mirror of https://github.com/mpv-player/mpv
Extending delay to have different delays for different channels
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8676 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
anders
parent
ec6de0f24a
commit
2ee32fa05a
199
libaf/af_delay.c
199
libaf/af_delay.c
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@ -1,7 +1,6 @@
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/* This audio filter doesn't really do anything useful but serves an
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example of how audio filters work. It delays the output signal by
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the number of seconds set by delay=n where n is the number of
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seconds.
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/* This audio filter delays the output signal for the different
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channels and can be used for simple position panning. Extension for
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this filter would be a reverb.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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@ -9,87 +8,96 @@
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#include "af.h"
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#define L 65536
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#define UPDATEQI(qi) qi=(qi+1)&(L-1)
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// Data for specific instances of this filter
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typedef struct af_delay_s
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{
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void* buf; // data block used for delaying audio signal
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int len; // local buffer length
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float tlen; // Delay in seconds
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void* q[AF_NCH]; // Circular queues used for delaying audio signal
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int wi[AF_NCH]; // Write index
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int ri; // Read index
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float d[AF_NCH]; // Delay [ms]
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}af_delay_t;
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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_delay_t* s = af->setup;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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int i;
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// Free prevous delay queues
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for(i=0;i<af->data->nch;i++){
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if(s->q[i])
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free(s->q[i]);
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}
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = ((af_data_t*)arg)->nch;
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af->data->format = ((af_data_t*)arg)->format;
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af->data->bps = ((af_data_t*)arg)->bps;
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return af->control(af,AF_CONTROL_DELAY_LEN | AF_CONTROL_SET,
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&((af_delay_t*)af->setup)->tlen);
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// Allocate new delay queues
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for(i=0;i<af->data->nch;i++){
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s->q[i] = calloc(L,af->data->bps);
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if(NULL == s->q[i])
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af_msg(AF_MSG_FATAL,"[delay] Out of memory\n");
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}
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return control(af,AF_CONTROL_DELAY_LEN | AF_CONTROL_SET,s->d);
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}
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case AF_CONTROL_COMMAND_LINE:{
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float d = 0;
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sscanf((char*)arg,"%f",&d);
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return af->control(af,AF_CONTROL_DELAY_LEN | AF_CONTROL_SET,&d);
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}
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case AF_CONTROL_DELAY_LEN | AF_CONTROL_SET:{
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af_delay_t* s = (af_delay_t*)af->setup;
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void* bt = s->buf; // Old buffer
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int lt = s->len; // Old len
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if(*((float*)arg) > 30 || *((float*)arg) < 0){
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af_msg(AF_MSG_ERROR,"Error setting delay length in af_delay. Delay must be between 0s and 30s\n");
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s->len=0;
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s->tlen=0.0;
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af->delay=0.0;
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return AF_ERROR;
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}
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// Set new len and allocate new buffer
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s->tlen = *((float*)arg);
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af->delay = s->tlen * 1000.0;
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s->len = af->data->rate*af->data->bps*af->data->nch*(int)s->tlen;
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s->buf = malloc(s->len);
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af_msg(AF_MSG_DEBUG0,"[delay] Delaying audio output by %0.2fs\n",s->tlen);
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af_msg(AF_MSG_DEBUG1,"[delay] Delaying audio output by %i bytes\n",s->len);
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// Out of memory error
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if(!s->buf){
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s->len = 0;
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free(bt);
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return AF_ERROR;
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}
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// Clear the new buffer
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memset(s->buf, 0, s->len);
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/* Copy old buffer to avoid click in output
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sound (at least most of it) and release it */
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if(bt){
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memcpy(s->buf,bt,min(lt,s->len));
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free(bt);
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int n = 1;
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int i = 0;
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char* cl = arg;
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while(n && i < AF_NCH ){
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sscanf(cl,"%f:%n",&s->d[i],&n);
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if(n==0 || cl[n-1] == '\0')
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break;
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cl=&cl[n];
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i++;
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}
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return AF_OK;
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}
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case AF_CONTROL_DELAY_LEN | AF_CONTROL_GET:
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*((float*)arg) = ((af_delay_t*)af->setup)->tlen;
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case AF_CONTROL_DELAY_LEN | AF_CONTROL_SET:{
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int i;
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if(AF_OK != af_from_ms(AF_NCH, arg, s->wi, af->data->rate, 0.0, 1000.0))
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return AF_ERROR;
