| /external/webrtc/webrtc/modules/audio_processing/aecm/ |
| aecm_core_c.c | 175 int16_t tmp16no1; local
187 tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
188 time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
222 tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
225 if(tmp16no1 > tmp16no2)
227 max_value = tmp16no1;
232 min_value = tmp16no1;
249 tmp16no1 = (int16_t)((max_value * alpha) >> 15);
251 freq_signal_abs[i] = (uint16_t)tmp16no1 + (uint16_t)tmp16no2;
263 tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real)
317 int16_t tmp16no1; local
[all...] |
| aecm_core.c | 909 int16_t tmp16no1; local
1111 int16_t tmp16no1; local
[all...] |
| aecm_core_mips.c | 638 int16_t tmp16no1; local
829 int16_t tmp16no1; local
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/ns/ |
| nsx_core_c.c | 35 int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart; local
102 tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
104 tmp16no2 += (int16_t)((tmp16no1 * frac) >> 14);
134 tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
136 tmp16no2 += (int16_t)((tmp16no1 * frac) >> 14);
181 tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
184 tmp16no1, frac, 14);
|
| nsx_core_mips.c | 34 int16_t tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac; local
137 tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
139 tmp16no2 += (int16_t)((tmp16no1 * frac) >> 14);
169 tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
171 tmp16no2 += (int16_t)((tmp16no1 * frac) >> 14);
216 tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
219 tmp16no1, frac, 14);
[all...] |
| nsx_core.c | 340 int16_t log2, tabind, logval, tmp16, tmp16no1, tmp16no2; local
426 tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
430 inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
1100 int16_t tmp16no1; local
1430 int16_t tmp16no1, tmp16no2; local
[all...] |
| nsx_core_neon.c | 140 int16_t log2, tabind, logval, tmp16, tmp16no1, tmp16no2; local
325 tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
329 inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
|
| /external/webrtc/webrtc/modules/audio_processing/agc/legacy/ |
| digital_agc.c | 81 int16_t i, tmp16, tmp16no1; local
91 tmp16no1 = analogTarget - targetLevelDbfs;
92 tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
93 maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
121 tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
122 limiterLvl = targetLevelDbfs + tmp16no1;
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