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Prepare to convert various types to size_t.

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This makes some behaviorally-invariant changes to make certain code that
currently only works correctly with signed types work safely regardless of the
signedness of the types in question.  This is preparation for a future change
that will convert a variety of types to size_t.

There are also some formatting changes (e.g. converting "enum hack" usage to real consts) to make it simpler to just change "int" to "size_t" in the future to change the types of those constants.

BUG=none
R=andrew@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=ajm

Review URL: https://codereview.webrtc.org/1174813003

Cr-Commit-Position: refs/heads/master@{#9413}
This commit is contained in:
Peter Kasting
2015-06-10 21:15:38 -07:00
parent 786dbdcc38
commit f045e4da43
42 changed files with 153 additions and 127 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
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@ -37,8 +37,8 @@ int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
int16_t* current_framesamples);
int16_t WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
IsacFixDecoderInstance* ISACdec_obj,
int16_t* current_framesample );
IsacFixDecoderInstance* ISACdec_obj,
int16_t* current_framesample );
int WebRtcIsacfix_EncodeImpl(int16_t* in,
IsacFixEncoderInstance* ISACenc_obj,

2
webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
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@ -309,7 +309,7 @@ int16_t WebRtcIsacfix_DecodePlcImpl(int16_t *signal_out16,
&((ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0]);
minCorr = WEBRTC_SPL_WORD32_MAX;
if ( (FRAMESAMPLES_HALF - 2*lag0 - 10) > 0 )
if ((FRAMESAMPLES_HALF - 10) > 2 * lag0)
{
minIdx = 11;
for( i = 0; i < 21; i++ )

10
webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
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@ -279,13 +279,15 @@ void WebRtcIsacfix_NormLatticeFilterAr(int16_t orderCoef,
ARfQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
}
for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
for (i=orderCoef;i>0;i--) //get the state of f&g for the first input, for all orders
{
tmp32 = (cthQ15[i] * ARfQ0vec[0] - sthQ15[i] * stateGQ0[i] + 16384) >> 15;
tmp32 = (cthQ15[i - 1] * ARfQ0vec[0] - sthQ15[i - 1] * stateGQ0[i - 1] +
16384) >> 15;
tmpAR = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
tmp32 = (sthQ15[i] * ARfQ0vec[0] + cthQ15[i] * stateGQ0[i] + 16384) >> 15;
ARgQ0vec[i+1] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
tmp32 = (sthQ15[i - 1] * ARfQ0vec[0] + cthQ15[i - 1] * stateGQ0[i - 1] +
16384) >> 15;
ARgQ0vec[i] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32); // Q0
ARfQ0vec[0] = tmpAR;
}
ARgQ0vec[0] = ARfQ0vec[0];

20
webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
View File

@ -46,21 +46,21 @@ HALF_SUBFRAME_LOOP: @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
add r2, r9, asl #1 @ Restore r2 to &cth_Q15[order_coef]
add r3, r9, asl #1 @ Restore r3 to &sth_Q15[order_coef]
ORDER_COEF_LOOP: @ for(k = order_coef - 1 ; k >= 0; k--)
ORDER_COEF_LOOP: @ for(k = order_coef ; k > 0; k--)
ldrh r7, [r3, #-2]! @ sth_Q15[k]
ldrh r6, [r2, #-2]! @ cth_Q15[k]
ldrh r7, [r3, #-2]! @ sth_Q15[k - 1]
ldrh r6, [r2, #-2]! @ cth_Q15[k - 1]
ldrh r8, [r0, #-2] @ ar_g_Q0[k]
smlabb r11, r7, r5, r12 @ sth_Q15[k] * tmpAR + 16384
smlabb r10, r6, r5, r12 @ cth_Q15[k] * tmpAR + 16384
smulbb r7, r7, r8 @ sth_Q15[k] * ar_g_Q0[k]
smlabb r11, r6, r8, r11 @ cth_Q15[k]*ar_g_Q0[k]+(sth_Q15[k]*tmpAR+16384)
ldrh r8, [r0, #-2] @ ar_g_Q0[k - 1]
smlabb r11, r7, r5, r12 @ sth_Q15[k - 1] * tmpAR + 16384
smlabb r10, r6, r5, r12 @ cth_Q15[k - 1] * tmpAR + 16384
smulbb r7, r7, r8 @ sth_Q15[k - 1] * ar_g_Q0[k - 1]
smlabb r11, r6, r8, r11 @ cth_Q15[k - 1]*ar_g_Q0[k - 1]+(sth_Q15[k - 1]*tmpAR+16384)
sub r10, r10, r7 @ cth_Q15[k]*tmpAR+16384-(sth_Q15[k]*ar_g_Q0[k])
sub r10, r10, r7 @ cth_Q15[k - 1]*tmpAR+16384-(sth_Q15[k - 1]*ar_g_Q0[k - 1])
ssat r11, #16, r11, asr #15
ssat r5, #16, r10, asr #15
strh r11, [r0], #-2 @ Output: ar_g_Q0[k+1]
strh r11, [r0], #-2 @ Output: ar_g_Q0[k]
subs r9, #1
bgt ORDER_COEF_LOOP

