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  /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/
entropy_coding.c 169 const WebRtc_Word32 gainQ10,
183 CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
185 /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
186 if(gainQ10>400000){
187 tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
191 tmpGain = gainQ10;
239 const WebRtc_Word32 gainQ10,
256 CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
258 /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
259 if(gainQ10>400000)
    [all...]
  /external/webrtc/src/modules/audio_coding/codecs/isac/main/source/
entropy_coding.c 96 const WebRtc_Word32 gainQ10,
111 CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
113 /* To avoid overflow, we shift down gainQ10 if it is large.
115 if (gainQ10 > 400000) {
116 tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
120 tmpGain = gainQ10;
281 WebRtc_Word16 gainQ10;
352 gainQ10 = WebRtcSpl_DivW32W16ResW16(p1, (WebRtc_Word16)(
354 *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0;
355 *fi++ = (double)((data[k + 1] * gainQ10 + 512) >> 10) / 128.0
    [all...]

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