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authorStefano Pigozzi <stefano.pigozzi@gmail.com>2012-11-01 12:00:00 +0100
committerStefano Pigozzi <stefano.pigozzi@gmail.com>2012-11-02 19:19:28 +0100
commit0374ddb79d4e20aa73ef91496beb2c0811c34ca7 (patch)
treec9066550112a2b905c2e87351c10bd484ebe04ba
parente0aef8cf1275cc988007fcb2a099cdd9f92fa374 (diff)
downloadmpv-0374ddb79d4e20aa73ef91496beb2c0811c34ca7.tar.bz2
mpv-0374ddb79d4e20aa73ef91496beb2c0811c34ca7.tar.xz
audio: untypedef af_data and rename it to mp_audio
this is to have something specular to mp_image
-rw-r--r--libaf/af.c24
-rw-r--r--libaf/af.h25
-rw-r--r--libaf/af_bs2b.c12
-rw-r--r--libaf/af_center.c12
-rw-r--r--libaf/af_channels.c24
-rw-r--r--libaf/af_delay.c14
-rw-r--r--libaf/af_dummy.c6
-rw-r--r--libaf/af_equalizer.c10
-rw-r--r--libaf/af_export.c12
-rw-r--r--libaf/af_extrastereo.c16
-rw-r--r--libaf/af_format.c36
-rw-r--r--libaf/af_hrtf.c10
-rw-r--r--libaf/af_karaoke.c12
-rw-r--r--libaf/af_ladspa.c10
-rw-r--r--libaf/af_lavcac3enc.c10
-rw-r--r--libaf/af_lavcresample.c8
-rw-r--r--libaf/af_pan.c22
-rw-r--r--libaf/af_resample.c14
-rw-r--r--libaf/af_scaletempo.c12
-rw-r--r--libaf/af_sinesuppress.c16
-rw-r--r--libaf/af_sub.c12
-rw-r--r--libaf/af_surround.c16
-rw-r--r--libaf/af_sweep.c6
-rw-r--r--libaf/af_tools.c4
-rw-r--r--libaf/af_volnorm.c20
-rw-r--r--libaf/af_volume.c14
-rw-r--r--libaf/control.h2
-rw-r--r--libmpcodecs/dec_audio.c4
28 files changed, 191 insertions, 192 deletions
diff --git a/libaf/af.c b/libaf/af.c
index 7596376a99..f2745c5b59 100644
--- a/libaf/af.c
+++ b/libaf/af.c
@@ -240,7 +240,7 @@ void af_remove(af_stream_t* s, af_instance_t* af)
free(af);
}
-static void print_fmt(af_data_t *d)
+static void print_fmt(struct mp_audio *d)
{
if (d) {
mp_msg(MSGT_AFILTER, MSGL_V, "%dHz/%dch/%s", d->rate, d->nch,
@@ -280,7 +280,7 @@ static void af_print_filter_chain(af_stream_t* s)
int af_reinit(af_stream_t* s, af_instance_t* af)
{
do{
- af_data_t in; // Format of the input to current filter
+ struct mp_audio in; // Format of the input to current filter
int rv=0; // Return value
// Check if there are any filters left in the list
@@ -293,9 +293,9 @@ int af_reinit(af_stream_t* s, af_instance_t* af)
// Check if this is the first filter
if(!af->prev)
- memcpy(&in,&(s->input),sizeof(af_data_t));
+ memcpy(&in,&(s->input),sizeof(struct mp_audio));
else
- memcpy(&in,af->prev->data,sizeof(af_data_t));
+ memcpy(&in,af->prev->data,sizeof(struct mp_audio));
// Reset just in case...
