| |
| |
| |
| |
| |
| |
| |
| |
| |
|
|
|
|
| |
| |
| |
| |
| |
|
|
| #include "common_audio/signal_processing/include/signal_processing_library.h" |
|
|
| int16_t WebRtcSpl_GetScalingSquare(int16_t* in_vector, |
| size_t in_vector_length, |
| size_t times) |
| { |
| int16_t nbits = WebRtcSpl_GetSizeInBits((uint32_t)times); |
| size_t i; |
| int16_t smax = -1; |
| int16_t sabs; |
| int16_t *sptr = in_vector; |
| int16_t t; |
| size_t looptimes = in_vector_length; |
|
|
| for (i = looptimes; i > 0; i--) |
| { |
| sabs = (*sptr > 0 ? *sptr++ : -*sptr++); |
| smax = (sabs > smax ? sabs : smax); |
| } |
| t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); |
|
|
| if (smax == 0) |
| { |
| return 0; |
| } else |
| { |
| return (t > nbits) ? 0 : nbits - t; |
| } |
| } |
|
|