webrtc / src / 3cc47ebd2d0efc6a48ddb5b142fa9ea9c1ae4435 / . / webrtc / common_audio / signal_processing / filter_ar_fast_q12.c

/* | |

* 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/base/checks.h" | |

#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" | |

// TODO(bjornv): Change the return type to report errors. | |

void WebRtcSpl_FilterARFastQ12(const int16_t* data_in, | |

int16_t* data_out, | |

const int16_t* __restrict coefficients, | |

size_t coefficients_length, | |

size_t data_length) { | |

size_t i = 0; | |

size_t j = 0; | |

RTC_DCHECK_GT(data_length, 0); | |

RTC_DCHECK_GT(coefficients_length, 1); | |

for (i = 0; i < data_length; i++) { | |

int32_t output = 0; | |

int32_t sum = 0; | |

for (j = coefficients_length - 1; j > 0; j--) { | |

sum += coefficients[j] * data_out[i - j]; | |

} | |

output = coefficients[0] * data_in[i]; | |

output -= sum; | |

// Saturate and store the output. | |

output = WEBRTC_SPL_SAT(134215679, output, -134217728); | |

data_out[i] = (int16_t)((output + 2048) >> 12); | |

} | |

} |