| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" |
| #include "webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h" |
| |
| /* Filter coefficicients in Q15. */ |
| static const int16_t kDampFilter[PITCH_DAMPORDER] = { |
| -2294, 8192, 20972, 8192, -2294 |
| }; |
| |
| void WebRtcIsacfix_PitchFilterCore(int loopNumber, |
| int16_t gain, |
| size_t index, |
| int16_t sign, |
| int16_t* inputState, |
| int16_t* outputBuf2, |
| const int16_t* coefficient, |
| int16_t* inputBuf, |
| int16_t* outputBuf, |
| int* index2) { |
| int i = 0, j = 0; /* Loop counters. */ |
| int16_t* ubufQQpos2 = &outputBuf2[PITCH_BUFFSIZE - (index + 2)]; |
| int16_t tmpW16 = 0; |
| |
| for (i = 0; i < loopNumber; i++) { |
| int32_t tmpW32 = 0; |
| |
| /* Filter to get fractional pitch. */ |
| for (j = 0; j < PITCH_FRACORDER; j++) { |
| tmpW32 += ubufQQpos2[*index2 + j] * coefficient[j]; |
| } |
| |
| /* Saturate to avoid overflow in tmpW16. */ |
| tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104); |
| tmpW32 += 8192; |
| tmpW16 = (int16_t)(tmpW32 >> 14); |
| |
| /* Shift low pass filter state. */ |
| memmove(&inputState[1], &inputState[0], |
| (PITCH_DAMPORDER - 1) * sizeof(int16_t)); |
| inputState[0] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND( |
| gain, tmpW16, 12); |
| |
| /* Low pass filter. */ |
| tmpW32 = 0; |
| /* TODO(kma): Define a static inline function WebRtcSpl_DotProduct() |
| in spl_inl.h to replace this and other similar loops. */ |
| for (j = 0; j < PITCH_DAMPORDER; j++) { |
| tmpW32 += inputState[j] * kDampFilter[j]; |
| } |
| |
| /* Saturate to avoid overflow in tmpW16. */ |
| tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208); |
| tmpW32 += 16384; |
| tmpW16 = (int16_t)(tmpW32 >> 15); |
| |
| /* Subtract from input and update buffer. */ |
| tmpW32 = inputBuf[*index2] - sign * tmpW16; |
| outputBuf[*index2] = WebRtcSpl_SatW32ToW16(tmpW32); |
| tmpW32 = inputBuf[*index2] + outputBuf[*index2]; |
| outputBuf2[*index2 + PITCH_BUFFSIZE] = WebRtcSpl_SatW32ToW16(tmpW32); |
| |
| (*index2)++; |
| } |
| } |