| /external/webrtc/src/common_audio/vad/main/source/ |
| vad_gmm.c | 26 WebRtc_Word16 tmp16, tmpDiv, tmpDiv2, expVal, tmp16_1, tmp16_2; local
34 tmp16 = WEBRTC_SPL_RSHIFT_W16(tmpDiv, 2); // From Q10 to Q8
35 tmpDiv2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2); // (Q8 * Q8)>>2 = Q14
37 tmp16 = WEBRTC_SPL_LSHIFT_W16(in_sample, 3); // Q7
38 tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
42 *delta = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmpDiv2, tmp16, 10); //(Q14*Q7)>>10 = Q11
45 tmp32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9); // One shift for /2
50 // Calculate tmp16 = log2(exp(1))*tmp32 , in Q1
[all...] |
| vad_core.c | 300 WebRtc_Word16 tmp16, tmp16_1, tmp16_2; local
500 tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
501 if (nmk3 < tmp16)
502 nmk3 = tmp16;
503 tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
504 if (nmk3 > tmp16)
505 nmk3 = tmp16;
517 tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
520 smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
532 tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3)
[all...] |
| vad_filterbank.c | 72 WebRtc_Word16 tmp16; local
81 tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
82 *out_vector++ = tmp16;
84 state32 = in32 - WEBRTC_SPL_MUL_16_16(filter_coefficients, tmp16);
|
| /frameworks/base/media/libstagefright/codecs/aacenc/basic_op/ |
| oper_32b.c | 216 Word16 tmp16; local
219 tmp16 = round16(tmp);
220 tmp = L_mult(tmp16, tmp16);
221 tmp16 = round16(tmp);
222 tmp = L_mult(tmp16, tmp16);
223 tmp16 = round16(tmp);
225 iLog4 = (-(iLog4 << 2) - norm_s(tmp16)) - 1;
|
| /external/libpng/contrib/pngminus/ |
| pnm2png.c | 209 png_uint_32 tmp16; local
354 tmp16 = get_value (pnm_file, bit_depth);
355 *pix_ptr = (png_byte) ((tmp16 >> 8) & 0xFF);
357 *pix_ptr = (png_byte) (tmp16 & 0xFF);
371 tmp16 = get_value (alpha_file, bit_depth);
372 *pix_ptr++ = (png_byte) ((tmp16 >> 8) & 0xFF);
373 *pix_ptr++ = (png_byte) (tmp16 & 0xFF);
|
| /external/webrtc/src/modules/audio_processing/agc/main/source/ |
| digital_agc.c | 77 WebRtc_Word16 i, tmp16, tmp16no1; local
137 tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(kCompRatio - 1, i - 1); // Q0
138 tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14
226 tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
228 tmp32no2 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16);
233 tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
234 tmp32no2 = WEBRTC_SPL_MUL_32_16(fracPart, tmp16);
650 WebRtc_Word16 k, subfr, tmp16; local
760 tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12);
761 tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm))
[all...] |
| analog_agc.c | 119 WebRtc_Word16 i, n, L, M, subFrames, tmp16, tmp_speech[16]; local
204 tmp16 = (WebRtc_Word16)(stt->micVol - stt->maxAnalog);
205 tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
206 tmp16 = (WebRtc_Word16)(stt->maxLevel - stt->maxAnalog);
207 targetGainIdx = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32, tmp16);
537 WebRtc_Word16 tmp16; local
550 tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
551 tmp16 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32)tmp16, ANALOG_TARGET_LEVEL);
552 stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
[all...] |
| /external/webrtc/src/modules/audio_processing/aecm/main/source/ |
| aecm_core.c | 273 WebRtc_Word16 tmp16; local
360 tmp16 = PART_LEN1 - i;
361 aecm->noiseEst[i] = (tmp16 * tmp16) << 4;
365 tmp16 = PART_LEN1 - i;
366 aecm->noiseEst[i] = ((tmp16 * tmp16) << 4) << 1;
377 tmp16 = PART_LEN1 - i;
378 aecm->noiseEst[i] = (tmp16 * tmp16) << 4
765 WebRtc_Word16 tmp16; local
965 WebRtc_Word16 tmp16; local
2366 WebRtc_Word16 tmp16; local
[all...] |
| /external/webrtc/src/modules/audio_processing/ns/main/source/ |
| nsx_core.c | 677 WebRtc_Word16 log2, tabind, logval, tmp16, tmp16no1, tmp16no2; local
1200 WebRtc_Word16 tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart; local
[all...] |
| /external/chromium/chrome/browser/extensions/ |
| extension_menu_manager_unittest.cc | 480 string16 tmp16; local
481 ASSERT_TRUE(info->GetString("selectionText", &tmp16));
482 ASSERT_EQ(params.selection_text, tmp16);
|
