mirror of
https://github.com/mpv-player/mpv
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78128bddda
I hate tabs. This replaces all tabs in all source files with spaces. The only exception is old-makefile. The replacement was made by running the GNU coreutils "expand" command on every file. Since the replacement was automatic, it's possible that some formatting was destroyed (but perhaps only if it was assuming that the end of a tab does not correspond to aligning the end to multiples of 8 spaces).
124 lines
3.2 KiB
C
124 lines
3.2 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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