44000;
static const int kTotalDelayMs = 0;
static const int kClockDriftMs = 0;
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
index 8fcbfd7..821b44c 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
@@ -66,7 +66,7 @@
const bool keyPressed,
uint32_t& newMicLevel) {
rec_buffer_bytes_ = nSamples * nBytesPerSample;
- if ((rec_buffer_bytes_ <= 0) ||
+ if ((rec_buffer_bytes_ == 0) ||
(rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
FakeAudioCaptureModule::kNumberBytesPerSample)) {
ADD_FAILURE();
diff --git a/webrtc/common_audio/fft4g.c b/webrtc/common_audio/fft4g.c
index cbc4dc3..ad5f383 100644
--- a/webrtc/common_audio/fft4g.c
+++ b/webrtc/common_audio/fft4g.c
@@ -648,7 +648,7 @@
ip[1] = 1;
if (nw > 2) {
nwh = nw >> 1;
- delta = (float)atan(1.0f) / nwh;
+ delta = atanf(1.0f) / nwh;
w[0] = 1;
w[1] = 0;
w[nwh] = (float)cos(delta * nwh);
@@ -676,7 +676,7 @@
ip[1] = nc;
if (nc > 1) {
nch = nc >> 1;
- delta = (float)atan(1.0f) / nch;
+ delta = atanf(1.0f) / nch;
c[0] = (float)cos(delta * nch);
c[nch] = 0.5f * c[0];
for (j = 1; j < nch; j++) {
diff --git a/webrtc/common_audio/lapped_transform_unittest.cc b/webrtc/common_audio/lapped_transform_unittest.cc
index c30651c..3becfe1 100644
--- a/webrtc/common_audio/lapped_transform_unittest.cc
+++ b/webrtc/common_audio/lapped_transform_unittest.cc
@@ -51,11 +51,12 @@
complex<float>* const* out_block) {
CHECK_EQ(in_channels, out_channels);
- float full_length = (frames - 1) * 2;
+ int full_length = (frames - 1) * 2;
++block_num_;
if (block_num_ > 0) {
- ASSERT_NEAR(in_block[0][0].real(), full_length, 1e-5f);
+ ASSERT_NEAR(in_block[0][0].real(), static_cast<float>(full_length),
+ 1e-5f);
ASSERT_NEAR(in_block[0][0].imag(), 0.0f, 1e-5f);
for (int i = 1; i < frames; ++i) {
ASSERT_NEAR(in_block[0][i].real(), 0.0f, 1e-5f);
diff --git a/webrtc/common_audio/real_fourier.cc b/webrtc/common_audio/real_fourier.cc
index 30c8ee3..cb707e4 100644
--- a/webrtc/common_audio/real_fourier.cc
+++ b/webrtc/common_audio/real_fourier.cc
@@ -31,7 +31,7 @@
int RealFourier::FftOrder(int length) {
CHECK_GT(length, 0);
- return WebRtcSpl_GetSizeInBits(length - 1);
+ return WebRtcSpl_GetSizeInBits(static_cast<uint32_t>(length - 1));
}
int RealFourier::FftLength(int order) {
diff --git a/webrtc/common_audio/signal_processing/auto_correlation.c b/webrtc/common_audio/signal_processing/auto_correlation.c
index fed1312..405a08e 100644
--- a/webrtc/common_audio/signal_processing/auto_correlation.c
+++ b/webrtc/common_audio/signal_processing/auto_correlation.c
@@ -36,7 +36,7 @@
scaling = 0;
} else {
// Number of bits in the sum loop.
- int nbits = WebRtcSpl_GetSizeInBits(in_vector_length);
+ int nbits = WebRtcSpl_GetSizeInBits((uint32_t)in_vector_length);
// Number of bits to normalize smax.
int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
diff --git a/webrtc/common_audio/signal_processing/complex_fft.c b/webrtc/common_audio/signal_processing/complex_fft.c
index aaeda52..f21b7d8 100644
--- a/webrtc/common_audio/signal_processing/complex_fft.c
+++ b/webrtc/common_audio/signal_processing/complex_fft.c
@@ -181,7 +181,7 @@
shift = 0;
round2 = 8192;
- tmp32 = (int32_t)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
+ tmp32 = WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
if (tmp32 > 13573)
{
shift++;
diff --git a/webrtc/common_audio/signal_processing/get_scaling_square.c b/webrtc/common_audio/signal_processing/get_scaling_square.c
index 9b6049c..3b9171d 100644
--- a/webrtc/common_audio/signal_processing/get_scaling_square.c
+++ b/webrtc/common_audio/signal_processing/get_scaling_square.c
@@ -21,7 +21,7 @@
int in_vector_length,
int times)
{
- int16_t nbits = WebRtcSpl_GetSizeInBits(times);
+ int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times);
int i;
int16_t smax = -1;
int16_t sabs;
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index 8135b98..d16dd3b 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -77,7 +77,7 @@
ASSERT_TRUE(cng_) << "Must call CreateCng() first.";
encoded_info_ = cng_->Encode(timestamp_, audio_, num_audio_samples_10ms_,
encoded_.size(), &encoded_[0]);
- timestamp_ += num_audio_samples_10ms_;
+ timestamp_ += static_cast<uint32_t>(num_audio_samples_10ms_);
}
// Expect |num_calls| calls to the encoder, all successful. The last call
diff --git a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
index 9862f12..cb7aa45 100644
--- a/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ b/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
@@ -370,7 +370,7 @@
}
if ((i == 93) && (index == 0))
index = 94;
- SIDdata[0] = index;
+ SIDdata[0] = (uint8_t)index;
/* Quantize coefficients with tweak for WebRtc implementation of RFC3389. */
if (inst->enc_nrOfCoefs == WEBRTC_CNG_MAX_LPC_ORDER) {
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
index a775a02..2ee9f6c 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
@@ -108,8 +108,8 @@
/* Find the highest absolute value to calculate proper
vector scale factor (so that it uses 12 bits) */
- temp1 = WebRtcSpl_MaxAbsValueW16(buf, (int16_t)lMem);
- temp2 = WebRtcSpl_MaxAbsValueW16(target, (int16_t)lTarget);
+ temp1 = WebRtcSpl_MaxAbsValueW16(buf, lMem);
+ temp2 = WebRtcSpl_MaxAbsValueW16(target, lTarget);
if ((temp1>0)&&(temp2>0)) {
temp1 = WEBRTC_SPL_MAX(temp1, temp2);
@@ -332,7 +332,8 @@
/* Subtract the best codebook vector, according
to measure, from the target vector */
- WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain), (int32_t)8192, (int16_t)14, (int)lTarget);
+ WebRtcSpl_AddAffineVectorToVector(target, pp, (int16_t)(-bestGain),
+ (int32_t)8192, (int16_t)14, lTarget);
/* record quantized gain */
gains[stage+1] = bestGain;
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode.c b/webrtc/modules/audio_coding/codecs/ilbc/decode.c
index 3a2e5a2..035460b 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode.c
@@ -206,7 +206,7 @@
}
/* Store lag (it is needed if next packet is lost) */
- (*iLBCdec_inst).last_lag = (int)lag;
+ (*iLBCdec_inst).last_lag = lag;
