blob: 726a88819ac01e47e3dbfb4e41ccd2739d2f3dcd [file] [log] [blame]
/*
* 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"
// TODO(Bjornv): Change the function parameter order to WebRTC code style.
// C version of WebRtcSpl_DownsampleFast() for generic platforms.
int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
size_t data_in_length,
int16_t* data_out,
size_t data_out_length,
const int16_t* __restrict coefficients,
size_t coefficients_length,
int factor,
size_t delay) {
size_t i = 0;
size_t j = 0;
int32_t out_s32 = 0;
size_t endpos = delay + factor * (data_out_length - 1) + 1;
// Return error if any of the running conditions doesn't meet.
if (data_out_length == 0 || coefficients_length == 0
|| data_in_length < endpos) {
return -1;
}
for (i = delay; i < endpos; i += factor) {
out_s32 = 2048; // Round value, 0.5 in Q12.
for (j = 0; j < coefficients_length; j++) {
out_s32 += coefficients[j] * data_in[i - j]; // Q12.
}
out_s32 >>= 12; // Q0.
// Saturate and store the output.
*data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
}
return 0;
}