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-rw-r--r--audio/filter/af_resample.c394
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diff --git a/audio/filter/af_resample.c b/audio/filter/af_resample.c
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+/*
+ * This audio filter changes the sample rate.
+ *
+ * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
+ *
+ * This file is part of MPlayer.
+ *
+ * MPlayer is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * MPlayer is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with MPlayer; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <inttypes.h>
+
+#include "libavutil/common.h"
+#include "libavutil/mathematics.h"
+#include "af.h"
+#include "dsp.h"
+
+/* Below definition selects the length of each poly phase component.
+ Valid definitions are L8 and L16, where the number denotes the
+ length of the filter. This definition affects the computational
+ complexity (see play()), the performance (see filter.h) and the
+ memory usage. The filter length is chosen to 8 if the machine is
+ slow and to 16 if the machine is fast and has MMX.
+*/
+
+#if !HAVE_MMX // This machine is slow
+#define L8
+#else
+#define L16
+#endif
+
+#include "af_resample_template.c"
+
+// Filtering types
+#define RSMP_LIN (0<<0) // Linear interpolation
+#define RSMP_INT (1<<0) // 16 bit integer
+#define RSMP_FLOAT (2<<0) // 32 bit floating point
+#define RSMP_MASK (3<<0)
+
+// Defines for sloppy or exact resampling
+#define FREQ_SLOPPY (0<<2)
+#define FREQ_EXACT (1<<2)
+#define FREQ_MASK (1<<2)
+
+// Accuracy for linear interpolation
+#define STEPACCURACY 32
+
+// local data
+typedef struct af_resample_s
+{
+ void* w; // Current filter weights
+ void** xq; // Circular buffers
+ uint32_t xi; // Index for circular buffers
+ uint32_t wi; // Index for w
+ uint32_t i; // Number of new samples to put in x queue
+ uint32_t dn; // Down sampling factor
+ uint32_t up; // Up sampling factor
+ uint64_t step; // Step size for linear interpolation
+ uint64_t pt; // Pointer remainder for linear interpolation
+ int setup; // Setup parameters cmdline or through postcreate
+} af_resample_t;
+
+// Fast linear interpolation resample with modest audio quality
+static int linint(struct mp_audio* c,struct mp_audio* l, af_resample_t* s)
+{
+ uint32_t len = 0; // Number of input samples
+ uint32_t nch = l->nch; // Words pre transfer
+ uint64_t step = s->step;
+ int16_t* in16 = ((int16_t*)c->audio);
+ int16_t* out16 = ((int16_t*)l->audio);
+ int32_t* in32 = ((int32_t*)c->audio);
+ int32_t* out32 = ((int32_t*)l->audio);
+ uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
+ uint64_t pt = s->pt;
+ uint16_t tmp;
+
+ switch (nch){
+ case 1:
+ while(pt < end){
+ out16[len++]=in16[pt>>STEPACCURACY];
+ pt+=step;
+ }
+ s->pt=pt & ((1LL<<STEPACCURACY)-1);
+ break;
+ case 2:
+ end/=2;
+ while(pt < end){
+ out32[len++]=in32[pt>>STEPACCURACY];
+ pt+=step;
+ }
+ len=(len<<1);
+ s->pt=pt & ((1LL<<STEPACCURACY)-1);
+ break;
+ default:
+ end /=nch;
+ while(pt < end){
+ tmp=nch;
+ do {
+ tmp--;
+ out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
+ } while (tmp);
+ len+=nch;
+ pt+=step;
+ }
+ s->pt=pt & ((1LL<<STEPACCURACY)-1);
+ }
+ return len;
+}
+
+/* Determine resampling type and format */
+static int set_types(struct af_instance* af, struct mp_audio* data)
+{
+ af_resample_t* s = af->setup;
+ int rv = AF_OK;
+ float rd = 0;
+
+ // Make sure this filter isn't redundant
