Add audio filter scaletempo

Patch by Robert Juliano, juliano.1 osu edu


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24924 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
uau 2007-11-01 06:52:38 +00:00
parent aa657df525
commit d33703496c
8 changed files with 642 additions and 9 deletions

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@ -5148,6 +5148,79 @@ Beware that this filter will turn your signal into mono.
Works well for 2 channel tracks; do not bother trying it
on anything but 2 channel stereo.
.
.TP
.B scaletempo[=option1:option2:...]
Scales audio tempo without altering pitch, optionally synced to playback
speed (default).
.br
This works by playing \'stride\' ms of audio at normal speed then
consuming \'stride*scale\' ms of input audio.
It pieces the strides together by blending 'overlap'% of stride with
audio following the previous stride.
It optionally performs a short statistical analysis on the next \'search\'
ms of audio to determine the best overlap position.
.PD 0
.RSs
.IPs scale=<amount>
Nominal amount to scale tempo.
Scales this amount in addition to speed.
(default: 1.0)
.IPs stride=<amount>
Length in milliseconds to output each stride.
Too high of value will cause noticable skips at high scale amounts and
an echo at low scale amounts.
Very low values will alter pitch.
Increasing improves performance.
(default: 60)
.IPs overlap=<percent>
Percentage of stride to overlap.
Decreasing improves performance.
(default: .20)
.IPs search=<amount>
Length in milliseconds to search for best overlap position.
Decreasing improves performance greatly.
On slow systems, you will probably want to set this very low.
(default: 14)
.IPs speed=<tempo|pitch|both|none>
Set response to speed change.
.RSss
.IPs tempo
Scale tempo in sync with speed (default)
.IPs pitch
Reverses effect of filter.
Scales pitch without altering tempo.
Add \'[ speed_mult 0.9438743126816935\' and \'] speed_mult 1.059463094352953\'
to your input.conf to step by musical semi-tones.
.I WARNING:
Looses synch with video.
.IPs both
Scale both tempo and pitch
.IPs none
Ignore speed changes
.RE
.RE
.sp 1
.RS
.I EXAMPLE:
.RE
.RSs
.IPs "mplayer \-af scaletempo \-speed 1.2 media.ogg"
Would playback media at 1.2x normal speed, with audio at normal pitch.
Changing playback speed, would change audio tempo to match.
.IPs "mplayer \-af scaletempo=scale=1.2:speed=none \-speed 1.2 media.ogg"
Would playback media at 1.2x normal speed, with audio at normal pitch,
but changing playback speed has no effect on audio tempo.
.IPs "mplayer \-af scaletempo=stride=30:overlap=.50:search=10 media.ogg"
Would tweak the quality and performace parameters.
.IPs "mplayer \-af format=floatne,scaletempo media.ogg"
Would make scaletempo use float code.
Maybe faster on some platforms.
.IPs "mplayer \-af scaletempo=scale=1.2:speed=pitch audio.ogg"
Would playback audio file at 1.2x normal speed, with audio at normal pitch.
Changing playback speed, would change pitch, leaving audio tempo at 1.2x.
.RE
.PD 1
.
.
.
