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  /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/
pitch_filter_c.c 31 WebRtc_Word16 tmpW16 = 0;
41 /* Saturate to avoid overflow in tmpW16. */
44 tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
50 gain, tmpW16, 12);
60 /* Saturate to avoid overflow in tmpW16. */
63 tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 15);
66 tmpW32 = inputBuf[*index2] - WEBRTC_SPL_MUL_16_16(sign, tmpW16);
pitch_filter.c 185 WebRtc_Word16 cnt = 0, frcQQ, indW16 = 0, tmpW16;
219 tmpW16 = WEBRTC_SPL_LSHIFT_W16(indW16, 7);
220 tmpW16 -= curLagQ7;
221 frcQQ = WEBRTC_SPL_RSHIFT_W16(tmpW16, 4);
244 tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 14);
245 tmpW32 = WEBRTC_SPL_MUL_16_16(tmpW16, tmpW16);
pitch_filter_armv6.S 93 @ Saturate to avoid overflow in tmpW16.
  /external/webrtc/src/modules/audio_processing/agc/
analog_agc.c 592 WebRtc_Word16 i, tmpW16;
597 tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(env[i], 20);
598 if (tmpW16 > 875)
600 stt->envSum += tmpW16;
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