/* copy data and run synthesis filter */
WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
index c04fd99..de42ea9 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
@@ -66,7 +66,7 @@
/* setup memory */
- WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCdec_inst->state_short_len));
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCdec_inst->state_short_len);
WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
iLBCdec_inst->state_short_len);
@@ -76,8 +76,7 @@
&decresidual[start_pos+iLBCdec_inst->state_short_len],
iLBC_encbits->cb_index, iLBC_encbits->gain_index,
mem+CB_MEML-ST_MEM_L_TBL,
- ST_MEM_L_TBL, (int16_t)diff
- );
+ ST_MEM_L_TBL, diff);
}
else {/* put adaptive part in the beginning */
@@ -87,7 +86,7 @@
meml_gotten = iLBCdec_inst->state_short_len;
WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
decresidual+start_pos, meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
/* construct decoded vector */
@@ -153,7 +152,7 @@
WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
/* loop over subframes to decode */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
index 3de8425..114ce1f 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/encode.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/encode.c
@@ -193,7 +193,7 @@
/* setup memory */
- WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCenc_inst->state_short_len));
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
decresidual+start_pos, iLBCenc_inst->state_short_len);
@@ -224,7 +224,7 @@
meml_gotten = iLBCenc_inst->state_short_len;
WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-iLBCenc_inst->state_short_len));
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML - iLBCenc_inst->state_short_len);
/* encode subframes */
WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
@@ -397,7 +397,7 @@
}
WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (int16_t)(CB_MEML-meml_gotten));
+ WebRtcSpl_MemSetW16(mem, 0, CB_MEML - meml_gotten);
#ifdef SPLIT_10MS
if (iLBCenc_inst->Nback_flag > 0)
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
index f282432..262a564 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
@@ -96,11 +96,11 @@
memmove(enh_period, &enh_period[new_blocks],
(ENH_NBLOCKS_TOT - new_blocks) * sizeof(*enh_period));
- k=WebRtcSpl_DownsampleFast(
+ k = WebRtcSpl_DownsampleFast(
enh_buf+ENH_BUFL-inLen, /* Input samples */
- (int16_t)(inLen+ENH_BUFL_FILTEROVERHEAD),
+ inLen + ENH_BUFL_FILTEROVERHEAD,
downsampled,
- (int16_t)(inLen / 2),
+ inLen / 2,
(int16_t*)WebRtcIlbcfix_kLpFiltCoefs, /* Coefficients in Q12 */
FILTERORDER_DS_PLUS1, /* Length of filter (order-1) */
FACTOR_DS,
@@ -114,8 +114,7 @@
regressor = target - 10;
/* scaling */
- max16=WebRtcSpl_MaxAbsValueW16(®ressor[-50],
- (int16_t)(ENH_BLOCKL_HALF+50-1));
+ max16 = WebRtcSpl_MaxAbsValueW16(®ressor[-50], ENH_BLOCKL_HALF + 50 - 1);
shifts = WebRtcSpl_GetSizeInBits((uint32_t)(max16 * max16)) - 25;
shifts = WEBRTC_SPL_MAX(0, shifts);
@@ -199,7 +198,7 @@
regressor=in+tlag-1;
/* scaling */
- max16=WebRtcSpl_MaxAbsValueW16(regressor, (int16_t)(plc_blockl+3-1));
+ max16 = WebRtcSpl_MaxAbsValueW16(regressor, plc_blockl + 3 - 1);
if (max16>5000)
shifts=2;
else
@@ -338,7 +337,7 @@
synt,
&iLBCdec_inst->old_syntdenum[
(iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
- LPC_FILTERORDER+1, (int16_t)lag);
+ LPC_FILTERORDER+1, lag);
WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
LPC_FILTERORDER);
@@ -349,7 +348,7 @@
enh_bufPtr1, synt,
&iLBCdec_inst->old_syntdenum[
(iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
- LPC_FILTERORDER+1, (int16_t)lag);
+ LPC_FILTERORDER+1, lag);
WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
LPC_FILTERORDER);
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
index d124b6b..6a68dec 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
@@ -62,7 +62,7 @@
}
/* Scale to maximum 20 bits in order to allow for the 11 bit window */
- maxW32 = WebRtcSpl_MaxValueW32(ssqEn, (int16_t)(iLBCenc_inst->nsub-1));
+ maxW32 = WebRtcSpl_MaxValueW32(ssqEn, iLBCenc_inst->nsub - 1);
scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
scale1 = WEBRTC_SPL_MAX(0, scale);
@@ -82,7 +82,7 @@
}
/* Extract the best choise of start state */
- pos = WebRtcSpl_MaxIndexW32(ssqEn, (int16_t)(iLBCenc_inst->nsub-1)) + 1;
+ pos = WebRtcSpl_MaxIndexW32(ssqEn, iLBCenc_inst->nsub - 1) + 1;
return(pos);
}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
index 048745a..ec3cf20 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
@@ -45,7 +45,7 @@
loops=dim1-dim2+1;
/* Calculate the cross correlations */
- WebRtcSpl_CrossCorrelation(corr, (int16_t*)seq2, seq1, dim2, loops, scale, 1);
+ WebRtcSpl_CrossCorrelation(corr, seq2, seq1, dim2, loops, scale, 1);
return;
}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
index 6329908..30c7a03 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
@@ -42,5 +42,5 @@
}
/* Find the minimum square distance */
- *index=WebRtcSpl_MinIndexW32(crit, (int16_t)arlength);
+ *index=WebRtcSpl_MinIndexW32(crit, arlength);
}
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/refiner.c b/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
index ca99b3a..2fff362 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
@@ -75,7 +75,7 @@
/* Calculate the rescaling factor for the correlation in order to
put the correlation in a int16_t vector instead */
- maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, (int16_t)corrdim);
+ maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, corrdim);
scalefact=WebRtcSpl_GetSizeInBits(maxtemp)-15;
@@ -97,7 +97,7 @@
WebRtcIlbcfix_EnhUpsample(corrVecUps,corrVec);
/* Find maximum */
- tloc=WebRtcSpl_MaxIndexW32(corrVecUps, (int16_t) (ENH_UPS0*corrdim));
+ tloc=WebRtcSpl_MaxIndexW32(corrVecUps, ENH_UPS0 * corrdim);
/* make vector can be upsampled without ever running outside
bounds */
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
index 80b3e1b..324b670 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
