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mpv/libaf/af.c

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/*
* 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 "config.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "osdep/strsep.h"
#include "af.h"
// Static list of filters
extern af_info_t af_info_dummy;
extern af_info_t af_info_delay;
extern af_info_t af_info_channels;
extern af_info_t af_info_format;
extern af_info_t af_info_resample;
extern af_info_t af_info_volume;
extern af_info_t af_info_equalizer;
extern af_info_t af_info_gate;
extern af_info_t af_info_comp;
extern af_info_t af_info_pan;
extern af_info_t af_info_surround;
extern af_info_t af_info_sub;
extern af_info_t af_info_export;
extern af_info_t af_info_volnorm;
extern af_info_t af_info_extrastereo;
extern af_info_t af_info_lavcac3enc;
extern af_info_t af_info_lavcresample;
extern af_info_t af_info_sweep;
extern af_info_t af_info_hrtf;
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;
extern af_info_t af_info_stats;
extern af_info_t af_info_bs2b;
static af_info_t* filter_list[]={
&af_info_dummy,
&af_info_delay,
&af_info_channels,
&af_info_format,
&af_info_resample,
&af_info_volume,
&af_info_equalizer,
&af_info_gate,
&af_info_comp,
&af_info_pan,
&af_info_surround,
&af_info_sub,
#ifdef HAVE_SYS_MMAN_H
&af_info_export,
#endif
&af_info_volnorm,
&af_info_extrastereo,
&af_info_lavcac3enc,
&af_info_lavcresample,
&af_info_sweep,
&af_info_hrtf,
#ifdef CONFIG_LADSPA
&af_info_ladspa,
#endif
&af_info_center,
&af_info_sinesuppress,
&af_info_karaoke,
&af_info_scaletempo,
&af_info_stats,
#ifdef CONFIG_LIBBS2B
&af_info_bs2b,
#endif
NULL
};
// CPU speed
int* af_cpu_speed = NULL;
/* Find a filter in the static list of filters using it's name. This
function is used internally */
static af_info_t* af_find(char*name)
{
int i=0;
while(filter_list[i]){
if(!strcmp(filter_list[i]->name,name))
return filter_list[i];
i++;
}
mp_msg(MSGT_AFILTER, MSGL_ERR, "Couldn't find audio filter '%s'\n",name);
return NULL;
}
/* Find filter in the dynamic filter list using it's name This
function is used for finding already initialized filters */
af_instance_t* af_get(af_stream_t* s, char* name)
{
af_instance_t* af=s->first;
// Find the filter
while(af != NULL){
if(!strcmp(af->info->name,name))
return af;
af=af->next;
}
return NULL;
}
/*/ Function for creating a new filter of type name. The name may
contain the commandline parameters for the filter */
static af_instance_t* af_create(af_stream_t* s, const char* name_with_cmd)
{
char* name = strdup(name_with_cmd);
char* cmdline = name;
// Allocate space for the new filter and reset all pointers
af_instance_t* new=malloc(sizeof(af_instance_t));
if (!name || !new) {
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Could not allocate memory\n");
goto err_out;
}
memset(new,0,sizeof(af_instance_t));
// Check for commandline parameters
strsep(&cmdline, "=");
// Find filter from name
if(NULL == (new->info=af_find(name)))
goto err_out;
/* Make sure that the filter is not already in the list if it is
non-reentrant */
if(new->info->flags & AF_FLAGS_NOT_REENTRANT){
if(af_get(s,name)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] There can only be one instance of"
" the filter '%s' in each stream\n",name);
goto err_out;
}
}
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Adding filter %s \n",name);