in.audio=NULL;
in.len=0;
@@ -319,9 +319,9 @@ int af_reinit(af_stream_t* s, af_instance_t* af)
return rv;
// Initialize channels filter
if(!new->prev)
- memcpy(&in,&(s->input),sizeof(af_data_t));
+ memcpy(&in,&(s->input),sizeof(struct mp_audio));
else
- memcpy(&in,new->prev->data,sizeof(af_data_t));
+ memcpy(&in,new->prev->data,sizeof(struct mp_audio));
if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
return rv;
}
@@ -336,9 +336,9 @@ int af_reinit(af_stream_t* s, af_instance_t* af)
return rv;
// Initialize format filter
if(!new->prev)
- memcpy(&in,&(s->input),sizeof(af_data_t));
+ memcpy(&in,&(s->input),sizeof(struct mp_audio));
else
- memcpy(&in,new->prev->data,sizeof(af_data_t));
+ memcpy(&in,new->prev->data,sizeof(struct mp_audio));
if(AF_OK != (rv = new->control(new,AF_CONTROL_REINIT,&in)))
return rv;
}
@@ -595,7 +595,7 @@ af_instance_t* af_add(af_stream_t* s, char* name){
}
// Filter data chunk through the filters in the list
-af_data_t* af_play(af_stream_t* s, af_data_t* data)
+struct mp_audio* af_play(af_stream_t* s, struct mp_audio* data)
{
af_instance_t* af=s->first;
// Iterate through all filters
@@ -611,7 +611,7 @@ af_data_t* af_play(af_stream_t* s, af_data_t* data)
* 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 af_lencalc(double mul, struct mp_audio* d)
{
int t = d->bps * d->nch;
return d->len * mul + t + 1;
@@ -647,7 +647,7 @@ double af_calc_delay(af_stream_t* s)
/* 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)
+int af_resize_local_buffer(af_instance_t* af, struct mp_audio* data)
{
// Calculate new length
register int len = af_lencalc(af->mul,data);
@@ -690,7 +690,7 @@ void af_help (void) {
}
}
-void af_fix_parameters(af_data_t *data)
+void af_fix_parameters(struct mp_audio *data)
{
if (data->nch < 0 || data->nch > AF_NCH) {
mp_msg(MSGT_AFILTER, MSGL_ERR, "Invalid number of channels %i, assuming 2.\n", data->nch);
diff --git a/libaf/af.h b/libaf/af.h
index 4542b32c60..e782759f77 100644
--- a/libaf/af.h
+++ b/libaf/af.h
@@ -37,15 +37,14 @@ struct af_instance_s;
#endif
// Audio data chunk
-typedef struct af_data_s
-{
+struct mp_audio {
void* audio; // data buffer
int len; // buffer length
int rate; // sample rate
int nch; // number of channels
int format; // format
int bps; // bytes per sample
-} af_data_t;
+};
// Flags used for defining the behavior of an audio filter
@@ -70,9 +69,9 @@ typedef struct af_instance_s
af_info_t* info;
int (*control)(struct af_instance_s* af, int cmd, void* arg);
void (*uninit)(struct af_instance_s* af);
- af_data_t* (*play)(struct af_instance_s* af, af_data_t* data);
+ struct mp_audio* (*play)(struct af_instance_s* af, struct mp_audio* data);
void* setup; // setup data for this specific instance and filter
- af_data_t* data; // configuration for outgoing data stream
+ struct mp_audio* data; // configuration for outgoing data stream
struct af_instance_s* next;
struct af_instance_s* prev;
double delay; /* Delay caused by the filter, in units of bytes read without
@@ -113,8 +112,8 @@ typedef struct af_stream
af_instance_t* first;
af_instance_t* last;
// Storage for input and output data formats
- af_data_t input;
- af_data_t output;
+ struct mp_audio input;
+ struct mp_audio output;
// Configuration for this stream
af_cfg_t cfg;
struct MPOpts *opts;
@@ -203,7 +202,7 @@ af_instance_t* af_get(af_stream_t* s, char* name);
* \return resulting data
* \ingroup af_chain
*/
-af_data_t* af_play(af_stream_t* s, af_data_t* data);
+struct mp_audio* af_play(af_stream_t* s, struct mp_audio* data);
/**
* \brief send control to all filters, starting with the last until
@@ -237,12 +236,12 @@ double af_calc_delay(af_stream_t* s);
/* Helper function called by the macro with the same name only to be
called from inside filters */
-int af_resize_local_buffer(af_instance_t* af, af_data_t* data);
+int af_resize_local_buffer(af_instance_t* af, struct mp_audio* data);
/* Helper function used to calculate the exact buffer length needed
when buffers are resized. The returned length is >= than what is
needed */
-int af_lencalc(double mul, af_data_t* data);
+int af_lencalc(double mul, struct mp_audio* data);
/**
* \brief convert dB to gain value
@@ -297,7 +296,7 @@ int af_to_ms(int n, int* in, float* out, int rate);
*
* compares the format, bps, rate and nch values of af->data with out
*/
-int af_test_output(struct af_instance_s* af, af_data_t* out);
+int af_test_output(struct af_instance_s* af, struct mp_audio* out);
/**
* \brief soft clipping function using sin()
@@ -312,13 +311,13 @@ float af_softclip(float a);