@@ -100,7 +100,7 @@
WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
WebRtcSpl_FilterARFastQ12(
sampleMa, sampleAr,
- syntDenum, LPC_FILTERORDER+1, (int16_t)(2*len));
+ syntDenum, LPC_FILTERORDER+1, 2 * len);
tmp1 = &sampleAr[len-1];
tmp2 = &sampleAr[2*len-1];
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/state_search.c b/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
index 5d85a84..b2214c7 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
@@ -71,7 +71,7 @@
WebRtcSpl_FilterARFastQ12(
sampleMa, sampleAr,
- syntDenum, LPC_FILTERORDER+1, (int16_t)(2*iLBCenc_inst->state_short_len));
+ syntDenum, LPC_FILTERORDER+1, 2 * iLBCenc_inst->state_short_len);
for(k=0;k<iLBCenc_inst->state_short_len;k++){
sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
index 3490461..328a5fe 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
+++ b/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
@@ -55,11 +55,11 @@
/* Find scale value and start position */
if (step==1) {
- max=WebRtcSpl_MaxAbsValueW16(regressor, (int16_t)(subl+searchLen-1));
+ max=WebRtcSpl_MaxAbsValueW16(regressor, subl + searchLen - 1);
rp_beg = regressor;
rp_end = ®ressor[subl];
} else { /* step==-1 */
- max=WebRtcSpl_MaxAbsValueW16(®ressor[-searchLen], (int16_t)(subl+searchLen-1));
+ max=WebRtcSpl_MaxAbsValueW16(®ressor[-searchLen], subl + searchLen - 1);
rp_beg = ®ressor[-1];
rp_end = ®ressor[subl-1];
}
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 16befba..4a4cddc 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
@@ -374,7 +374,7 @@
/* compute inverse receiving rate for last packet, in Q19 */
numBytesInv = (uint16_t) WebRtcSpl_DivW32W16(
524288 + ((pksize + HEADER_SIZE) >> 1),
- pksize + HEADER_SIZE);
+ (int16_t)(pksize + HEADER_SIZE));
/* 8389 is ~ 1/128000 in Q30 */
byteSecondsPerBit = (uint32_t)(arrTimeDiff * 8389);
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
index d2cfb3a..1a7ff92 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
@@ -447,7 +447,7 @@
/* inverse pitch filter */
pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] =
- ((ISACdec_obj->plcstr_obj).stretchLag<<7);
+ (int16_t)((ISACdec_obj->plcstr_obj).stretchLag<<7);
pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12);
pitchGains_Q12[2] = (int16_t)(pitchGains_Q12[3] * 1010 >> 10);
pitchGains_Q12[1] = (int16_t)(pitchGains_Q12[2] * 1010 >> 10);
@@ -749,7 +749,8 @@
k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0;
}
- (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = (ISACdec_obj->plcstr_obj).stretchLag << 7;
+ (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 =
+ (int16_t)((ISACdec_obj->plcstr_obj).stretchLag << 7);
/* --- Inverse Pitch Filter --- */
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
index 1a6372a..757c0b8 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/encode.c
@@ -498,7 +498,7 @@
{
int ii;
int status;
- int16_t BWno = BWnumber;
+ int16_t BWno = (int16_t)BWnumber;
int stream_length = 0;
int16_t model;
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
index f8abc8a..03bceec 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.c
@@ -425,7 +425,8 @@
return -1;
}
- write_be16(ISAC_inst->ISACenc_obj.bitstr_obj.stream, stream_len, encoded);
+ write_be16(ISAC_inst->ISACenc_obj.bitstr_obj.stream, (size_t)stream_len,
+ encoded);
return stream_len;
}
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
index a7a80ab..7f4272b 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -62,7 +62,8 @@
/* everything in samples */
BN_data->sample_count = BN_data->sample_count + current_framesamples;
- BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+ BN_data->arrival_time += static_cast<uint32_t>(
+ ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate));
BN_data->send_time += current_framesamples;
if (BN_data->arrival_time < BN_data->sample_count)
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
index b4c0ee4..e2a778a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/test/test_iSACfixfloat.c
@@ -68,8 +68,8 @@
/* everything in samples */
BN_data->sample_count = BN_data->sample_count + current_framesamples;
- BN_data->arrival_time +=
- ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
+ BN_data->arrival_time += (uint32_t)
+ (((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate));
BN_data->send_time += current_framesamples;
if (BN_data->arrival_time < BN_data->sample_count)
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
index db78e6d..3492bfa 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
@@ -504,7 +504,7 @@
int16_t streamLenUB = 0;
int16_t streamLen = 0;
int16_t k = 0;
- int garbageLen = 0;
+ uint8_t garbageLen = 0;
int32_t bottleneck = 0;
int16_t bottleneckIdx = 0;
int16_t jitterInfo = 0;
@@ -645,7 +645,7 @@
memcpy(encoded, instLB->ISACencLB_obj.bitstr_obj.stream, streamLenLB);
streamLen = streamLenLB;
if (streamLenUB > 0) {
- encoded[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
+ encoded[streamLenLB] = (uint8_t)(streamLenUB + 1 + LEN_CHECK_SUM_WORD8);
memcpy(&encoded[streamLenLB + 1],
instUB->ISACencUB_obj.bitstr_obj.stream,
streamLenUB);
@@ -703,7 +703,7 @@
}
minBytes = (minBytes > limit) ? limit : minBytes;
- garbageLen = (minBytes > streamLen) ? (minBytes - streamLen) : 0;
+ garbageLen = (minBytes > streamLen) ? (uint8_t)(minBytes - streamLen) : 0;
/* Save data for creation of multiple bit-streams. */
/* If bit-stream too short then add garbage at the end. */
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index c564991..0574047 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -52,7 +52,8 @@
double starttime, runtime, length_file;
int16_t stream_len = 0;
- int16_t declen = 0, lostFrame = 0, declenTC = 0;
+ int16_t declen = 0, declenTC = 0;
+ bool lostFrame = false;
int16_t shortdata[SWBFRAMESAMPLES_10ms];
int16_t vaddata[SWBFRAMESAMPLES_10ms * 3];
@@ -696,7 +697,7 @@
if (!lostFrame) {
lostFrame = ((rand() % 100) < packetLossPercent);
} else {
- lostFrame = 0;
+ lostFrame = false;
}
// RED.