// Initialize the new filter
if(AF_OK == new->info->open(new) &&
AF_ERROR < new->control(new,AF_CONTROL_POST_CREATE,&s->cfg)){
if(cmdline){
if(AF_ERROR>=new->control(new,AF_CONTROL_COMMAND_LINE,cmdline))
goto err_out;
}
free(name);
return new;
}
err_out:
free(new);
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Couldn't create or open audio filter '%s'\n",
name);
free(name);
return NULL;
}
/* Create and insert a new filter of type name before the filter in the
argument. This function can be called during runtime, the return
value is the new filter */
static af_instance_t* af_prepend(af_stream_t* s, af_instance_t* af, const char* name)
{
// Create the new filter and make sure it is OK
af_instance_t* new=af_create(s,name);
if(!new)
return NULL;
// Update pointers
new->next=af;
if(af){
new->prev=af->prev;
af->prev=new;
}
else
s->last=new;
if(new->prev)
new->prev->next=new;
else
s->first=new;
return new;
}
/* Create and insert a new filter of type name after the filter in the
argument. This function can be called during runtime, the return
value is the new filter */
static af_instance_t* af_append(af_stream_t* s, af_instance_t* af, const char* name)
{
// Create the new filter and make sure it is OK
af_instance_t* new=af_create(s,name);
if(!new)
return NULL;
// Update pointers
new->prev=af;
if(af){
new->next=af->next;
af->next=new;
}
else
s->first=new;
if(new->next)
new->next->prev=new;
else
s->last=new;
return new;
}
// Uninit and remove the filter "af"
void af_remove(af_stream_t* s, af_instance_t* af)
{
if(!af) return;
// Print friendly message
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Removing filter %s \n",af->info->name);
// Notify filter before changing anything
af->control(af,AF_CONTROL_PRE_DESTROY,0);
// Detach pointers
if(af->prev)
af->prev->next=af->next;
else
s->first=af->next;
if(af->next)
af->next->prev=af->prev;
else
s->last=af->prev;
// Uninitialize af and free memory
af->uninit(af);
free(af);
}
static void print_fmt(af_data_t *d)
{
if (d) {
mp_msg(MSGT_AFILTER, MSGL_V, "%dHz/%dch/%s", d->rate, d->nch,
af_fmt2str_short(d->format));
} else {
mp_msg(MSGT_AFILTER, MSGL_V, "(?)");
}
}
static void af_print_filter_chain(af_stream_t* s)
{
mp_msg(MSGT_AFILTER, MSGL_V, "Audio filter chain:\n");
mp_msg(MSGT_AFILTER, MSGL_V, " [in] ");
print_fmt(&s->input);
mp_msg(MSGT_AFILTER, MSGL_V, "\n");
af_instance_t *af = s->first;
while (af) {
mp_msg(MSGT_AFILTER, MSGL_V, " [%s] ", af->info->name);
print_fmt(af->data);
mp_msg(MSGT_AFILTER, MSGL_V, "\n");
af = af->next;
}
mp_msg(MSGT_AFILTER, MSGL_V, " [out] ");
print_fmt(&s->output);
mp_msg(MSGT_AFILTER, MSGL_V, "\n");
}
// Warning:
// A failed af_reinit() leaves the audio chain behind in a useless, broken
// state (for example, format filters that were tentatively inserted stay
// inserted).
// In that case, you should always rebuild the filter chain, or abort.
int af_reinit(af_stream_t* s, af_instance_t* af)
{
do{
af_data_t in; // Format of the input to current filter
int rv=0; // Return value
// Check if there are any filters left in the list
if(NULL == af){
if(!(af=af_append(s,s->first,"dummy")))
return AF_UNKNOWN;
else
return AF_ERROR;
}
// Check if this is the first filter
if(!af->prev)
memcpy(&in,&(s->input),sizeof(af_data_t));
else
memcpy(&in,af->prev->data,sizeof(af_data_t));
// Reset just in case...
