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-rw-r--r--audio/filter/af_lavcresample.c213
1 files changed, 213 insertions, 0 deletions
diff --git a/audio/filter/af_lavcresample.c b/audio/filter/af_lavcresample.c
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--- /dev/null
+++ b/audio/filter/af_lavcresample.c
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+/*
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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 <string.h>
+#include <inttypes.h>
+
+#include "config.h"
+#include "af.h"
+#include "libavcodec/avcodec.h"
+#include "libavutil/rational.h"
+
+// Data for specific instances of this filter
+typedef struct af_resample_s{
+ struct AVResampleContext *avrctx;
+ int16_t *in[AF_NCH];
+ int in_alloc;
+ int index;
+
+ int filter_length;
+ int linear;
+ int phase_shift;
+ double cutoff;
+
+ int ctx_out_rate;
+ int ctx_in_rate;
+ int ctx_filter_size;
+ int ctx_phase_shift;
+ int ctx_linear;
+ double ctx_cutoff;
+}af_resample_t;
+
+
+// Initialization and runtime control
+static int control(struct af_instance* af, int cmd, void* arg)
+{
+ af_resample_t* s = (af_resample_t*)af->setup;
+ struct mp_audio *data= (struct mp_audio*)arg;
+ int out_rate, test_output_res; // helpers for checking input format
+
+ switch(cmd){
+ case AF_CONTROL_REINIT:
+ if((af->data->rate == data->rate) || (af->data->rate == 0))
+ return AF_DETACH;
+
+ af->data->nch = data->nch;
+ if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
+ af->data->format = AF_FORMAT_S16_NE;
+ af->data->bps = 2;
+ af->mul = (double)af->data->rate / data->rate;
+ af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
+
+ if (s->ctx_out_rate != af->data->rate || s->ctx_in_rate != data->rate || s->ctx_filter_size != s->filter_length ||
+ s->ctx_phase_shift != s->phase_shift || s->ctx_linear != s->linear || s->ctx_cutoff != s->cutoff) {
+ if(s->avrctx) av_resample_close(s->avrctx);
+ s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);
+ s->ctx_out_rate = af->data->rate;
+ s->ctx_in_rate = data->rate;
+ s->ctx_filter_size = s->filter_length;
+ s->ctx_phase_shift = s->phase_shift;
+ s->ctx_linear = s->linear;
+ s->ctx_cutoff = s->cutoff;
+ }
+
+ // hack to make af_test_output ignore the samplerate change
+ out_rate = af->data->rate;
+ af->data->rate = data->rate;
+ test_output_res = af_test_output(af, (struct mp_audio*)arg);
+ af->data->rate = out_rate;
+ return test_output_res;
+ case AF_CONTROL_COMMAND_LINE:{
+ s->cutoff= 0.0;
+ sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
+ if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
+ return AF_OK;
+ }
+ case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
+ af->data->rate = *(int*)arg;
+ return AF_OK;
+ }
+ return AF_UNKNOWN;
+}
+
+// Deallocate memory
+static void uninit(struct af_instance* af)
+{
+ if(af->data)
+ free(af->data->audio);
+ free(af->data);
+ if(af->setup){
+ int i;
+ af_resample_t *s = af->setup;
+ if(s->avrctx) av_resample_close(s->avrctx);
+ for (i=0; i < AF_NCH; i++)
+ free(s->in[i]);
+ free(s);
+ }
+}
+
+// Filter data through filter
+static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
+{
+ af_resample_t *s = af->setup;
+ int i, j, consumed, ret = 0;
+ int16_t *in = (int16_t*)data->audio;
+ int16_t *out;
+ int chans = data->nch;
+ int in_len = data->len/(2*chans);
+ int out_len = in_len * af->mul + 10;
+ int16_t tmp[AF_NCH][out_len];
+
+ if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
+ return NULL;
+
+ out= (int16_t*)af->data->audio;
+
+ out_len= min(out_len, af->data->len/(2*chans));
+
+ if(s->in_alloc < in_len + s->index){
+ s->in_alloc= in_len + s->index;
+ for(i=0; i<chans; i++){
+ s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
+ }
+ }
+
+ if(chans==1){
+ memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
+ }else if(chans==2){
+ for(j=0; j<in_len; j++){
+ s->in[0][j + s->index]= *(in++);
+ s->in[1][j + s->index]= *(in++);
+ }
+ }else{
+ for(j=0; j<in_len; j++){
+ for(i=0; i<chans; i++){
+ s->in[i][j + s->index]= *(in++);
+ }
+ }
+ }
+ in_len += s->index;
+
+ for(i=0; i<chans; i++){
+ ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
+ }
+ out_len= ret;
+
+ s->index= in_len - consumed;
+ for(i=0; i<chans; i++){
+ memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
+ }
+
+ if(chans==1){
+ memcpy(out, tmp[0], out_len*sizeof(int16_t));
+ }else if(chans==2){
+ for(j=0; j<out_len; j++){
+ *(out++)= tmp[0][j];
+ *(out++)= tmp[1][j];
+ }
+ }else{
+ for(j=0; j<out_len; j++){
+ for(i=0; i<chans; i++){
+ *(out++)= tmp[i][j];
+ }
+ }
+ }
+
+ data->audio = af->data->audio;
+ data->len = out_len*chans*2;
+ data->rate = af->data->rate;
+ return data;
+}
+
+static int af_open(struct af_instance* af){
+ af_resample_t *s = calloc(1,sizeof(af_resample_t));
+ af->control=control;
+ af->uninit=uninit;
+ af->play=play;
+ af->mul=1;
+ af->data=calloc(1,sizeof(struct mp_audio));
+ s->filter_length= 16;
+ s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
+ s->phase_shift= 10;
+// s->setup = RSMP_INT | FREQ_SLOPPY;
+ af->setup=s;
+ return AF_OK;
+}
+
+struct af_info af_info_lavcresample = {
+ "Sample frequency conversion using libavcodec",
+ "lavcresample",
+ "Michael Niedermayer",
+ "",
+ AF_FLAGS_REENTRANT,
+ af_open
+};
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