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
index 8f5b4cf..7ea8bae 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
@@ -98,7 +98,7 @@
char histFileName[500];
char averageFileName[500];
unsigned int hist[600];
- unsigned int tmpSumStreamLen = 0;
+ double tmpSumStreamLen = 0;
unsigned int packetCntr = 0;
unsigned int lostPacketCntr = 0;
uint8_t payload[1200];
@@ -374,7 +374,7 @@
if (packetCntr == 100) {
// kbps
fprintf(averageFile, "%8.3f ",
- (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
+ tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
packetCntr = 0;
tmpSumStreamLen = 0;
}
@@ -493,7 +493,7 @@
if (averageFile != NULL) {
if (packetCntr > 0) {
fprintf(averageFile, "%8.3f ",
- (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
+ tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
}
fprintf(averageFile, "\n");
fclose(averageFile);
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index c05d773..e69b0c8 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -115,9 +115,9 @@
// Calculate the number of bytes we expect the encoder to produce,
// then multiply by two to give a wide margin for error.
int frame_size_ms = num_10ms_frames_per_packet_ * 10;
- int bytes_per_millisecond = bitrate_bps_ / (1000 * 8) + 1;
- size_t approx_encoded_bytes =
- static_cast<size_t>(frame_size_ms * bytes_per_millisecond);
+ size_t bytes_per_millisecond =
+ static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
+ size_t approx_encoded_bytes = frame_size_ms * bytes_per_millisecond;
return 2 * approx_encoded_bytes;
}
@@ -206,7 +206,7 @@
CHECK_GE(status, 0); // Fails only if fed invalid data.
input_buffer_.clear();
EncodedInfo info;
- info.encoded_bytes = status;
+ info.encoded_bytes = static_cast<size_t>(status);
info.encoded_timestamp = first_timestamp_in_buffer_;
info.payload_type = payload_type_;
info.send_even_if_empty = true; // Allows Opus to send empty packets.
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index a30b1cb..aaaced1 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
@@ -196,7 +196,7 @@
EncodeABlock();
// Check if payload has FEC.
- int16_t fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
+ int fec = WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_);
// If FEC is disabled or the target packet loss rate is set to 0, there
// should be no FEC in the bit stream.
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index f9cf89a..ae5a04f 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -461,8 +461,8 @@
// |audio_frame|.
uint32_t playout_timestamp = 0;
if (GetPlayoutTimestamp(&playout_timestamp)) {
- audio_frame->timestamp_ =
- playout_timestamp - audio_frame->samples_per_channel_;
+ audio_frame->timestamp_ = playout_timestamp -
+ static_cast<uint32_t>(audio_frame->samples_per_channel_);
} else {
// Remain 0 until we have a valid |playout_timestamp|.
audio_frame->timestamp_ = 0;
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
index 56830a4..b96db6b 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
@@ -79,7 +79,7 @@
}
int32_t encoded_bytes = acm_->Add10MsAudio(input_frame_);
EXPECT_GE(encoded_bytes, 0);
- input_frame_.timestamp_ += input_block_size_samples_;
+ input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
if (encoded_bytes > 0) {
// Encoded packet received.
return CreatePacket();
diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
index d0c031e..1819d59 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
@@ -92,7 +92,7 @@
}
data_to_send_ = false;
CHECK_GE(acm_->Add10MsData(input_frame_), 0);
- input_frame_.timestamp_ += input_block_size_samples_;
+ input_frame_.timestamp_ += static_cast<uint32_t>(input_block_size_samples_);
if (data_to_send_) {
// Encoded packet received.
return CreatePacket();
diff --git a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
index b659649..ce98636 100644
--- a/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
+++ b/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
@@ -431,8 +431,8 @@
if (!down_mix && !resample) {
// No pre-processing is required.
- expected_in_ts_ += in_frame.samples_per_channel_;
- expected_codec_ts_ += in_frame.samples_per_channel_;
+ expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
+ expected_codec_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
*ptr_out = &in_frame;
return 0;
}
@@ -477,8 +477,9 @@
codec_manager_.CurrentEncoder()->SampleRateHz();
}
- expected_codec_ts_ += preprocess_frame_.samples_per_channel_;
- expected_in_ts_ += in_frame.samples_per_channel_;
+ expected_codec_ts_ +=
+ static_cast<uint32_t>(preprocess_frame_.samples_per_channel_);
+ expected_in_ts_ += static_cast<uint32_t>(in_frame.samples_per_channel_);
return 0;
}
diff --git a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
index 3d0d312..ffbbc8c 100644
--- a/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
+++ b/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
@@ -144,7 +144,7 @@
acm_b_->SetInitialPlayoutDelay(initial_delay_ms);
while (rms < kAmp / 2) {
in_audio_frame.timestamp_ = timestamp;
- timestamp += in_audio_frame.samples_per_channel_;
+ timestamp += static_cast<uint32_t>(in_audio_frame.samples_per_channel_);
ASSERT_GE(acm_a_->Add10MsData(in_audio_frame), 0);
ASSERT_EQ(0, acm_b_->PlayoutData10Ms(codec.plfreq, &out_audio_frame));
rms = FrameRms(out_audio_frame);
diff --git a/webrtc/modules/audio_coding/neteq/background_noise.cc b/webrtc/modules/audio_coding/neteq/background_noise.cc
index 4fbc84c..a59f444 100644
--- a/webrtc/modules/audio_coding/neteq/background_noise.cc
+++ b/webrtc/modules/audio_coding/neteq/background_noise.cc
@@ -239,7 +239,7 @@
parameters.low_energy_update_threshold = 0;
// Normalize residual_energy to 29 or 30 bits before sqrt.