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s->ri = 0;
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for(i=0;i<AF_NCH;i++){
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af_msg(AF_MSG_DEBUG0,"[delay] Channel %i delayed by %0.3fms\n",
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i,clamp(s->d[i],0.0,1000.0));
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af_msg(AF_MSG_DEBUG1,"[delay] Channel %i delayed by %i samples\n",
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i,s->wi[i]);
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}
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return AF_OK;
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}
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case AF_CONTROL_DELAY_LEN | AF_CONTROL_GET:{
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int i;
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for(i=0;i<AF_NCH;i++){
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if(s->ri > s->wi[i])
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s->wi[i] = L - (s->ri - s->wi[i]);
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else
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s->wi[i] = s->wi[i] - s->ri;
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}
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return af_to_ms(AF_NCH, s->wi, arg, af->data->rate);
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}
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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->audio)
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free(af->data->audio);
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int i;
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if(af->data)
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free(af->data);
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if(((af_delay_t*)(af->setup))->buf)
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free(((af_delay_t*)(af->setup))->buf);
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for(i=0;i<AF_NCH;i++)
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if(((af_delay_t*)(af->setup))->q[i])
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free(((af_delay_t*)(af->setup))->q[i]);
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if(af->setup)
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free(af->setup);
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}
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@ -97,34 +105,59 @@ static void uninit(struct af_instance_s* af)
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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af_data_t* c = data; // Current working data
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af_data_t* l = af->data; // Local data
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af_delay_t* s = (af_delay_t*)af->setup; // Setup for this instance
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if(AF_OK != RESIZE_LOCAL_BUFFER(af , data))
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return NULL;
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if(s->len > c->len){ // Delay bigger than buffer
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// Copy beginning of buffer to beginning of output buffer
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memcpy(l->audio,s->buf,c->len);
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// Move buffer left
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memmove(s->buf,s->buf+c->len,s->len-c->len);
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// Save away current audio to end of buffer
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memcpy(s->buf+s->len-c->len,c->audio,c->len);
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af_data_t* c = data; // Current working data
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af_delay_t* s = af->setup; // Setup for this instance
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int nch = c->nch; // Number of channels
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int len = c->len/c->bps; // Number of sample in data chunk
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int ri = 0;
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int ch,i;
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for(ch=0;ch<nch;ch++){
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switch(c->bps){
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case 1:{
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int8_t* a = c->audio;
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int8_t* q = s->q[ch];
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int wi = s->wi[ch];
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ri = s->ri;
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for(i=ch;i<len;i+=nch){
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q[wi] = a[i];
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a[i] = q[ri];
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UPDATEQI(wi);
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UPDATEQI(ri);
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}
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s->wi[ch] = wi;
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break;
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}
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case 2:{
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int16_t* a = c->audio;
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int16_t* q = s->q[ch];
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int wi = s->wi[ch];
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ri = s->ri;
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for(i=ch;i<len;i+=nch){
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q[wi] = a[i];
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a[i] = q[ri];
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UPDATEQI(wi);
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UPDATEQI(ri);
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}
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s->wi[ch] = wi;
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break;
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}
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case 4:{
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int32_t* a = c->audio;
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int32_t* q = s->q[ch];
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int wi = s->wi[ch];
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ri = s->ri;
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for(i=ch;i<len;i+=nch){
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q[wi] = a[i];
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a[i] = q[ri];
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UPDATEQI(wi);
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UPDATEQI(ri);
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}
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s->wi[ch] = wi;
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break;
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}
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}
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}
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else{
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// Copy end of previous block to beginning of output buffer
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memcpy(l->audio,s->buf,s->len);
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// Copy current block except end
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memcpy(l->audio+s->len,c->audio,c->len-s->len);
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// Save away end of current block for next call
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memcpy(s->buf,c->audio+c->len-s->len,s->len);
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}
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// Set output data
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c->audio=l->audio;
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s->ri = ri;
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return c;
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}
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