.SH "VIDEO FILTERS"

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@ -30,6 +30,7 @@ SRCS_COMMON = asxparser.c \
playtreeparser.c \
spudec.c \
sub_cc.c \
subopt-helper.c \
subreader.c \
vobsub.c \
@ -41,7 +42,6 @@ SRCS_MPLAYER = mplayer.c \
mp_msg.c \
mixer.c \
parser-mpcmd.c \
subopt-helper.c \
command.c \
SRCS_MENCODER = mencoder.c \

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@ -1254,6 +1254,7 @@ static char help_text[]=
// ======================= AF Audio Filters ================================
// libaf
#define MSGTR_AF_ValueOutOfRange MSGTR_VO_ValueOutOfRange
// af_ladspa.c

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@ -16,6 +16,7 @@ SRCS_COMMON = af.c \
af_karaoke.c \
af_pan.c \
af_resample.c \
af_scaletempo.c \
af_sinesuppress.c \
af_sub.c \
af_surround.c \

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@ -31,6 +31,7 @@ extern af_info_t af_info_ladspa;
extern af_info_t af_info_center;
extern af_info_t af_info_sinesuppress;
extern af_info_t af_info_karaoke;
extern af_info_t af_info_scaletempo;
static af_info_t* filter_list[]={
&af_info_dummy,
@ -61,6 +62,7 @@ static af_info_t* filter_list[]={
&af_info_center,
&af_info_sinesuppress,
&af_info_karaoke,
&af_info_scaletempo,
NULL
};

547
libaf/af_scaletempo.c Normal file
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@ -0,0 +1,547 @@
/*
* scaletempo audio filter
* Copyright (c) 2007 Robert Juliano
*
* 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
*
* scale tempo while maintaining pitch
* (WSOLA technique with cross correlation)
* inspired by SoundTouch library by Olli Parviainen
*
* basic algorithm
* - produce 'stride' output samples per loop
* - consume stride*scale input samples per loop
*
* to produce smoother transitions between strides, blend next overlap
* samples from last stride with correlated samples of current input
*
*/
#include <stdlib.h>
#include <string.h>
#include <limits.h>
#include "af.h"
#include "libavutil/common.h"
#include "subopt-helper.h"
#include "help_mp.h"
// Data for specific instances of this filter
typedef struct af_scaletempo_s
{
// stride
float scale;
float speed;
float frames_stride_scaled;
float frames_stride_error;
int bytes_per_frame;
int bytes_stride;
float bytes_stride_scaled;
int bytes_queue;
int bytes_queued;
int bytes_to_slide;
int8_t* buf_queue;
// overlap
int samples_overlap;
int samples_standing;
int bytes_overlap;
int bytes_standing;
int8_t* buf_overlap;
int8_t* table_blend;
void (*output_overlap)(struct af_scaletempo_s* s, int8_t* out_buf, int bytes_off);
// best overlap
int frames_search;
int num_channels;
int8_t* buf_pre_corr;
int8_t* table_window;
int (*best_overlap_offset)(struct af_scaletempo_s* s);
short shift_corr;
// command line
float scale_nominal;
float ms_stride;
float percent_overlap;
float ms_search;
short speed_tempo;
short speed_pitch;
} af_scaletempo_t;
int fill_queue(struct af_instance_s* af, af_data_t* data, int offset) {
af_scaletempo_t* s = af->setup;
int bytes_in = data->len - offset;
int offset_unchanged = offset;
if (s->bytes_to_slide > 0) {
if (s->bytes_to_slide < s->bytes_queued) {
int bytes_move = s->bytes_queued - s->bytes_to_slide;
memmove(s->buf_queue,
s->buf_queue + s->bytes_to_slide,
bytes_move);
s->bytes_to_slide = 0;
s->bytes_queued = bytes_move;
