mirror of
https://github.com/mpv-player/mpv
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124 lines
3.1 KiB
C
124 lines
3.1 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <math.h>
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#include <string.h>
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#include "common/common.h"
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#include "af.h"
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/* Convert to gain value from dB. Returns AF_OK if of and AF_ERROR if
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* fail. input <= -200dB will become 0 gain. */
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int af_from_dB(int n, float* in, float* out, float k, float mi, float ma)
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{
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int i = 0;
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// Sanity check
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if(!in || !out)
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return AF_ERROR;
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for(i=0;i<n;i++){
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if(in[i]<=-200)
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out[i]=0.0;
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else
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out[i]=pow(10.0,MPCLAMP(in[i],mi,ma)/k);
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}
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return AF_OK;
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}
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/* Convert from gain value to dB. Returns AF_OK if of and AF_ERROR if
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* fail. gain=0 will become -200 dB. k is just a multiplier. */
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int af_to_dB(int n, float* in, float* out, float k)
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{
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int i = 0;
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// Sanity check
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if(!in || !out)
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return AF_ERROR;
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for(i=0;i<n;i++){
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if(in[i] == 0.0)
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out[i]=-200.0;
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else
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out[i]=k*log10(in[i]);
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}
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return AF_OK;
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}
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/* Convert from ms to sample time */
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int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma)
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{
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int i = 0;
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// Sanity check
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if(!in || !out)
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return AF_ERROR;
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for(i=0;i<n;i++)
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out[i]=(int)((float)rate * MPCLAMP(in[i],mi,ma)/1000.0);
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return AF_OK;
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}
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/* Convert from sample time to ms */
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int af_to_ms(int n, int* in, float* out, int rate)
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{
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int i = 0;
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// Sanity check
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if(!in || !out || !rate)
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return AF_ERROR;
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for(i=0;i<n;i++)
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out[i]=1000.0 * (float)in[i]/((float)rate);
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return AF_OK;
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}
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/*
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* test if output format matches
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* af: audio filter
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* out: needed format, will be overwritten by available
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* format if they do not match
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* returns: AF_FALSE if formats do not match, AF_OK if they match
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*
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* compares the format, rate and nch values of af->data with out
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* Note: logically, *out=*af->data always happens, because out contains the
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* format only, no actual audio data or memory allocations. *out always
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* contains the parameters from af->data after the function returns.
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*/
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int af_test_output(struct af_instance* af, struct mp_audio* out)
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{
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if((af->data->format != out->format) ||
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(af->data->bps != out->bps) ||
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(af->data->rate != out->rate) ||
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!mp_chmap_equals(&af->data->channels, &out->channels)){
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*out = *af->data;
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return AF_FALSE;
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}
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return AF_OK;
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}
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/* Soft clipping, the sound of a dream, thanks to Jon Wattes
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post to Musicdsp.org */
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float af_softclip(float a)
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{
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if (a >= M_PI/2)
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return 1.0;
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else if (a <= -M_PI/2)
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return -1.0;
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else
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return sin(a);
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}
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