+ if((af->data->rate == data->rate) || (af->data->rate == 0))
+ return AF_DETACH;
+ /* If sloppy and small resampling difference (2%) */
+ rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate;
+ if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) &&
+ (data->format != (AF_FORMAT_FLOAT_NE))) ||
+ ((s->setup & RSMP_MASK) == RSMP_LIN)){
+ s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
+ af->data->format = AF_FORMAT_S16_NE;
+ af->data->bps = 2;
+ mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Using linear interpolation. \n");
+ }
+ else{
+ /* If the input format is float or if float is explicitly selected
+ use float, otherwise use int */
+ if((data->format == (AF_FORMAT_FLOAT_NE)) ||
+ ((s->setup & RSMP_MASK) == RSMP_FLOAT)){
+ s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
+ af->data->format = AF_FORMAT_FLOAT_NE;
+ af->data->bps = 4;
+ }
+ else{
+ s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
+ af->data->format = AF_FORMAT_S16_NE;
+ af->data->bps = 2;
+ }
+ mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Using %s processing and %s frequecy"
+ " conversion.\n",
+ ((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer",
+ ((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact");
+ }
+
+ if(af->data->format != data->format || af->data->bps != data->bps)
+ rv = AF_FALSE;
+ data->format = af->data->format;
+ data->bps = af->data->bps;
+ af->data->nch = data->nch;
+ return rv;
+}
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+ switch(cmd){
+ case AF_CONTROL_REINIT:{
+ af_resample_t* s = af->setup;
+ struct mp_audio* n = arg; // New configuration
+ int i,d = 0;
+ int rv = AF_OK;
+
+ // Free space for circular buffers
+ if(s->xq){
+ free(s->xq[0]);
+ free(s->xq);
+ s->xq = NULL;
+ }
+
+ if(AF_DETACH == (rv = set_types(af,n)))
+ return AF_DETACH;
+
+ // If linear interpolation
+ if((s->setup & RSMP_MASK) == RSMP_LIN){
+ s->pt=0LL;
+ s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
+ mp_msg(MSGT_AFILTER, MSGL_DBG2, "[resample] Linear interpolation step: 0x%016"PRIX64".\n",
+ s->step);
+ af->mul = (double)af->data->rate / n->rate;
+ return rv;
+ }
+
+ // Calculate up and down sampling factors
+ d=av_gcd(af->data->rate,n->rate);
+
+ // If sloppy resampling is enabled limit the upsampling factor
+ if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){
+ int up=af->data->rate/2;
+ int dn=n->rate/2;
+ int m=2;
+ while(af->data->rate/(d*m) > 5000){
+ d=av_gcd(up,dn);
+ up/=2; dn/=2; m*=2;
+ }
+ d*=m;
+ }
+
+ // Create space for circular buffers
+ s->xq = malloc(n->nch*sizeof(void*));
+ s->xq[0] = calloc(n->nch, 2*L*af->data->bps);
+ for(i=1;i<n->nch;i++)
+ s->xq[i] = (uint8_t *)s->xq[i-1] + 2*L*af->data->bps;
+ s->xi = 0;
+
+ // Check if the design needs to be redone
+ if(s->up != af->data->rate/d || s->dn != n->rate/d){
+ float* w;
+ float* wt;
+ float fc;
+ int j;
+ s->up = af->data->rate/d;
+ s->dn = n->rate/d;
+ s->wi = 0;
+ s->i = 0;
+
+ // Calculate cutoff frequency for filter
+ fc = 1/(float)(max(s->up,s->dn));
+ // Allocate space for polyphase filter bank and prototype filter
+ w = malloc(sizeof(float) * s->up *L);
+ free(s->w);
+ s->w = malloc(L*s->up*af->data->bps);
+
+ // Design prototype filter type using Kaiser window with beta = 10
+ if(NULL == w || NULL == s->w ||
+ -1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] Unable to design prototype filter.\n");
+ return AF_ERROR;