- int norm_shift = WebRtcSpl_NormW32(residual_energy) - 1;
+ int16_t norm_shift = WebRtcSpl_NormW32(residual_energy) - 1;
if (norm_shift & 0x1) {
norm_shift -= 1; // Even number of shifts required.
}
@@ -251,7 +251,8 @@
// Add 13 to the |scale_shift_|, since the random numbers table is in
// Q13.
// TODO(hlundin): Move the "13" to where the |scale_shift_| is used?
- parameters.scale_shift = 13 + ((kLogResidualLength + norm_shift) / 2);
+ parameters.scale_shift =
+ static_cast<int16_t>(13 + ((kLogResidualLength + norm_shift) / 2));
initialized_ = true;
}
diff --git a/webrtc/modules/audio_coding/neteq/background_noise.h b/webrtc/modules/audio_coding/neteq/background_noise.h
index fd4e6a5..baf1818 100644
--- a/webrtc/modules/audio_coding/neteq/background_noise.h
+++ b/webrtc/modules/audio_coding/neteq/background_noise.h
@@ -79,7 +79,7 @@
static const int kVecLen = 256;
static const int kLogVecLen = 8; // log2(kVecLen).
static const int kResidualLength = 64;
- static const int kLogResidualLength = 6; // log2(kResidualLength)
+ static const int16_t kLogResidualLength = 6; // log2(kResidualLength)
struct ChannelParameters {
// Constructor.
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
index 89fdb51..e985ee0 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
@@ -67,7 +67,8 @@
return kNormal;
}
- const uint32_t five_seconds_samples = 5 * 8000 * fs_mult_;
+ const uint32_t five_seconds_samples =
+ static_cast<uint32_t>(5 * 8000 * fs_mult_);
// Check if the required packet is available.
if (target_timestamp == available_timestamp) {
return ExpectedPacketAvailable(prev_mode, play_dtmf);
@@ -87,10 +88,11 @@
uint32_t target_timestamp,
uint32_t available_timestamp) {
// Signed difference between target and available timestamp.
- int32_t timestamp_diff = (generated_noise_samples_ + target_timestamp) -
- available_timestamp;
- int32_t optimal_level_samp =
- (delay_manager_->TargetLevel() * packet_length_samples_) >> 8;
+ int32_t timestamp_diff = static_cast<int32_t>(
+ static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) -
+ available_timestamp);
+ int32_t optimal_level_samp = static_cast<int32_t>(
+ (delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
int32_t excess_waiting_time_samp = -timestamp_diff - optimal_level_samp;
if (excess_waiting_time_samp > optimal_level_samp / 2) {
@@ -182,11 +184,11 @@
// safety precaution), but make sure that the number of samples in buffer
// is no higher than 4 times the optimal level. (Note that TargetLevel()
// is in Q8.)
- int32_t timestamp_diff = (generated_noise_samples_ + target_timestamp) -
- available_timestamp;
- if (timestamp_diff >= 0 ||
+ if (static_cast<uint32_t>(generated_noise_samples_ + target_timestamp) >=
+ available_timestamp ||
cur_size_samples >
- 4 * ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8)) {
+ ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
+ 4) {
// Time to play this new packet.
return kNormal;
} else {
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index 1378241..d5f0f9c 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -227,7 +227,7 @@
if (mix_factor_increment != 0) {
parameters.current_voice_mix_factor = parameters.voice_mix_factor;
}
- int temp_scale = 16384 - parameters.current_voice_mix_factor;
+ int16_t temp_scale = 16384 - parameters.current_voice_mix_factor;
WebRtcSpl_ScaleAndAddVectorsWithRound(
voiced_vector + temp_lenght, parameters.current_voice_mix_factor,
unvoiced_vector + temp_lenght, temp_scale, 14,
@@ -669,7 +669,8 @@
// even, which is suitable for the sqrt.
unvoiced_scale += ((unvoiced_scale & 0x1) ^ 0x1);
unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
- int32_t unvoiced_gain = WebRtcSpl_SqrtFloor(unvoiced_energy);
+ int16_t unvoiced_gain =
+ static_cast<int16_t>(WebRtcSpl_SqrtFloor(unvoiced_energy));
parameters.ar_gain_scale = 13
+ (unvoiced_scale + 7 - unvoiced_prescale) / 2;
parameters.ar_gain = unvoiced_gain;
@@ -709,8 +710,9 @@
// the division.
// Shift the denominator from Q13 to Q5 before the division. The result of
// the division will then be in Q20.
- int16_t temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12,
- (distortion_lag * slope) >> 8);
+ int16_t temp_ratio = WebRtcSpl_DivW32W16(
+ (slope - 8192) << 12,
+ static_cast<int16_t>((distortion_lag * slope) >> 8));
if (slope > 14746) {
// slope > 1.8.
// Divide by 2, with proper rounding.
@@ -723,8 +725,8 @@
} else {
// Calculate (1 - slope) / distortion_lag.
// Shift |slope| by 7 to Q20 before the division. The result is in Q20.
- parameters.mute_slope = WebRtcSpl_DivW32W16((8192 - slope) << 7,
- distortion_lag);
+ parameters.mute_slope = WebRtcSpl_DivW32W16(
+ (8192 - slope) << 7, static_cast<int16_t>(distortion_lag));
if (parameters.voice_mix_factor <= 13107) {
// Make sure the mute factor decreases from 1.0 to 0.9 in no more than
// 6.25 ms.
@@ -810,7 +812,8 @@
// Normalize and move data from 32-bit to 16-bit vector.
int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
kNumCorrelationLags);
- int16_t norm_shift2 = std::max(18 - WebRtcSpl_NormW32(max_correlation), 0);
+ int16_t norm_shift2 = static_cast<int16_t>(
+ std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
norm_shift2);
// Total scale factor (right shifts) of correlation value.
