small warning fix:
[mplayer/glamo.git] / libaf / af_lavcresample.c
blobdb970a16150d23333a7bf0d6227c410372b458b3
1 // Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
2 // #inlcude <GPL_v2.h>
4 #include <stdio.h>
5 #include <stdlib.h>
6 #include <string.h>
7 #include <inttypes.h>
9 #include "../config.h"
10 #include "af.h"
12 #ifdef USE_LIBAVCODEC
14 #ifdef USE_LIBAVCODEC_SO
15 #include <ffmpeg/avcodec.h>
16 #include <ffmpeg/rational.h>
17 #else
18 #include "../libavcodec/avcodec.h"
19 #include "../libavcodec/rational.h"
20 #endif
22 #define CHANS 6
24 int64_t ff_gcd(int64_t a, int64_t b);
26 // Data for specific instances of this filter
27 typedef struct af_resample_s{
28 struct AVResampleContext *avrctx;
29 int16_t *in[CHANS];
30 int in_alloc;
31 int index;
33 int filter_length;
34 int linear;
35 int phase_shift;
36 double cutoff;
37 }af_resample_t;
40 // Initialization and runtime control
41 static int control(struct af_instance_s* af, int cmd, void* arg)
43 af_resample_t* s = (af_resample_t*)af->setup;
44 af_data_t *data= (af_data_t*)arg;
45 int out_rate, test_output_res; // helpers for checking input format
47 switch(cmd){
48 case AF_CONTROL_REINIT:
49 if((af->data->rate == data->rate) || (af->data->rate == 0))
50 return AF_DETACH;
52 af->data->nch = data->nch;
53 if (af->data->nch > CHANS) af->data->nch = CHANS;
54 af->data->format = AF_FORMAT_S16_NE;
55 af->data->bps = 2;
56 af->mul.n = af->data->rate;
57 af->mul.d = data->rate;
58 af_frac_cancel(&af->mul);
59 af->delay = 500*s->filter_length/(double)min(af->data->rate, data->rate);
61 if(s->avrctx) av_resample_close(s->avrctx);
62 s->avrctx= av_resample_init(af->mul.n, /*in_rate*/af->mul.d, s->filter_length, s->phase_shift, s->linear, s->cutoff);
64 // hack to make af_test_output ignore the samplerate change
65 out_rate = af->data->rate;
66 af->data->rate = data->rate;
67 test_output_res = af_test_output(af, (af_data_t*)arg);
68 af->data->rate = out_rate;
69 return test_output_res;
70 case AF_CONTROL_COMMAND_LINE:{
71 sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
72 if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 1.0/s->filter_length, 0.80);
73 return AF_OK;
75 case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
76 af->data->rate = *(int*)arg;
77 return AF_OK;
79 return AF_UNKNOWN;
82 // Deallocate memory
83 static void uninit(struct af_instance_s* af)
85 if(af->data)
86 free(af->data);
87 if(af->setup){
88 af_resample_t *s = af->setup;
89 if(s->avrctx) av_resample_close(s->avrctx);
90 free(s);
94 // Filter data through filter
95 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
97 af_resample_t *s = af->setup;
98 int i, j, consumed, ret;
99 int16_t *in = (int16_t*)data->audio;
100 int16_t *out;
101 int chans = data->nch;
102 int in_len = data->len/(2*chans);
103 int out_len = (in_len*af->mul.n) / af->mul.d + 10;
104 int16_t tmp[CHANS][out_len];
106 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
107 return NULL;
109 out= (int16_t*)af->data->audio;
111 out_len= min(out_len, af->data->len/(2*chans));
113 if(s->in_alloc < in_len + s->index){
114 s->in_alloc= in_len + s->index;
115 for(i=0; i<chans; i++){
116 s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t)); //FIXME free this maybe ;)
120 if(chans==1){
121 memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
122 }else if(chans==2){
123 for(j=0; j<in_len; j++){
124 s->in[0][j + s->index]= *(in++);
125 s->in[1][j + s->index]= *(in++);
127 }else{
128 for(j=0; j<in_len; j++){
129 for(i=0; i<chans; i++){
130 s->in[i][j + s->index]= *(in++);
134 in_len += s->index;
136 for(i=0; i<chans; i++){
137 ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
139 out_len= ret;
141 s->index= in_len - consumed;
142 for(i=0; i<chans; i++){
143 memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
146 if(chans==1){
147 memcpy(out, tmp[0], out_len*sizeof(int16_t));
148 }else if(chans==2){
149 for(j=0; j<out_len; j++){
150 *(out++)= tmp[0][j];
151 *(out++)= tmp[1][j];
153 }else{
154 for(j=0; j<out_len; j++){
155 for(i=0; i<chans; i++){
156 *(out++)= tmp[i][j];
161 data->audio = af->data->audio;
162 data->len = out_len*chans*2;
163 data->rate = af->data->rate;
164 return data;
167 static int open(af_instance_t* af){
168 af_resample_t *s = calloc(1,sizeof(af_resample_t));
169 af->control=control;
170 af->uninit=uninit;
171 af->play=play;
172 af->mul.n=1;
173 af->mul.d=1;
174 af->data=calloc(1,sizeof(af_data_t));
175 s->filter_length= 16;
176 s->cutoff= max(1.0 - 1.0/s->filter_length, 0.80);
177 s->phase_shift= 10;
178 // s->setup = RSMP_INT | FREQ_SLOPPY;
179 af->setup=s;
180 return AF_OK;
183 af_info_t af_info_lavcresample = {
184 "Sample frequency conversion using libavcodec",
185 "lavcresample",
186 "Michael Niedermayer",
188 AF_FLAGS_REENTRANT,
189 open
191 #endif