} else {
int bytes_skip;
s->bytes_to_slide -= s->bytes_queued;
bytes_skip = FFMIN(s->bytes_to_slide, bytes_in);
s->bytes_queued = 0;
s->bytes_to_slide -= bytes_skip;
offset += bytes_skip;
bytes_in -= bytes_skip;
}
}
if (bytes_in > 0) {
int bytes_copy = FFMIN(s->bytes_queue - s->bytes_queued, bytes_in);
memcpy(s->buf_queue + s->bytes_queued,
(int8_t*)data->audio + offset,
bytes_copy);
s->bytes_queued += bytes_copy;
offset += bytes_copy;
}
return offset - offset_unchanged;
}
int _best_overlap_offset_float(af_scaletempo_t* s) {
float *pw, *po, *ppc, *search_start;
float best_corr = INT_MIN;
int best_off = 0;
int i, off;
pw = (float*)s->table_window;
po = (float*)s->buf_overlap + s->num_channels;
ppc = (float*)s->buf_pre_corr;
for (i=s->num_channels; i<s->samples_overlap; i++) {
*ppc++ = *pw++ * *po++;
}
search_start = (float*)s->buf_queue + s->num_channels;
for (off=0; off<s->frames_search; off++) {
float corr = 0;
float* ps = search_start;
ppc = (float*)s->buf_pre_corr;
for (i=s->num_channels; i<s->samples_overlap; i++) {
corr += *ppc++ * *ps++;
}
if (corr > best_corr) {
best_corr = corr;
best_off = off;
}
search_start += s->num_channels;
}
return best_off * 4;
}
int _best_overlap_offset_s16(af_scaletempo_t* s) {
int32_t *pw, *ppc;
int16_t *po, *search_start;
int32_t best_corr = INT_MIN;
int best_off = 0;
int i, off;
pw = (int32_t*)s->table_window;
po = (int16_t*)s->buf_overlap + s->num_channels;
ppc = (int32_t*)s->buf_pre_corr;
for (i=s->num_channels; i<s->samples_overlap; i++) {
*ppc++ = ( *pw++ * *po++ ) >> 15;
}
search_start = (int16_t*)s->buf_queue + s->num_channels;
for (off=0; off<s->frames_search; off++) {
int32_t corr = 0;
int16_t* ps = search_start;
ppc = (int32_t*)s->buf_pre_corr;
for (i=s->num_channels; i<s->samples_overlap; i++) {
corr += ( *ppc++ * *ps++ ) >> s->shift_corr;
}
if (corr > best_corr) {
best_corr = corr;
best_off = off;
}
search_start += s->num_channels;
}
return best_off * 2;
}
void _output_overlap_float(af_scaletempo_t* s, int8_t* buf_out, int bytes_off) {
float* pout = (float*)buf_out;
float* pb = (float*)s->table_blend;
float* po = (float*)s->buf_overlap;
float* pin = (float*)(s->buf_queue + bytes_off);
int i;
for (i=0; i<s->samples_overlap; i++) {
*pout++ = *po - *pb++ * ( *po - *pin++ ); po++;
}
}
void _output_overlap_s16(af_scaletempo_t* s, int8_t* buf_out, int bytes_off) {
int16_t* pout = (int16_t*)buf_out;
int32_t* pb = (int32_t*)s->table_blend;
int16_t* po = (int16_t*)s->buf_overlap;
int16_t* pin = (int16_t*)(s->buf_queue + bytes_off);
int i;
for (i=0; i<s->samples_overlap; i++) {
*pout++ = *po - ( ( *pb++ * ( *po - *pin++ ) ) >> 16 ); po++;
}
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_scaletempo_t* s = af->setup;
int offset_in;
int max_bytes_out;
int8_t* pout;
if (s->scale == 1.0) {
return data;
}
// RESIZE_LOCAL_BUFFER - can't use macro
max_bytes_out = ((int)(data->len / s->bytes_stride_scaled) + 1) * s->bytes_stride;
if (max_bytes_out > af->data->len) {
af_msg(AF_MSG_VERBOSE, "[libaf] Reallocating memory in module %s, "
"old len = %i, new len = %i\n",af->info->name,af->data->len,max_bytes_out);
af->data->audio = realloc(af->data->audio, max_bytes_out);
if (!af->data->audio) {
af_msg(AF_MSG_FATAL, "[libaf] Could not allocate memory\n");
return NULL;
}