@@ -928,7 +931,7 @@
}
}
-void Expand::GenerateRandomVector(int seed_increment,
+void Expand::GenerateRandomVector(int16_t seed_increment,
size_t length,
int16_t* random_vector) {
// TODO(turajs): According to hlundin The loop should not be needed. Should be
diff --git a/webrtc/modules/audio_coding/neteq/expand.h b/webrtc/modules/audio_coding/neteq/expand.h
index 5679ec1..0000642 100644
--- a/webrtc/modules/audio_coding/neteq/expand.h
+++ b/webrtc/modules/audio_coding/neteq/expand.h
@@ -66,7 +66,7 @@
protected:
static const int kMaxConsecutiveExpands = 200;
- void GenerateRandomVector(int seed_increment,
+ void GenerateRandomVector(int16_t seed_increment,
size_t length,
int16_t* random_vector);
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index 44fc511..8399a78 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -108,10 +108,11 @@
// Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
// and so on.
int increment = 4194 / fs_mult_;
- *external_mute_factor = DspHelper::RampSignal(input_channel,
- interpolation_length,
- *external_mute_factor,
- increment);
+ *external_mute_factor =
+ static_cast<int16_t>(DspHelper::RampSignal(input_channel,
+ interpolation_length,
+ *external_mute_factor,
+ increment));
DspHelper::UnmuteSignal(&input_channel[interpolation_length],
input_length_per_channel - interpolation_length,
external_mute_factor, increment,
@@ -125,7 +126,8 @@
}
// Do overlap and mix linearly.
- int increment = 16384 / (interpolation_length + 1); // In Q14.
+ int16_t increment =
+ static_cast<int16_t>(16384 / (interpolation_length + 1)); // In Q14.
int16_t mute_factor = 16384 - increment;
memmove(temp_data, expanded_channel,
sizeof(int16_t) * best_correlation_index);
@@ -246,7 +248,8 @@
// energy_expanded / energy_input is in Q14.
energy_expanded = WEBRTC_SPL_SHIFT_W32(energy_expanded, temp_shift + 14);
// Calculate sqrt(energy_expanded / energy_input) in Q14.
- mute_factor = WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14);
+ mute_factor = static_cast<int16_t>(
+ WebRtcSpl_SqrtFloor((energy_expanded / energy_input) << 14));
} else {
// Set to 1 (in Q14) when |expanded| has higher energy than |input|.
mute_factor = 16384;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 6512515..3a3ad98 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -788,7 +788,8 @@
}
case kAudioRepetitionIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
- sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
+ sync_buffer_->IncreaseEndTimestamp(
+ static_cast<uint32_t>(output_size_samples_));
// Skipping break on purpose. Execution should move on into the
// next case.
FALLTHROUGH();
@@ -881,7 +882,7 @@
}
} else {
// Use dead reckoning to estimate the |playout_timestamp_|.
- playout_timestamp_ += output_size_samples_;
+ playout_timestamp_ += static_cast<uint32_t>(output_size_samples_);
}
if (decode_return_value) return decode_return_value;
@@ -940,9 +941,10 @@
}
// Check if it is time to play a DTMF event.
- if (dtmf_buffer_->GetEvent(end_timestamp +
- decision_logic_->generated_noise_samples(),
- dtmf_event)) {
+ if (dtmf_buffer_->GetEvent(
+ static_cast<uint32_t>(
+ end_timestamp + decision_logic_->generated_noise_samples()),
+ dtmf_event)) {
*play_dtmf = true;
}
@@ -1030,7 +1032,8 @@
if (decision_logic_->generated_noise_samples() > 0 &&
last_mode_ != kModeDtmf) {
// Make a jump in timestamp due to the recently played comfort noise.
- uint32_t timestamp_jump = decision_logic_->generated_noise_samples();
+ uint32_t timestamp_jump =
+ static_cast<uint32_t>(decision_logic_->generated_noise_samples());
sync_buffer_->IncreaseEndTimestamp(timestamp_jump);
timestamp_ += timestamp_jump;
}
@@ -1224,7 +1227,8 @@
if (*decoded_length < 0) {
// Error returned from the decoder.
*decoded_length = 0;
- sync_buffer_->IncreaseEndTimestamp(decoder_frame_length_);
+ sync_buffer_->IncreaseEndTimestamp(
+ static_cast<uint32_t>(decoder_frame_length_));
int error_code = 0;
if (decoder)
error_code = decoder->ErrorCode();
@@ -1719,7 +1723,8 @@
// algorithm_buffer_->PopFront(sync_buffer_->FutureLength());
// }
- sync_buffer_->IncreaseEndTimestamp(output_size_samples_);
+ sync_buffer_->IncreaseEndTimestamp(
+ static_cast<uint32_t>(output_size_samples_));
expand_->Reset();
last_mode_ = kModeDtmf;
@@ -1749,7 +1754,7 @@
stats_.AddZeros(length);
}
if (increase_timestamp) {
- sync_buffer_->IncreaseEndTimestamp(length);
+ sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
}
expand_->Reset();
}
diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
index 3bdaa69b..8a66262 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -343,7 +343,8 @@
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_header, packet_->payload(),
packet_->payload_length_bytes(),
- packet_->time_ms() * (output_sample_rate_ / 1000)));
+ static_cast<uint32_t>(
+ packet_->time_ms() * (output_sample_rate_ / 1000))));
}
// Get next packet.
packet_.reset(rtp_source_->NextPacket());
diff --git a/webrtc/modules/audio_coding/neteq/normal.cc b/webrtc/modules/audio_coding/neteq/normal.cc
index 18ba79b..a0e5d2d 100644
--- a/webrtc/modules/audio_coding/neteq/normal.cc
+++ b/webrtc/modules/audio_coding/neteq/normal.cc
@@ -50,7 +50,7 @@
// fs_shift = log2(fs_mult), rounded down.
// Note that |fs_shift| is not "exact" for 48 kHz.
// TODO(hlundin): Investigate this further.
- const int fs_shift = 30 - WebRtcSpl_NormW32(fs_mult);
+ const int fs_shift = 30 - WebRtcSpl_NormW32(static_cast<int32_t>(fs_mult));
// Check if last RecOut call resulted in an Expand. If so, we have to take
// care of some cross-fading and unmuting.