void af_help(void);
/**
- * \brief fill the missing parameters in the af_data_t structure
+ * \brief fill the missing parameters in the struct mp_audio structure
* \param data structure to fill
* \ingroup af_filter
*
* Currently only sets bps based on format
*/
-void af_fix_parameters(af_data_t *data);
+void af_fix_parameters(struct mp_audio *data);
/** Memory reallocation macro: if a local buffer is used (i.e. if the
filter doesn't operate on the incoming buffer this macro must be
diff --git a/libaf/af_bs2b.c b/libaf/af_bs2b.c
index 14d31c35be..100ad02aa1 100644
--- a/libaf/af_bs2b.c
+++ b/libaf/af_bs2b.c
@@ -38,7 +38,7 @@ struct af_bs2b {
};
#define PLAY(name, type) \
-static af_data_t *play_##name(struct af_instance_s *af, af_data_t *data) \
+static struct mp_audio *play_##name(struct af_instance_s *af, struct mp_audio *data) \
{ \
/* filter is called for all pairs of samples available in the buffer */ \
bs2b_cross_feed_##name(((struct af_bs2b*)(af->setup))->filter, \
@@ -103,10 +103,10 @@ static int control(struct af_instance_s *af, int cmd, void *arg)
// Sanity check
if (!arg) return AF_ERROR;
- format = ((af_data_t*)arg)->format;
- af->data->rate = ((af_data_t*)arg)->rate;
+ format = ((struct mp_audio*)arg)->format;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
af->data->nch = 2; // bs2b is useful only for 2ch audio
- af->data->bps = ((af_data_t*)arg)->bps;
+ af->data->bps = ((struct mp_audio*)arg)->bps;
af->data->format = format;
/* check for formats supported by libbs2b
@@ -179,7 +179,7 @@ static int control(struct af_instance_s *af, int cmd, void *arg)
mp_msg(MSGT_AFILTER, MSGL_V, "[bs2b] using format %s\n",
af_fmt2str(af->data->format,buf,256));
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE: {
const opt_t subopts[] = {
@@ -243,7 +243,7 @@ static int af_open(af_instance_t *af)
af->control = control;
af->uninit = uninit;
af->mul = 1;
- if (!(af->data = calloc(1, sizeof(af_data_t))))
+ if (!(af->data = calloc(1, sizeof(struct mp_audio))))
return AF_ERROR;
if (!(af->setup = s = calloc(1, sizeof(struct af_bs2b)))) {
free(af->data);
diff --git a/libaf/af_center.c b/libaf/af_center.c
index 1cc3626439..e0897d5e65 100644
--- a/libaf/af_center.c
+++ b/libaf/af_center.c
@@ -47,12 +47,12 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = max(s->ch+1,((af_data_t*)arg)->nch);
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = max(s->ch+1,((struct mp_audio*)arg)->nch);
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
int ch=1;
@@ -83,9 +83,9 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
af_center_t* s = af->setup; // Setup for this instance
float* a = c->audio; // Audio data
int len = c->len/4; // Number of samples in current audio block
@@ -109,7 +109,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=s=calloc(1,sizeof(af_center_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_channels.c b/libaf/af_channels.c
index b42cde380a..671d9aa32a 100644
--- a/libaf/af_channels.c
+++ b/libaf/af_channels.c
@@ -143,11 +143,11 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
if(!s->router){
int i;
// Make sure this filter isn't redundant
- if(af->data->nch == ((af_data_t*)arg)->nch)
+ if(af->data->nch == ((struct mp_audio*)arg)->nch)
return AF_DETACH;
// If mono: fake stereo
- if(((af_data_t*)arg)->nch == 1){
+ if(((struct mp_audio*)arg)->nch == 1){
s->nr = min(af->data->nch,2);
for(i=0;i<s->nr;i++){
s->route[i][FR] = 0;
@@ -155,7 +155,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
}
}
else{
- s->nr = min(af->data->nch, ((af_data_t*)arg)->nch);
+ s->nr = min(af->data->nch, ((struct mp_audio*)arg)->nch);
for(i=0;i<s->nr;i++){
s->route[i][FR] = i;
s->route[i][TO] = i;
@@ -163,11 +163,11 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
}
}
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->format = ((af_data_t*)arg)->format;
- af->data->bps = ((af_data_t*)arg)->bps;
- af->mul = (double)af->data->nch / ((af_data_t*)arg)->nch;
- return check_routes(s,((af_data_t*)arg)->nch,af->data->nch);
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->format = ((struct mp_audio*)arg)->format;
+ af->data->bps = ((struct mp_audio*)arg)->bps;
+ af->mul = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
+ return check_routes(s,((struct mp_audio*)arg)->nch,af->data->nch);
case AF_CONTROL_COMMAND_LINE:{
int nch = 0;
int n = 0;
@@ -256,10 +256,10 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
- af_data_t* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
af_channels_t* s = af->setup;
int i;
