mirror of
https://github.com/mpv-player/mpv
synced 2024-12-27 01:22:30 +00:00
7deec05ea0
Change the audio filters to use a double instead of rationals for the ratio of output to input size. The rationals could overflow when calculating the overall ratio of a filter chain and gave no real advantage compared to doubles. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24916 b3059339-0415-0410-9bf9-f77b7e298cf2
150 lines
3.4 KiB
C
150 lines
3.4 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU General Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2004 Alex Beregszaszi & Pierre Lombard
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//
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//=============================================================================
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <inttypes.h>
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#include <math.h>
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#include <limits.h>
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#include "af.h"
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// Data for specific instances of this filter
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typedef struct af_extrastereo_s
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{
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float mul;
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}af_extrastereo_t;
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static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data);
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static af_data_t* play_float(struct af_instance_s* af, af_data_t* data);
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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af_extrastereo_t* s = (af_extrastereo_t*)af->setup;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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// Sanity check
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if(!arg) return AF_ERROR;
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = 2;
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if (((af_data_t*)arg)->format == AF_FORMAT_FLOAT_NE)
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{
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af->data->format = AF_FORMAT_FLOAT_NE;
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af->data->bps = 4;
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af->play = play_float;
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}// else
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{
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af->data->format = AF_FORMAT_S16_NE;
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af->data->bps = 2;
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af->play = play_s16;
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}
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return af_test_output(af,(af_data_t*)arg);
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}
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case AF_CONTROL_COMMAND_LINE:{
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float f;
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sscanf((char*)arg,"%f", &f);
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s->mul = f;
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return AF_OK;
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}
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case AF_CONTROL_ES_MUL | AF_CONTROL_SET:
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s->mul = *(float*)arg;
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return AF_OK;
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case AF_CONTROL_ES_MUL | AF_CONTROL_GET:
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*(float*)arg = s->mul;
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data);
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if(af->setup)
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free(af->setup);
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}
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// Filter data through filter
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static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data)
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{
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af_extrastereo_t *s = af->setup;
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register int i = 0;
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int16_t *a = (int16_t*)data->audio; // Audio data
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int len = data->len/2; // Number of samples
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int avg, l, r;
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for (i = 0; i < len; i+=2)
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{
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avg = (a[i] + a[i + 1]) / 2;
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l = avg + (int)(s->mul * (a[i] - avg));
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r = avg + (int)(s->mul * (a[i + 1] - avg));
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a[i] = clamp(l, SHRT_MIN, SHRT_MAX);
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a[i + 1] = clamp(r, SHRT_MIN, SHRT_MAX);
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}
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return data;
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}
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static af_data_t* play_float(struct af_instance_s* af, af_data_t* data)
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{
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af_extrastereo_t *s = af->setup;
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register int i = 0;
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float *a = (float*)data->audio; // Audio data
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int len = data->len/4; // Number of samples
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float avg, l, r;
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for (i = 0; i < len; i+=2)
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{
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avg = (a[i] + a[i + 1]) / 2;
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l = avg + (s->mul * (a[i] - avg));
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r = avg + (s->mul * (a[i + 1] - avg));
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a[i] = af_softclip(l);
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a[i + 1] = af_softclip(r);
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}
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return data;
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}
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// Allocate memory and set function pointers
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static int af_open(af_instance_t* af){
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af->control=control;
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af->uninit=uninit;
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af->play=play_s16;
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af->mul=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=calloc(1,sizeof(af_extrastereo_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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((af_extrastereo_t*)af->setup)->mul = 2.5;
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return AF_OK;
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}
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// Description of this filter
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af_info_t af_info_extrastereo = {
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"Extra stereo",
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"extrastereo",
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"Alex Beregszaszi & Pierre Lombard",
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"",
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AF_FLAGS_NOT_REENTRANT,
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af_open
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};
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