+ }
+ // Copy data from prototype to polyphase filter
+ wt=w;
+ for(j=0;j<L;j++){//Columns
+ for(i=0;i<s->up;i++){//Rows
+ if((s->setup & RSMP_MASK) == RSMP_INT){
+ float t=(float)s->up*32767.0*(*wt);
+ ((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
+ }
+ else
+ ((float*)s->w)[i*L+j] = (float)s->up*(*wt);
+ wt++;
+ }
+ }
+ free(w);
+ mp_msg(MSGT_AFILTER, MSGL_V, "[resample] New filter designed up: %i "
+ "down: %i\n", s->up, s->dn);
+ }
+
+ // Set multiplier and delay
+ af->delay = 0; // not set correctly, but shouldn't be too large anyway
+ af->mul = (double)s->up / s->dn;
+ return rv;
+ }
+ case AF_CONTROL_COMMAND_LINE:{
+ af_resample_t* s = af->setup;
+ int rate=0;
+ int type=RSMP_INT;
+ int sloppy=1;
+ sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type);
+ s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) |
+ (clamp(type,RSMP_LIN,RSMP_FLOAT));
+ return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
+ }
+ case AF_CONTROL_POST_CREATE:
+ if((((struct af_cfg*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
+ ((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
+ return AF_OK;
+ case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
+ // Reinit must be called after this function has been called
+
+ // Sanity check
+ if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] The output sample frequency "
+ "must be between 8kHz and 192kHz. Current value is %i \n",
+ ((int*)arg)[0]);
+ return AF_ERROR;
+ }
+
+ af->data->rate=((int*)arg)[0];
+ mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Changing sample rate "
+ "to %iHz\n",af->data->rate);
+ return AF_OK;
+ }
+ return AF_UNKNOWN;
+}
+
+// Deallocate memory
+static void uninit(struct af_instance* af)
+{
+ af_resample_t *s = af->setup;
+ if (s) {
+ if (s->xq) free(s->xq[0]);
+ free(s->xq);
+ free(s->w);
+ free(s);
+ }
+ if(af->data)
+ free(af->data->audio);
+ free(af->data);
+}
+
+// Filter data through filter
+static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
+{
+ int len = 0; // Length of output data
+ struct mp_audio* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
+ af_resample_t* s = af->setup;
+
+ if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
+ return NULL;
+
+ // Run resampling
+ switch(s->setup & RSMP_MASK){
+ case(RSMP_INT):
+# define FORMAT_I 1
+ if(s->up>s->dn){
+# define UP
+# include "af_resample_template.c"
+# undef UP
+ }
+ else{
+# define DN
+# include "af_resample_template.c"
+# undef DN
+ }
+ break;
+ case(RSMP_FLOAT):
+# undef FORMAT_I
+# define FORMAT_F 1
+ if(s->up>s->dn){
+# define UP
+# include "af_resample_template.c"
+# undef UP
+ }
+ else{
+# define DN
+# include "af_resample_template.c"
+# undef DN
+ }
+ break;
+ case(RSMP_LIN):
+ len = linint(c, l, s);
+ break;
+ }
+
+ // Set output data
+ c->audio = l->audio;
+ c->len = len*l->bps;
+ c->rate = l->rate;
+
+ return c;
+}
+
+// Allocate memory and set function pointers
+static int af_open(struct af_instance* af){
+ af->control=control;
+ af->uninit=uninit;
+ af->play=play;
+ af->mul=1;
+ af->data=calloc(1,sizeof(struct mp_audio));
+ af->setup=calloc(1,sizeof(af_resample_t));
+ if(af->data == NULL || af->setup == NULL)
+ return AF_ERROR;
+ ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
+ return AF_OK;
+}
+
+// Description of this plugin
+struct af_info af_info_resample = {
+ "Sample frequency conversion",
+ "resample",
+ "Anders",
+ "",
+ AF_FLAGS_REENTRANT,
+ af_open
+};