@@ -288,7 +288,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_channels_t));
if((af->data == NULL) || (af->setup == NULL))
return AF_ERROR;
diff --git a/libaf/af_delay.c b/libaf/af_delay.c
index f0a9704eaa..15e0c7071f 100644
--- a/libaf/af_delay.c
+++ b/libaf/af_delay.c
@@ -52,10 +52,10 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
for(i=0;i<af->data->nch;i++)
free(s->q[i]);
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
- af->data->format = ((af_data_t*)arg)->format;
- af->data->bps = ((af_data_t*)arg)->bps;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
+ af->data->format = ((struct mp_audio*)arg)->format;
+ af->data->bps = ((struct mp_audio*)arg)->bps;
// Allocate new delay queues
for(i=0;i<af->data->nch;i++){
@@ -118,9 +118,9 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
af_delay_t* s = af->setup; // Setup for this instance
int nch = c->nch; // Number of channels
int len = c->len/c->bps; // Number of sample in data chunk
@@ -182,7 +182,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_delay_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_dummy.c b/libaf/af_dummy.c
index ba921eb09b..26aa9b5e22 100644
--- a/libaf/af_dummy.c
+++ b/libaf/af_dummy.c
@@ -30,7 +30,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:
- memcpy(af->data,(af_data_t*)arg,sizeof(af_data_t));
+ memcpy(af->data,(struct mp_audio*)arg,sizeof(struct mp_audio));
mp_msg(MSGT_AFILTER, MSGL_V, "[dummy] Was reinitialized: %iHz/%ich/%s\n",
af->data->rate,af->data->nch,af_fmt2str_short(af->data->format));
return AF_OK;
@@ -45,7 +45,7 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
// Do something necessary to get rid of annoying warning during compile
if(!af)
@@ -59,7 +59,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=malloc(sizeof(af_data_t));
+ af->data=malloc(sizeof(struct mp_audio));
if(af->data == NULL)
return AF_ERROR;
return AF_OK;
diff --git a/libaf/af_equalizer.c b/libaf/af_equalizer.c
index 318b7a72d0..112926dee6 100644
--- a/libaf/af_equalizer.c
+++ b/libaf/af_equalizer.c
@@ -96,8 +96,8 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
@@ -186,9 +186,9 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup
uint32_t ci = af->data->nch; // Index for channels
uint32_t nch = af->data->nch; // Number of channels
@@ -230,7 +230,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_equalizer_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_export.c b/libaf/af_export.c
index b5e5a884c0..0239791905 100644
--- a/libaf/af_export.c
+++ b/libaf/af_export.c
@@ -84,8 +84,8 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
close(s->fd);
// Accept only int16_t as input format (which sucks)
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
@@ -129,7 +129,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
msync(s->mmap_area, mapsize, MS_ASYNC);
// Use test_output to return FALSE if necessary
- return af_test_output(af, (af_data_t*)arg);
+ return af_test_output(af, (struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
int i=0;
@@ -201,9 +201,9 @@ static void uninit( struct af_instance_s* af )
af audio filter instance
data audio data
*/
-static af_data_t* play( struct af_instance_s* af, af_data_t* data )
+static struct mp_audio* play( struct af_instance_s* af, struct mp_audio* data )
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
af_export_t* s = af->setup; // Setup for this instance
int16_t* a = c->audio; // Incomming sound
int nch = c->nch; // Number of channels
@@ -252,7 +252,7 @@ static int af_open( af_instance_t* af )
af->uninit = uninit;
af->play = play;
af->mul=1;
- af->data = calloc(1, sizeof(af_data_t));
+ af->data = calloc(1, sizeof(struct mp_audio));
af->setup = calloc(1, sizeof(af_export_t));
if((af->data == NULL) || (af->setup == NULL))
return AF_ERROR;
diff --git a/libaf/af_extrastereo.c b/libaf/af_extrastereo.c
index 347c257137..c1ae719c31 100644
--- a/libaf/af_extrastereo.c
+++ b/libaf/af_extrastereo.c
@@ -34,8 +34,8 @@ typedef struct af_extrastereo_s
float mul;
}af_extrastereo_t;
-static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data);
-static af_data_t* play_float(struct af_instance_s* af, af_data_t* data);
+static struct mp_audio* play_s16(struct af_instance_s* af, struct mp_audio* data);
+static struct mp_audio* play_float(struct af_instance_s* af, struct mp_audio* data);
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
@@ -47,9 +47,9 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
af->data->nch = 2;
- if (((af_data_t*)arg)->format == AF_FORMAT_FLOAT_NE)