in.audio=NULL;
in.len=0;
rv = af->control(af,AF_CONTROL_REINIT,&in);
switch(rv){
case AF_OK:
af = af->next;
break;
case AF_FALSE:{ // Configuration filter is needed
// Do auto insertion only if force is not specified
if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
af_instance_t* new = NULL;
// Insert channels filter
if((af->prev?af->prev->data->nch:s->input.nch) != in.nch){
// Create channels filter
if(NULL == (new = af_prepend(s,af,"channels")))
return AF_ERROR;
// Set number of output channels
if(AF_OK != (rv = new->control(new,AF_CONTROL_CHANNELS,&in.nch)))
return rv;
// Initialize channels filter
if(!new->prev)
memcpy(&in,&(s->input),sizeof(af_data_t));
else
memcpy(&in,new->prev->data,sizeof(af_data_t));
if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
return rv;
}
// Insert format filter
if((af->prev?af->prev->data->format:s->input.format) != in.format){
// Create format filter
if(NULL == (new = af_prepend(s,af,"format")))
return AF_ERROR;
// Set output bits per sample
in.format |= af_bits2fmt(in.bps*8);
if(AF_OK != (rv = new->control(new,AF_CONTROL_FORMAT_FMT,&in.format)))
return rv;
// Initialize format filter
if(!new->prev)
memcpy(&in,&(s->input),sizeof(af_data_t));
else
memcpy(&in,new->prev->data,sizeof(af_data_t));
if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
return rv;
}
if(!new){ // Should _never_ happen
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to correct audio format. "
"This error should never uccur, please send bugreport.\n");
return AF_ERROR;
}
af=new->next;
}
else {
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Automatic filter insertion disabled "
"but formats do not match. Giving up.\n");
return AF_ERROR;
}
break;
}
case AF_DETACH:{ // Filter is redundant and wants to be unloaded
// Do auto remove only if force is not specified
if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
af_instance_t* aft=af->prev;
af_remove(s,af);
if(aft)
af=aft->next;
else
af=s->first; // Restart configuration
}
break;
}
default:
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Reinitialization did not work, audio"
" filter '%s' returned error code %i\n",af->info->name,rv);
return AF_ERROR;
}
}while(af);
af_print_filter_chain(s);
return AF_OK;
}
// Uninit and remove all filters
void af_uninit(af_stream_t* s)
{
while(s->first)
af_remove(s,s->first);
}
/**
* Extend the filter chain so we get the required output format at the end.
* \return AF_ERROR on error, AF_OK if successful.
*/
static int fixup_output_format(af_stream_t* s)
{
af_instance_t* af = NULL;
// Check number of output channels fix if not OK
// If needed always inserted last -> easy to screw up other filters
if(s->output.nch && s->last->data->nch!=s->output.nch){
if(!strcmp(s->last->info->name,"format"))
af = af_prepend(s,s->last,"channels");
else
af = af_append(s,s->last,"channels");
// Init the new filter
if(!af || (AF_OK != af->control(af,AF_CONTROL_CHANNELS,&(s->output.nch))))
return AF_ERROR;
if(AF_OK != af_reinit(s,af))
return AF_ERROR;
}
// Check output format fix if not OK
if(s->output.format != AF_FORMAT_UNKNOWN &&
s->last->data->format != s->output.format){
if(strcmp(s->last->info->name,"format"))
af = af_append(s,s->last,"format");
else
af = s->last;
// Init the new filter
s->output.format |= af_bits2fmt(s->output.bps*8);
if(!af || (AF_OK != af->control(af,AF_CONTROL_FORMAT_FMT,&(s->output.format))))
return AF_ERROR;
if(AF_OK != af_reinit(s,af))
return AF_ERROR;
}
// Re init again just in case
if(AF_OK != af_reinit(s,s->first))
return AF_ERROR;
if (s->output.format == AF_FORMAT_UNKNOWN)
s->output.format = s->last->data->format;
if (!s->output.nch) s->output.nch = s->last->data->nch;
if (!s->output.rate) s->output.rate = s->last->data->rate;
if((s->last->data->format != s->output.format) ||
(s->last->data->nch != s->output.nch) ||
(s->last->data->rate != s->output.rate)) {
return AF_ERROR;
}
return AF_OK;
}
/**
* Automatic downmix to stereo in case the codec does not implement it.