10
webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
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@ -35,11 +35,13 @@ void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples
int32_t tmp32_2 = 0;
tmpAR = ar_f_Q0[n + 1];
for (k = order_coef - 1; k >= 0; k--) {
tmp32 = (cth_Q15[k] * tmpAR - sth_Q15[k] * ar_g_Q0[k] + 16384) >> 15;
tmp32_2 = (sth_Q15[k] * tmpAR + cth_Q15[k] * ar_g_Q0[k] + 16384) >> 15;
for (k = order_coef; k > 0; k--) {
tmp32 = (cth_Q15[k - 1] * tmpAR - sth_Q15[k - 1] * ar_g_Q0[k - 1] +
16384) >> 15;
tmp32_2 = (sth_Q15[k - 1] * tmpAR + cth_Q15[k - 1] * ar_g_Q0[k - 1] +
16384) >> 15;
tmpAR = (int16_t)WebRtcSpl_SatW32ToW16(tmp32);
ar_g_Q0[k + 1] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32_2);
ar_g_Q0[k] = (int16_t)WebRtcSpl_SatW32ToW16(tmp32_2);
}
ar_f_Q0[n + 1] = tmpAR;
ar_g_Q0[0] = tmpAR;

7
webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter.c
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@ -69,7 +69,6 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
int16_t oldLagQ7;
int16_t oldGainQ12, lagdeltaQ7, curLagQ7, gaindeltaQ12, curGainQ12;
int indW32 = 0, frcQQ = 0;
int32_t tmpW32;
const int16_t* fracoeffQQ = NULL;
// Assumptions in ARM assembly for WebRtcIsacfix_PitchFilterCoreARM().
@ -123,8 +122,7 @@ void WebRtcIsacfix_PitchFilter(int16_t* indatQQ, // Q10 if type is 1 or 4,
curGainQ12 += gaindeltaQ12;
curLagQ7 += lagdeltaQ7;
indW32 = CalcLrIntQ(curLagQ7, 7);
tmpW32 = (indW32 << 7) - curLagQ7;
frcQQ = (tmpW32 >> 4) + 4;
frcQQ = ((indW32 << 7) + 64 - curLagQ7) >> 4;
if (frcQQ == PITCH_FRACS) {
frcQQ = 0;
@ -195,8 +193,7 @@ void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
// Update parameters for each segment.
curLagQ7 += lagdeltaQ7;
indW16 = (int16_t)CalcLrIntQ(curLagQ7, 7);
tmpW16 = (indW16 << 7) - curLagQ7;
frcQQ = (tmpW16 >> 4) + 4;
frcQQ = ((indW16 << 7) + 64 - curLagQ7) >> 4;
if (frcQQ == PITCH_FRACS) {
frcQQ = 0;

9
webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
View File

@ -65,18 +65,21 @@ float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
// ISAC takes 10 ms everycall
const int subblocks = block_duration_ms_ / 10;
const int subblock_length = 10 * input_sampling_khz_;
int value;
int value = 0;
clock_t clocks = clock();
size_t pointer = 0;
for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
bit_stream);
if (idx == subblocks - 1)
EXPECT_GT(value, 0);
else
EXPECT_EQ(0, value);
}
clocks = clock() - clocks;
EXPECT_GT(value, 0);
assert(value <= max_bytes);
*encoded_bytes = value;
assert(*encoded_bytes <= max_bytes);
return 1000.0 * clocks / CLOCKS_PER_SEC;
}

2
webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
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@ -232,7 +232,7 @@ int main(int argc, char* argv[])
CodingMode = 0;
testNum = 0;
testCE = 0;
for (i = 1; i < argc-2;i++) {
for (i = 1; i + 2 < argc; i++) {
/* Instantaneous mode */
if (!strcmp ("-I", argv[i])) {
printf("\nInstantaneous BottleNeck\n");

2
webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
View File

@ -185,7 +185,7 @@ int main(int argc, char* argv[]) {
char transCodingFileName[500];
int16_t totFileLoop = 0;
int16_t numFileLoop = 0;
for (i = 1; i < argc - 2; i++) {
for (i = 1; i + 2 < argc; i++) {
if (!strcmp("-LOOP", argv[i])) {
i++;
totFileLoop = (int16_t)atol(argv[i]);