+ if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
{
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
@@ -61,7 +61,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
af->play = play_s16;
}
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
float f;
@@ -87,7 +87,7 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_s16(struct af_instance_s* af, struct mp_audio* data)
{
af_extrastereo_t *s = af->setup;
register int i = 0;
@@ -109,7 +109,7 @@ static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data)
return data;
}
-static af_data_t* play_float(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_float(struct af_instance_s* af, struct mp_audio* data)
{
af_extrastereo_t *s = af->setup;
register int i = 0;
@@ -137,7 +137,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play_s16;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_extrastereo_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_format.c b/libaf/af_format.c
index ea9f39e2e6..a9d1fe6c88 100644
--- a/libaf/af_format.c
+++ b/libaf/af_format.c
@@ -52,10 +52,10 @@ static void float2int(float* in, void* out, int len, int bps);
// From signed int to float
static void int2float(void* in, float* out, int len, int bps);
-static af_data_t* play(struct af_instance_s* af, af_data_t* data);
-static af_data_t* play_swapendian(struct af_instance_s* af, af_data_t* data);
-static af_data_t* play_float_s16(struct af_instance_s* af, af_data_t* data);
-static af_data_t* play_s16_float(struct af_instance_s* af, af_data_t* data);
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data);
+static struct mp_audio* play_swapendian(struct af_instance_s* af, struct mp_audio* data);
+static struct mp_audio* play_float_s16(struct af_instance_s* af, struct mp_audio* data);
+static struct mp_audio* play_s16_float(struct af_instance_s* af, struct mp_audio* data);
// Helper functions to check sanity for input arguments
@@ -92,7 +92,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
case AF_CONTROL_REINIT:{
char buf1[256];
char buf2[256];
- af_data_t *data = arg;
+ struct mp_audio *data = arg;
// Make sure this filter isn't redundant
if(af->data->format == data->format &&
@@ -176,10 +176,10 @@ static void uninit(struct af_instance_s* af)
af->setup = 0;
}
-static af_data_t* play_swapendian(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_swapendian(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* l = af->data; // Local data
- af_data_t* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
int len = c->len/c->bps; // Length in samples of current audio block
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
@@ -193,10 +193,10 @@ static af_data_t* play_swapendian(struct af_instance_s* af, af_data_t* data)
return c;
}
-static af_data_t* play_float_s16(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_float_s16(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* l = af->data; // Local data
- af_data_t* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
int len = c->len/4; // Length in samples of current audio block
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
@@ -212,10 +212,10 @@ static af_data_t* play_float_s16(struct af_instance_s* af, af_data_t* data)
return c;
}
-static af_data_t* play_s16_float(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_s16_float(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* l = af->data; // Local data
- af_data_t* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
int len = c->len/2; // Length in samples of current audio block
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
@@ -232,10 +232,10 @@ static af_data_t* play_s16_float(struct af_instance_s* af, af_data_t* data)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* l = af->data; // Local data
- af_data_t* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
int len = c->len/c->bps; // Length in samples of current audio block
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
@@ -318,7 +318,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
if(af->data == NULL)
return AF_ERROR;
return AF_OK;
diff --git a/libaf/af_hrtf.c b/libaf/af_hrtf.c
index 4edf224de9..1aab8adcf6 100644
--- a/libaf/af_hrtf.c
+++ b/libaf/af_hrtf.c
@@ -290,7 +290,7 @@ static int control(struct af_instance_s *af, int cmd, void* arg)
switch(cmd) {
case AF_CONTROL_REINIT:
- af->data->rate = ((af_data_t*)arg)->rate;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
if(af->data->rate != 48000) {
// automatic samplerate adjustment in the filter chain
// is not yet supported.