*/
static void af_downmix(af_stream_t* s)
{
static const char * const downmix_strs[AF_NCH + 1] = {
/* FL FR RL RR FC LF AL AR */
[3] = "pan=2:" "0.6:0:" "0:0.6:" "0.4:0.4",
[4] = "pan=2:" "0.6:0:" "0:0.6:" "0.4:0:" "0:0.4",
[5] = "pan=2:" "0.5:0:" "0:0.5:" "0.2:0:" "0:0.2:" "0.3:0.3",
[6] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:" "0:0.2:" "0.3:0.3:" "0.1:0.1",
[7] = "pan=2:" "0.4:0:" "0:0.4:" "0.2:0:" "0:0.2:" "0.3:0.3:" "0.1:0:" "0:0.1",
[8] = "pan=2:" "0.4:0:" "0:0.4:" "0.15:0:" "0:0.15:" "0.25:0.25:" "0.1:0.1:" "0.1:0:" "0:0.1",
};
const char *af_pan_str = downmix_strs[s->input.nch];
if (af_pan_str)
af_append(s, s->first, af_pan_str);
}
/* Initialize the stream "s". This function creates a new filter list
if necessary according to the values set in input and output. Input
and output should contain the format of the current movie and the
formate of the preferred output respectively. The function is
reentrant i.e. if called with an already initialized stream the
stream will be reinitialized.
If one of the prefered output parameters is 0 the one that needs
no conversion is used (i.e. the output format in the last filter).
The return value is 0 if success and -1 if failure */
int af_init(af_stream_t* s)
{
struct MPOpts *opts = s->opts;
int i=0;
// Sanity check
if(!s) return -1;
// Precaution in case caller is misbehaving
s->input.audio = s->output.audio = NULL;
s->input.len = s->output.len = 0;
// Figure out how fast the machine is
if(AF_INIT_AUTO == (AF_INIT_TYPE_MASK & s->cfg.force))
s->cfg.force = (s->cfg.force & ~AF_INIT_TYPE_MASK) | AF_INIT_TYPE;
// Check if this is the first call
if(!s->first){
// Append a downmix pan filter at the beginning of the chain if needed
if (s->input.nch != opts->audio_output_channels
&& opts->audio_output_channels == 2)
af_downmix(s);
// Add all filters in the list (if there are any)
if (s->cfg.list) {
while(s->cfg.list[i]){
if(!af_append(s,s->last,s->cfg.list[i++]))
return -1;
}
}
}
// If we do not have any filters otherwise
// add dummy to make automatic format conversion work
if (!s->first && !af_append(s, s->first, "dummy"))
return -1;
// Init filters
if(AF_OK != af_reinit(s,s->first))
return -1;
// make sure the chain is not empty and valid (e.g. because of AF_DETACH)
if (!s->first)
if (!af_append(s,s->first,"dummy") || AF_OK != af_reinit(s,s->first))
return -1;
// Check output format
if((AF_INIT_TYPE_MASK & s->cfg.force) != AF_INIT_FORCE){
af_instance_t* af = NULL; // New filter
// Check output frequency if not OK fix with resample
if(s->output.rate && s->last->data->rate!=s->output.rate){
// try to find a filter that can change samplrate
af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
&(s->output.rate));
if (!af) {
char *resampler = "resample";
if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW)
resampler = "lavcresample";
if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){
if(!strcmp(s->first->info->name,"format"))
af = af_append(s,s->first,resampler);
else
af = af_prepend(s,s->first,resampler);
}
else{
if(!strcmp(s->last->info->name,"format"))
af = af_prepend(s,s->last,resampler);
else
af = af_append(s,s->last,resampler);
}
// Init the new filter
if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
&(s->output.rate))))
return -1;
// Use lin int if the user wants fast
if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_FAST) {
char args[32];
sprintf(args, "%d", s->output.rate);
if (strcmp(resampler, "lavcresample") == 0)
strcat(args, ":1");
else
strcat(args, ":0:0");
af->control(af, AF_CONTROL_COMMAND_LINE, args);
}
}
if(AF_OK != af_reinit(s,af))
return -1;
}
if (AF_OK != fixup_output_format(s)) {