@@ -99,14 +99,15 @@
// We want background_noise_.energy() / energy in Q14.
int32_t bgn_energy =
background_noise_.Energy(channel_ix) << (scaling+14);
- int16_t energy_scaled = energy << scaling;
- int16_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
- mute_factor = WebRtcSpl_SqrtFloor(static_cast<int32_t>(ratio) << 14);
+ int16_t energy_scaled = static_cast<int16_t>(energy << scaling);
+ int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
+ mute_factor = WebRtcSpl_SqrtFloor(ratio << 14);
} else {
mute_factor = 16384; // 1.0 in Q14.
}
if (mute_factor > external_mute_factor_array[channel_ix]) {
- external_mute_factor_array[channel_ix] = std::min(mute_factor, 16384);
+ external_mute_factor_array[channel_ix] =
+ static_cast<int16_t>(std::min(mute_factor, 16384));
}
// If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
@@ -118,10 +119,11 @@
int32_t scaled_signal = (*output)[channel_ix][i] *
external_mute_factor_array[channel_ix];
// Shift 14 with proper rounding.
- (*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
+ (*output)[channel_ix][i] =
+ static_cast<int16_t>((scaled_signal + 8192) >> 14);
// Increase mute_factor towards 16384.
- external_mute_factor_array[channel_ix] =
- std::min(external_mute_factor_array[channel_ix] + increment, 16384);
+ external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
+ external_mute_factor_array[channel_ix] + increment, 16384));
}
// Interpolate the expanded data into the new vector.
@@ -135,8 +137,8 @@
assert(channel_ix < output->Channels());
assert(i < output->Size());
(*output)[channel_ix][i] =
- (fraction * (*output)[channel_ix][i] +
- (32 - fraction) * expanded[channel_ix][i] + 8) >> 5;
+ static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
+ (32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
fraction += increment;
}
}
@@ -187,10 +189,11 @@
int32_t scaled_signal = (*output)[channel_ix][i] *
external_mute_factor_array[channel_ix];
// Shift 14 with proper rounding.
- (*output)[channel_ix][i] = (scaled_signal + 8192) >> 14;
+ (*output)[channel_ix][i] =
+ static_cast<int16_t>((scaled_signal + 8192) >> 14);
// Increase mute_factor towards 16384.
- external_mute_factor_array[channel_ix] =
- std::min(16384, external_mute_factor_array[channel_ix] + increment);
+ external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
+ 16384, external_mute_factor_array[channel_ix] + increment));
}
}
}
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index 14e9385..f637eb8 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -83,7 +83,7 @@
}
void StatisticsCalculator::IncreaseCounter(int num_samples, int fs_hz) {
- timestamps_since_last_report_ += num_samples;
+ timestamps_since_last_report_ += static_cast<uint32_t>(num_samples);
if (timestamps_since_last_report_ >
static_cast<uint32_t>(fs_hz * kMaxReportPeriod)) {
lost_timestamps_ = 0;
@@ -121,7 +121,8 @@
}
stats->added_zero_samples = added_zero_samples_;
- stats->current_buffer_size_ms = num_samples_in_buffers * 1000 / fs_hz;
+ stats->current_buffer_size_ms =
+ static_cast<uint16_t>(num_samples_in_buffers * 1000 / fs_hz);
const int ms_per_packet = decision_logic.packet_length_samples() /
(fs_hz / 1000);
stats->preferred_buffer_size_ms = (delay_manager.TargetLevel() >> 8) *
@@ -167,14 +168,14 @@
ResetWaitingTimeStatistics();
}
-int StatisticsCalculator::CalculateQ14Ratio(uint32_t numerator,
- uint32_t denominator) {
+uint16_t StatisticsCalculator::CalculateQ14Ratio(uint32_t numerator,
+ uint32_t denominator) {
if (numerator == 0) {
return 0;
} else if (numerator < denominator) {
// Ratio must be smaller than 1 in Q14.
assert((numerator << 14) / denominator < (1 << 14));
- return (numerator << 14) / denominator;
+ return static_cast<uint16_t>((numerator << 14) / denominator);
} else {
// Will not produce a ratio larger than 1, since this is probably an error.
return 1 << 14;
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.h b/webrtc/modules/audio_coding/neteq/statistics_calculator.h
index cd4d867..a2cd9be 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.h
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.h
@@ -91,7 +91,7 @@
static const int kLenWaitingTimes = 100;
// Calculates numerator / denominator, and returns the value in Q14.
- static int CalculateQ14Ratio(uint32_t numerator, uint32_t denominator);
+ static uint16_t CalculateQ14Ratio(uint32_t numerator, uint32_t denominator);
uint32_t preemptive_samples_;
uint32_t accelerate_samples_;
diff --git a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
index c097f5f..192d374 100644
--- a/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
+++ b/webrtc/modules/audio_coding/neteq/test/RTPencode.cc
@@ -621,8 +621,8 @@
}
/* write RTP packet to file */
- length = htons(12 + enc_len + 8);
- plen = htons(12 + enc_len);
+ length = htons(static_cast<unsigned short>(12 + enc_len + 8));
+ plen = htons(static_cast<unsigned short>(12 + enc_len));
offset = (uint32_t)sendtime; //(timestamp/(fs/1000));
offset = htonl(offset);
if (fwrite(&length, 2, 1, out_file) != 1) {
@@ -673,7 +673,7 @@
memmove(&rtp_data[RTPheaderLen + red_len[0]], &rtp_data[12], enc_len);
memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
- red_len[1] = enc_len;
+ red_len[1] = static_cast<uint16_t>(enc_len);
red_TS[1] = timestamp;
if (vad)
red_PT[1] = payloadType;
@@ -689,7 +689,7 @@
memmove(&rtp_data[RTPheaderLen - 4], &rtp_data[12], enc_len);
// memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
- red_len[1] = enc_len;
+ red_len[1] = static_cast<uint16_t>(enc_len);
red_TS[1] = timestamp;
if (vad)
red_PT[1] = payloadType;