@@ -299,7 +299,7 @@ static int control(struct af_instance_s *af, int cmd, void* arg)
af->data->rate);
return AF_ERROR;
}
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
if(af->data->nch == 2) {
/* 2 channel input */
if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
@@ -311,7 +311,7 @@ static int control(struct af_instance_s *af, int cmd, void* arg)
af->data->nch = 5;
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
- test_output_res = af_test_output(af, (af_data_t*)arg);
+ test_output_res = af_test_output(af, (struct mp_audio*)arg);
af->mul = 2.0 / af->data->nch;
// after testing input set the real output format
af->data->nch = 2;
@@ -381,7 +381,7 @@ frequencies).
2. A bass compensation is introduced to ensure that 0-200 Hz are not
damped (without any real 3D acoustical image, however).
*/
-static af_data_t* play(struct af_instance_s *af, af_data_t *data)
+static struct mp_audio* play(struct af_instance_s *af, struct mp_audio *data)
{
af_hrtf_t *s = af->setup;
short *in = data->audio; // Input audio data
@@ -603,7 +603,7 @@ static int af_open(af_instance_t* af)
af->uninit = uninit;
af->play = play;
af->mul = 1;
- af->data = calloc(1, sizeof(af_data_t));
+ af->data = calloc(1, sizeof(struct mp_audio));
af->setup = calloc(1, sizeof(af_hrtf_t));
if((af->data == NULL) || (af->setup == NULL))
return AF_ERROR;
diff --git a/libaf/af_karaoke.c b/libaf/af_karaoke.c
index 780349dfee..1e8e313fa9 100644
--- a/libaf/af_karaoke.c
+++ b/libaf/af_karaoke.c
@@ -34,11 +34,11 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
{
switch(cmd){
case AF_CONTROL_REINIT:
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
af->data->format= AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
}
return AF_UNKNOWN;
}
@@ -50,9 +50,9 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
float* a = c->audio; // Audio data
int len = c->len/4; // Number of samples in current audio block
int nch = c->nch; // Number of channels
@@ -79,7 +79,7 @@ static int af_open(af_instance_t* af){
af->uninit = uninit;
af->play = play;
af->mul = 1;
- af->data = calloc(1,sizeof(af_data_t));
+ af->data = calloc(1,sizeof(struct mp_audio));
if(af->data == NULL)
return AF_ERROR;
diff --git a/libaf/af_ladspa.c b/libaf/af_ladspa.c
index 30693f09f2..0c83024b70 100644
--- a/libaf/af_ladspa.c
+++ b/libaf/af_ladspa.c
@@ -498,8 +498,8 @@ static int control(struct af_instance_s *af, int cmd, void *arg) {
/* accept FLOAT, let af_format do conversion */
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
@@ -507,7 +507,7 @@ static int control(struct af_instance_s *af, int cmd, void *arg) {
* filter, has to be done in play() :-/
*/
- return af_test_output(af, (af_data_t*)arg);
+ return af_test_output(af, (struct mp_audio*)arg);
case AF_CONTROL_COMMAND_LINE: {
char *buf;
char *line = arg;
@@ -710,7 +710,7 @@ static void uninit(struct af_instance_s *af) {
* \return Either AF_ERROR or AF_OK
*/
-static af_data_t* play(struct af_instance_s *af, af_data_t *data) {
+static struct mp_audio* play(struct af_instance_s *af, struct mp_audio *data) {
af_ladspa_t *setup = af->setup;
const LADSPA_Descriptor *pdes = setup->plugin_descriptor;
float *audio = (float*)data->audio;
@@ -889,7 +889,7 @@ static int af_open(af_instance_t *af) {
af->play=play;
af->mul=1;
- af->data = calloc(1, sizeof(af_data_t));
+ af->data = calloc(1, sizeof(struct mp_audio));
if (af->data == NULL)
return af_ladspa_malloc_failed((char*)af_info_ladspa.name);
diff --git a/libaf/af_lavcac3enc.c b/libaf/af_lavcac3enc.c
index ca0fd39a4e..a67eb28daf 100644
--- a/libaf/af_lavcac3enc.c
+++ b/libaf/af_lavcac3enc.c
@@ -61,7 +61,7 @@ typedef struct af_ac3enc_s {
static int control(struct af_instance_s *af, int cmd, void *arg)
{
af_ac3enc_t *s = af->setup;
- af_data_t *data = arg;
+ struct mp_audio *data = arg;
int i, bit_rate, test_output_res;
static const int default_bit_rate[AC3_MAX_CHANNELS+1] = \
{0, 96000, 192000, 256000, 384000, 448000, 448000};
@@ -170,11 +170,11 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
af_ac3enc_t *s = af->setup;
- af_data_t *c = data; // Current working data
- af_data_t *l;
+ struct mp_audio *c = data; // Current working data
+ struct mp_audio *l;
int len, left, outsize = 0, destsize;