af->data->len = max_bytes_out;
}
offset_in = fill_queue(af, data, 0);
pout = af->data->audio;
while (s->bytes_queued >= s->bytes_queue) {
int ti;
float tf;
int bytes_off = 0;
// output stride
if (s->output_overlap) {
if (s->best_overlap_offset)
bytes_off = s->best_overlap_offset(s);
s->output_overlap(s, pout, bytes_off);
}
memcpy(pout + s->bytes_overlap,
s->buf_queue + bytes_off + s->bytes_overlap,
s->bytes_standing);
pout += s->bytes_stride;
// input stride
memcpy(s->buf_overlap,
s->buf_queue + bytes_off + s->bytes_stride,
s->bytes_overlap);
tf = s->frames_stride_scaled + s->frames_stride_error;
ti = (int)tf;
s->frames_stride_error = tf - ti;
s->bytes_to_slide = ti * s->bytes_per_frame;
offset_in += fill_queue(af, data, offset_in);
}
data->audio = af->data->audio;
data->len = (int)pout - (int)af->data->audio;
return data;
}
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_scaletempo_t* s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
af_data_t* data = (af_data_t*)arg;
float srate = data->rate / 1000;
int nch = data->nch;
int bps;
int use_int = 0;
int frames_stride, frames_overlap;
int i, j;
af_msg(AF_MSG_VERBOSE,
"[scaletempo] %.3f speed * %.3f scale_nominal = %.3f\n",
s->speed, s->scale_nominal, s->scale);
if (s->scale == 1.0) {
if (s->speed_tempo && s->speed_pitch)
return AF_DETACH;
memcpy(af->data, data, sizeof(af_data_t));
return af_test_output(af, data);
}
af->data->rate = data->rate;
af->data->nch = data->nch;
if ( data->format == AF_FORMAT_S16_LE
|| data->format == AF_FORMAT_S16_BE ) {
use_int = 1;
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = bps = 2;
} else {
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = bps = 4;
}
frames_stride = srate * s->ms_stride;
s->bytes_stride = frames_stride * bps * nch;
s->bytes_stride_scaled = s->scale * s->bytes_stride;
s->frames_stride_scaled = s->scale * frames_stride;
s->frames_stride_error = 0;
af->mul = (double)s->bytes_stride / s->bytes_stride_scaled;
frames_overlap = frames_stride * s->percent_overlap;
if (frames_overlap <= 0) {
s->bytes_standing = s->bytes_stride;
s->samples_standing = s->bytes_standing / bps;
s->output_overlap = NULL;
} else {
s->samples_overlap = frames_overlap * nch;
s->bytes_overlap = frames_overlap * nch * bps;
s->bytes_standing = s->bytes_stride - s->bytes_overlap;
s->samples_standing = s->bytes_standing / bps;
s->buf_overlap = realloc(s->buf_overlap, s->bytes_overlap);
s->table_blend = realloc(s->table_blend, s->bytes_overlap * 4);
if(!s->buf_overlap || !s->table_blend) {
af_msg(AF_MSG_FATAL, "[scaletempo] Out of memory\n");
return AF_ERROR;
}
bzero(s->buf_overlap, s->bytes_overlap);
if (use_int) {
int32_t* pb = (int32_t*)s->table_blend;
int64_t blend = 0;
for (i=0; i<frames_overlap; i++) {
int32_t v = blend / frames_overlap;
for (j=0; j<nch; j++) {
*pb++ = v;
}
blend += 65536; // 2^16
}
s->output_overlap = _output_overlap_s16;
} else {
float* pb = (float*)s->table_blend;
for (i=0; i<frames_overlap; i++) {
float v = i / (float)frames_overlap;
for (j=0; j<nch; j++) {
*pb++ = v;
}
}
s->output_overlap = _output_overlap_float;
}
}
s->frames_search = (frames_overlap > 1) ? srate * s->ms_search : 0;
if (s->frames_search <= 0) {
s->best_overlap_offset = NULL;
} else {
if (use_int) {
int64_t t = frames_overlap;