// Something is stuffed audio out will not work
mp_msg(MSGT_AFILTER, MSGL_ERR, "[libaf] Unable to setup filter system can not"
" meet sound-card demands, please send bugreport. \n");
af_uninit(s);
return -1;
}
}
return 0;
}
/* Add filter during execution. This function adds the filter "name"
to the stream s. The filter will be inserted somewhere nice in the
list of filters. The return value is a pointer to the new filter,
If the filter couldn't be added the return value is NULL. */
af_instance_t* af_add(af_stream_t* s, char* name){
af_instance_t* new;
// Sanity check
if(!s || !s->first || !name)
return NULL;
// Insert the filter somwhere nice
if(!strcmp(s->first->info->name,"format"))
new = af_append(s, s->first, name);
else
new = af_prepend(s, s->first, name);
if(!new)
return NULL;
// Reinitalize the filter list
if(AF_OK != af_reinit(s, s->first) ||
AF_OK != fixup_output_format(s)){
while (s->first)
af_remove(s, s->first);
af_init(s);
return NULL;
}
return new;
}
// Filter data chunk through the filters in the list
af_data_t* af_play(af_stream_t* s, af_data_t* data)
{
af_instance_t* af=s->first;
// Iterate through all filters
do{
if (data->len <= 0) break;
data=af->play(af,data);
af=af->next;
}while(af && data);
return data;
}
/* Calculate the minimum output buffer size for given input data d
* when using the RESIZE_LOCAL_BUFFER macro. The +t+1 part ensures the
* value is >= len*mul rounded upwards to whole samples even if the
* double 'mul' is inexact. */
int af_lencalc(double mul, af_data_t* d)
{
int t = d->bps * d->nch;
return d->len * mul + t + 1;
}
// Calculate average ratio of filter output size to input size
double af_calc_filter_multiplier(af_stream_t* s)
{
af_instance_t* af=s->first;
double mul = 1;
// Iterate through all filters and calculate total multiplication factor
do{
mul *= af->mul;
af=af->next;
}while(af);
return mul;
}
/* Calculate the total delay [bytes output] caused by the filters */
double af_calc_delay(af_stream_t* s)
{
af_instance_t* af=s->first;
register double delay = 0.0;
// Iterate through all filters
while(af){
delay += af->delay;
delay *= af->mul;
af=af->next;
}
return delay;
}
/* Helper function called by the macro with the same name this
function should not be called directly */
int af_resize_local_buffer(af_instance_t* af, af_data_t* data)
{
// Calculate new length
register int len = af_lencalc(af->mul,data);
mp_msg(MSGT_AFILTER, MSGL_V, "[libaf] Reallocating memory in module %s, "
"old len = %i, new len = %i\n",af->info->name,af->data->len,len);
// If there is a buffer free it
free(af->data->audio);
// Create new buffer and check that it is OK
af->data->audio = malloc(len);
if(!af->data->audio){
mp_msg(MSGT_AFILTER, MSGL_FATAL, "[libaf] Could not allocate memory \n");
return AF_ERROR;
}
af->data->len=len;
return AF_OK;
}
// documentation in af.h
af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg) {
int res = AF_UNKNOWN;
af_instance_t* filt = s->last;
while (filt) {
res = filt->control(filt, cmd, arg);
if (res == AF_OK)
return filt;
filt = filt->prev;
}
return NULL;
}
void af_help (void) {
int i = 0;
mp_msg(MSGT_AFILTER, MSGL_INFO, "Available audio filters:\n");
while (filter_list[i]) {
if (filter_list[i]->comment && filter_list[i]->comment[0])
mp_msg(MSGT_AFILTER, MSGL_INFO, " %-15s: %s (%s)\n", filter_list[i]->name, filter_list[i]->info, filter_list[i]->comment);
else
mp_msg(MSGT_AFILTER, MSGL_INFO, " %-15s: %s\n", filter_list[i]->name, filter_list[i]->info);
i++;
}
}
void af_fix_parameters(af_data_t *data)
{
data->bps = af_fmt2bits(data->format)/8;
}
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