char *buf, *src, *dest;
int max_output_len;
@@ -282,7 +282,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=s;
s->lavc_acodec = avcodec_find_encoder_by_name("ac3");
diff --git a/libaf/af_lavcresample.c b/libaf/af_lavcresample.c
index f1483aca77..e32d3acd9d 100644
--- a/libaf/af_lavcresample.c
+++ b/libaf/af_lavcresample.c
@@ -53,7 +53,7 @@ typedef struct af_resample_s{
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_resample_t* s = (af_resample_t*)af->setup;
- af_data_t *data= (af_data_t*)arg;
+ struct mp_audio *data= (struct mp_audio*)arg;
int out_rate, test_output_res; // helpers for checking input format
switch(cmd){
@@ -83,7 +83,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// 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, (af_data_t*)arg);
+ 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:{
@@ -116,7 +116,7 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
af_resample_t *s = af->setup;
int i, j, consumed, ret = 0;
@@ -194,7 +194,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ 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;
diff --git a/libaf/af_pan.c b/libaf/af_pan.c
index e3f7d29d1c..6acc4be079 100644
--- a/libaf/af_pan.c
+++ b/libaf/af_pan.c
@@ -44,16 +44,16 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
- af->data->nch = s->nch ? s->nch: ((af_data_t*)arg)->nch;
- af->mul = (double)af->data->nch / ((af_data_t*)arg)->nch;
+ af->data->nch = s->nch ? s->nch: ((struct mp_audio*)arg)->nch;
+ af->mul = (double)af->data->nch / ((struct mp_audio*)arg)->nch;
- if((af->data->format != ((af_data_t*)arg)->format) ||
- (af->data->bps != ((af_data_t*)arg)->bps)){
- ((af_data_t*)arg)->format = af->data->format;
- ((af_data_t*)arg)->bps = af->data->bps;
+ if((af->data->format != ((struct mp_audio*)arg)->format) ||
+ (af->data->bps != ((struct mp_audio*)arg)->bps)){
+ ((struct mp_audio*)arg)->format = af->data->format;
+ ((struct mp_audio*)arg)->bps = af->data->bps;
return AF_FALSE;
}
return AF_OK;
@@ -148,10 +148,10 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
- af_data_t* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
af_pan_t* s = af->setup; // Setup for this instance
float* in = c->audio; // Input audio data
float* out = NULL; // Output audio data
@@ -192,7 +192,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_pan_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_resample.c b/libaf/af_resample.c
index 6809c48125..ad6a26a02d 100644
--- a/libaf/af_resample.c
+++ b/libaf/af_resample.c
@@ -75,7 +75,7 @@ typedef struct af_resample_s
} af_resample_t;
// Fast linear interpolation resample with modest audio quality
-static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
+static int linint(struct mp_audio* c,struct mp_audio* l, af_resample_t* s)
{
uint32_t len = 0; // Number of input samples
uint32_t nch = l->nch; // Words pre transfer
@@ -122,7 +122,7 @@ static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
}
/* Determine resampling type and format */
-static int set_types(struct af_instance_s* af, af_data_t* data)
+static int set_types(struct af_instance_s* af, struct mp_audio* data)
{
af_resample_t* s = af->setup;
int rv = AF_OK;
@@ -175,7 +175,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
switch(cmd){
case AF_CONTROL_REINIT:{
af_resample_t* s = af->setup;
- af_data_t* n = arg; // New configuration
+ struct mp_audio* n = arg; // New configuration
int i,d = 0;
int rv = AF_OK;
@@ -317,11 +317,11 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
int len = 0; // Length of output data
- af_data_t* c = data; // Current working data
- af_data_t* l = af->data; // Local data
+ struct mp_audio* c = data; // Current working data
+ struct mp_audio* l = af->data; // Local data
af_resample_t* s = af->setup;
if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
@@ -375,7 +375,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_resample_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_scaletempo.c b/libaf/af_scaletempo.c
index c0197d5b40..151a33b874 100644
--- a/libaf/af_scaletempo.c
+++ b/libaf/af_scaletempo.c
@@ -78,7 +78,7 @@ typedef struct af_scaletempo_s
short speed_pitch;
} af_scaletempo_t;
-static int fill_queue(struct af_instance_s* af, af_data_t* data, int offset)