int32_t n = 8589934588LL / (t * t); // 4 * (2^31 - 1) / t^2
int32_t* pw;
s->buf_pre_corr = realloc(s->buf_pre_corr, s->bytes_overlap * 2);
s->table_window = realloc(s->table_window, s->bytes_overlap * 2 - nch * bps * 2);
if(!s->buf_pre_corr && !s->table_window) {
af_msg(AF_MSG_FATAL, "[scaletempo] Out of memory\n");
return AF_ERROR;
}
pw = (int32_t*)s->table_window;
for (i=1; i<frames_overlap; i++) {
int32_t v = ( i * (t - i) * n ) >> 15;
for (j=0; j<nch; j++) {
*pw++ = v;
}
}
s->shift_corr = av_log2( 2*(s->samples_overlap - nch) - 1 );
s->best_overlap_offset = _best_overlap_offset_s16;
} else {
float* pw;
s->buf_pre_corr = realloc(s->buf_pre_corr, s->bytes_overlap);
s->table_window = realloc(s->table_window, s->bytes_overlap - nch * bps);
if(!s->buf_pre_corr || !s->table_window) {
af_msg(AF_MSG_FATAL, "[scaletempo] Out of memory\n");
return AF_ERROR;
}
pw = (float*)s->table_window;
for (i=1; i<frames_overlap; i++) {
float v = i * (frames_overlap - i);
for (j=0; j<nch; j++) {
*pw++ = v;
}
}
s->best_overlap_offset = _best_overlap_offset_float;
}
}
s->bytes_per_frame = bps * nch;
s->num_channels = nch;
s->bytes_queue
= (s->frames_search + frames_stride + frames_overlap) * bps * nch;
s->buf_queue = realloc(s->buf_queue, s->bytes_queue);
if(!s->buf_queue) {
af_msg(AF_MSG_FATAL, "[scaletempo] Out of memory\n");
return AF_ERROR;
}
af_msg (AF_MSG_DEBUG0, "[scaletempo] "
"%.2f stride_in, %i stride_out, %i standing, "
"%i overlap, %i search, %i queue, %s mode\n",
s->frames_stride_scaled,
(int)(s->bytes_stride / nch / bps),
(int)(s->bytes_standing / nch / bps),
(int)(s->bytes_overlap / nch / bps),
s->frames_search,
(int)(s->bytes_queue / nch / bps),
(use_int?"s16":"float"));
return af_test_output(af, (af_data_t*)arg);
}
case AF_CONTROL_PLAYBACK_SPEED | AF_CONTROL_SET:{
if (s->speed_tempo) {
if (s->speed_pitch) {
break;
}
s->speed = *(float*)arg;
s->scale = s->speed * s->scale_nominal;
} else {
if (s->speed_pitch) {
s->speed = 1 / *(float*)arg;
s->scale = s->speed * s->scale_nominal;
break;
}
}
return AF_OK;
}
case AF_CONTROL_SCALETEMPO_AMOUNT | AF_CONTROL_SET:{
s->scale = *(float*)arg;
s->scale = s->speed * s->scale_nominal;
return AF_OK;
}
case AF_CONTROL_SCALETEMPO_AMOUNT | AF_CONTROL_GET:
*(float*)arg = s->scale;
return AF_OK;
case AF_CONTROL_COMMAND_LINE:{
strarg_t speed;
opt_t subopts[] = {
{"scale", OPT_ARG_FLOAT, &s->scale_nominal, NULL},
{"stride", OPT_ARG_FLOAT, &s->ms_stride, NULL},
{"overlap", OPT_ARG_FLOAT, &s->percent_overlap, NULL},
{"search", OPT_ARG_FLOAT, &s->ms_search, NULL},
{"speed", OPT_ARG_STR, &speed, NULL},
{NULL},
};
if (subopt_parse(arg, subopts) != 0) {
return AF_ERROR;
}
if (s->scale_nominal <= 0) {
af_msg(AF_MSG_ERROR, "[scaletempo] "
MSGTR_ErrorParsingCommandLine ": " MSGTR_AF_ValueOutOfRange
": scale > 0\n");
return AF_ERROR;
}
if (s->ms_stride <= 0) {
af_msg(AF_MSG_ERROR, "[scaletempo] "
MSGTR_ErrorParsingCommandLine ": " MSGTR_AF_ValueOutOfRange
": stride > 0\n");
return AF_ERROR;
}
if (s->percent_overlap < 0 || s->percent_overlap > 1) {
af_msg(AF_MSG_ERROR, "[scaletempo] "
MSGTR_ErrorParsingCommandLine ": " MSGTR_AF_ValueOutOfRange
": 0 <= overlap <= 1\n");
return AF_ERROR;
}
if (s->ms_search < 0) {
af_msg(AF_MSG_ERROR, "[scaletempo] "
MSGTR_ErrorParsingCommandLine ": " MSGTR_AF_ValueOutOfRange