@@ -714,8 +714,8 @@
do {
#endif // MULTIPLE_SAME_TIMESTAMP
/* write RTP packet to file */
- length = htons(12 + enc_len + 8);
- plen = htons(12 + enc_len);
+ length = htons(static_cast<unsigned short>(12 + enc_len + 8));
+ plen = htons(static_cast<unsigned short>(12 + enc_len));
offset = (uint32_t)sendtime;
//(timestamp/(fs/1000));
offset = htonl(offset);
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index febacbc..18a242f 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -563,7 +563,7 @@
}
}
- return nSamplesOut;
+ return static_cast<int32_t>(nSamplesOut);
}
// ----------------------------------------------------------------------------
@@ -590,7 +590,7 @@
_playFile.Write(&_playBuffer[0], _playSize);
}
- return _playSamples;
+ return static_cast<int32_t>(_playSamples);
}
} // namespace webrtc
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc
index 82569e8..3de5344 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc
@@ -172,7 +172,7 @@
return -1;
}
- _recordingFramesIn10MS = kRecordingFixedSampleRate/100;
+ _recordingFramesIn10MS = static_cast<uint32_t>(kRecordingFixedSampleRate/100);
if (_ptrAudioBuffer) {
_ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);
@@ -190,7 +190,7 @@
return 0;
}
- _playoutFramesIn10MS = kPlayoutFixedSampleRate/100;
+ _playoutFramesIn10MS = static_cast<uint32_t>(kPlayoutFixedSampleRate/100);
_playing = true;
_playoutFramesLeft = 0;
diff --git a/webrtc/modules/audio_processing/ns/ns_core.c b/webrtc/modules/audio_processing/ns/ns_core.c
index 9e230dd..1bd7af4 100644
--- a/webrtc/modules/audio_processing/ns/ns_core.c
+++ b/webrtc/modules/audio_processing/ns/ns_core.c
@@ -898,10 +898,10 @@
imag[0] = 0;
real[0] = time_data[0];
- magn[0] = fabs(real[0]) + 1.f;
+ magn[0] = fabsf(real[0]) + 1.f;
imag[magnitude_length - 1] = 0;
real[magnitude_length - 1] = time_data[1];
- magn[magnitude_length - 1] = fabs(real[magnitude_length - 1]) + 1.f;
+ magn[magnitude_length - 1] = fabsf(real[magnitude_length - 1]) + 1.f;
for (i = 1; i < magnitude_length - 1; ++i) {
real[i] = time_data[2 * i];
imag[i] = time_data[2 * i + 1];
@@ -1090,10 +1090,10 @@
sumMagn += magn[i];
if (self->blockInd < END_STARTUP_SHORT) {
if (i >= kStartBand) {
- tmpFloat2 = log((float)i);
+ tmpFloat2 = logf((float)i);
sum_log_i += tmpFloat2;
sum_log_i_square += tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[i]);
+ tmpFloat1 = logf(magn[i]);
sum_log_magn += tmpFloat1;
sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
}
@@ -1136,7 +1136,7 @@
if (self->pinkNoiseExp > 0.f) {
// Use pink noise estimate.
parametric_num =
- exp(self->pinkNoiseNumerator / (float)(self->blockInd + 1));
+ expf(self->pinkNoiseNumerator / (float)(self->blockInd + 1));
parametric_num *= (float)(self->blockInd + 1);
parametric_exp = self->pinkNoiseExp / (float)(self->blockInd + 1);
}
@@ -1150,7 +1150,7 @@
// Use pink noise estimate.
float use_band = (float)(i < kStartBand ? kStartBand : i);
self->parametricNoise[i] =
- parametric_num / pow(use_band, parametric_exp);
+ parametric_num / powf(use_band, parametric_exp);
}
// Weight quantile noise with modeled noise.
noise[i] *= (self->blockInd);
diff --git a/webrtc/modules/audio_processing/ns/nsx_core_mips.c b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
index be65c25..0e4b28f 100644
--- a/webrtc/modules/audio_processing/ns/nsx_core_mips.c
+++ b/webrtc/modules/audio_processing/ns/nsx_core_mips.c
@@ -758,7 +758,7 @@
int16_t *imag = inst->imag;
int32_t loop_count = 2;
int16_t tmp_1, tmp_2, tmp_3, tmp_4, tmp_5, tmp_6;
- int16_t tmp16 = (inst->anaLen << 1) - 4;
+ int16_t tmp16 = (int16_t)(inst->anaLen << 1) - 4;
int16_t* freq_buf_f = freq_buf;
int16_t* freq_buf_s = &freq_buf[tmp16];
diff --git a/webrtc/modules/utility/source/coder.cc b/webrtc/modules/utility/source/coder.cc
index dc0799a..1baeaef 100644
--- a/webrtc/modules/utility/source/coder.cc
+++ b/webrtc/modules/utility/source/coder.cc
@@ -85,7 +85,7 @@
AudioFrame audioFrame;
audioFrame.CopyFrom(audio);
audioFrame.timestamp_ = _encodeTimestamp;
- _encodeTimestamp += audioFrame.samples_per_channel_;
+ _encodeTimestamp += static_cast<uint32_t>(audioFrame.samples_per_channel_);
// For any codec with a frame size that is longer than 10 ms the encoded
// length in bytes should be zero until a a full frame has been encoded.
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 063ffa6..6dd64c7 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -1097,7 +1097,7 @@
Channel::UpdateLocalTimeStamp()
{
- _timeStamp += _audioFrame.samples_per_channel_;
+ _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
return 0;
}
@@ -3454,7 +3454,7 @@
return 0xFFFFFFFF;
}
- _timeStamp += _audioFrame.samples_per_channel_;
+ _timeStamp += static_cast<uint32_t>(_audioFrame.samples_per_channel_);
return 0;
}
diff --git a/webrtc/voice_engine/utility_unittest.cc b/webrtc/voice_engine/utility_unittest.cc
index 8f7efa8..a5dd70b 100644
--- a/webrtc/voice_engine/utility_unittest.cc
+++ b/webrtc/voice_engine/utility_unittest.cc
@@ -54,7 +54,7 @@
frame->sample_rate_hz_ = sample_rate_hz;
frame->samples_per_channel_ = sample_rate_hz / 100;
for (int i = 0; i < frame->samples_per_channel_; i++) {
- frame->data_[i] = data * i;
+ frame->data_[i] = static_cast<int16_t>(data * i);
}
}
@@ -72,8 +72,8 @@
frame->sample_rate_hz_ = sample_rate_hz;
frame->samples_per_channel_ = sample_rate_hz / 100;
for (int i = 0; i < frame->samples_per_channel_; i++) {
- frame->data_[i * 2] = left * i;
- frame->data_[i * 2 + 1] = right * i;
+ frame->data_[i * 2] = static_cast<int16_t>(left * i);
+ frame->data_[i * 2 + 1] = static_cast<int16_t>(right * i);
}
}