": search >= 0\n");
return AF_ERROR;
}
if (speed.len > 0) {
if (strcmp(speed.str, "pitch") == 0) {
s->speed_tempo = 0;
s->speed_pitch = 1;
} else if (strcmp(speed.str, "tempo") == 0) {
s->speed_tempo = 1;
s->speed_pitch = 0;
} else if (strcmp(speed.str, "none") == 0) {
s->speed_tempo = 0;
s->speed_pitch = 0;
} else if (strcmp(speed.str, "both") == 0) {
s->speed_tempo = 1;
s->speed_pitch = 1;
} else {
af_msg(AF_MSG_ERROR, "[scaletempo] "
MSGTR_ErrorParsingCommandLine ": " MSGTR_AF_ValueOutOfRange
": speed=[pitch|tempo|none|both]\n");
return AF_ERROR;
}
}
s->scale = s->speed * s->scale_nominal;
af_msg(AF_MSG_DEBUG0, "[scaletempo] %6.3f scale, %6.2f stride, %6.2f overlap, %6.2f search, speed = %s\n", s->scale_nominal, s->ms_stride, s->percent_overlap, s->ms_search, (s->speed_tempo?(s->speed_pitch?"tempo and speed":"tempo"):(s->speed_pitch?"pitch":"none")));
return AF_OK;
}
}
return AF_UNKNOWN;
}
// Deallocate memory
static void uninit(struct af_instance_s* af)
{
af_scaletempo_t* s = af->setup;
free(af->data->audio);
free(af->data);
free(s->buf_queue);
free(s->buf_overlap);
free(s->buf_pre_corr);
free(s->table_blend);
free(s->table_window);
free(af->setup);
}
// Allocate memory and set function pointers
static int af_open(af_instance_t* af){
af_scaletempo_t* s;
af->control = control;
af->uninit = uninit;
af->play = play;
af->mul = 1;
af->data = calloc(1,sizeof(af_data_t));
af->setup = calloc(1,sizeof(af_scaletempo_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
s = af->setup;
s->scale = s->speed = s->scale_nominal = 1.0;
s->speed_tempo = 1;
s->speed_pitch = 0;
s->ms_stride = 60;
s->percent_overlap = .20;
s->ms_search = 14;
return AF_OK;
}
// Description of this filter
af_info_t af_info_scaletempo = {
"Scale audio tempo while maintaining pitch",
"scaletempo",
"Robert Juliano",
"",
AF_FLAGS_REENTRANT,
af_open
};

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@ -231,4 +231,7 @@ typedef struct af_control_ext_s{
#define AF_CONTROL_SS_FREQ 0x00002300 | AF_CONTROL_FILTER_SPECIFIC
#define AF_CONTROL_SS_DECAY 0x00002400 | AF_CONTROL_FILTER_SPECIFIC
#define AF_CONTROL_PLAYBACK_SPEED 0x00002500 | AF_CONTROL_FILTER_SPECIFIC
#define AF_CONTROL_SCALETEMPO_AMOUNT 0x00002600 | AF_CONTROL_FILTER_SPECIFIC
#endif /*__af_control_h */

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@ -1203,14 +1203,20 @@ int build_afilter_chain(sh_audio_t *sh_audio, ao_data_t *ao_data)
mpctx->mixer.afilter = NULL;
return 0;
}
new_srate = sh_audio->samplerate * playback_speed;
if (new_srate != ao_data->samplerate) {
// limits are taken from libaf/af_resample.c
if (new_srate < 8000)
new_srate = 8000;
if (new_srate > 192000)
new_srate = 192000;
playback_speed = (float)new_srate / (float)sh_audio->samplerate;
if(af_control_any_rev(sh_audio->afilter,
AF_CONTROL_PLAYBACK_SPEED | AF_CONTROL_SET,
&playback_speed)) {
new_srate = sh_audio->samplerate;
} else {
new_srate = sh_audio->samplerate * playback_speed;
if (new_srate != ao_data->samplerate) {
// limits are taken from libaf/af_resample.c
if (new_srate < 8000)
new_srate = 8000;
if (new_srate > 192000)
new_srate = 192000;
playback_speed = (float)new_srate / (float)sh_audio->samplerate;
}
}
result = init_audio_filters(sh_audio, new_srate,
&ao_data->samplerate, &ao_data->channels, &ao_data->format);