+static int fill_queue(struct af_instance_s* af, struct mp_audio* data, int offset)
{
af_scaletempo_t* s = af->setup;
int bytes_in = data->len - offset;
@@ -219,7 +219,7 @@ static void output_overlap_s16(af_scaletempo_t* s, void* buf_out,
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
af_scaletempo_t* s = af->setup;
int offset_in;
@@ -290,7 +290,7 @@ 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;
+ struct mp_audio* data = (struct mp_audio*)arg;
float srate = data->rate / 1000;
int nch = data->nch;
int bps;
@@ -305,7 +305,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
if (s->scale == 1.0) {
if (s->speed_tempo && s->speed_pitch)
return AF_DETACH;
- memcpy(af->data, data, sizeof(af_data_t));
+ memcpy(af->data, data, sizeof(struct mp_audio));
af->delay = 0;
af->mul = 1;
return af_test_output(af, data);
@@ -439,7 +439,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
(int)(s->bytes_queue / nch / bps),
(use_int?"s16":"float"));
- return af_test_output(af, (af_data_t*)arg);
+ return af_test_output(af, (struct mp_audio*)arg);
}
case AF_CONTROL_PLAYBACK_SPEED | AF_CONTROL_SET:{
if (s->speed_tempo) {
@@ -554,7 +554,7 @@ static int af_open(af_instance_t* af){
af->uninit = uninit;
af->play = play;
af->mul = 1;
- af->data = calloc(1,sizeof(af_data_t));
+ af->data = calloc(1,sizeof(struct mp_audio));
af->setup = calloc(1,sizeof(af_scaletempo_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_sinesuppress.c b/libaf/af_sinesuppress.c
index 3a69a86585..bb30e19f22 100644
--- a/libaf/af_sinesuppress.c
+++ b/libaf/af_sinesuppress.c
@@ -41,8 +41,8 @@ typedef struct af_sinesuppress_s
double pos;
}af_sinesuppress_t;
-static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data);
-//static af_data_t* play_float(struct af_instance_s* af, af_data_t* data);
+static struct mp_audio* play_s16(struct af_instance_s* af, struct mp_audio* data);
+//static struct mp_audio* play_float(struct af_instance_s* af, struct mp_audio* data);
// Initialization and runtime control
static int control(struct af_instance_s* af, int cmd, void* arg)
@@ -54,10 +54,10 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
af->data->nch = 1;
#if 0
- if (((af_data_t*)arg)->format == AF_FORMAT_FLOAT_NE)
+ if (((struct mp_audio*)arg)->format == AF_FORMAT_FLOAT_NE)
{
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
@@ -70,7 +70,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
af->play = play_s16;
}
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
float f1,f2;
@@ -103,7 +103,7 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_s16(struct af_instance_s* af, struct mp_audio* data)
{
af_sinesuppress_t *s = af->setup;
register int i = 0;
@@ -134,7 +134,7 @@ static af_data_t* play_s16(struct af_instance_s* af, af_data_t* data)
}
#if 0
-static af_data_t* play_float(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play_float(struct af_instance_s* af, struct mp_audio* data)
{
af_sinesuppress_t *s = af->setup;
register int i = 0;
@@ -163,7 +163,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play_s16;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_sinesuppress_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_sub.c b/libaf/af_sub.c
index 4330515ddf..2be755984b 100644
--- a/libaf/af_sub.c
+++ b/libaf/af_sub.c
@@ -69,8 +69,8 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = max(s->ch+1,((af_data_t*)arg)->nch);
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = max(s->ch+1,((struct mp_audio*)arg)->nch);
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
@@ -81,7 +81,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
(-1 == af_filter_szxform(sp[1].a, sp[1].b, Q, s->fc,
(float)af->data->rate, &s->k, s->w[1])))
return AF_ERROR;
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
int ch=5;
@@ -139,9 +139,9 @@ static void uninit(struct af_instance_s* af)
#endif
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
af_sub_t* s = af->setup; // Setup for this instance
float* a = c->audio; // Audio data
int len = c->len/4; // Number of samples in current audio block
@@ -167,7 +167,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=s=calloc(1,sizeof(af_sub_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_surround.c b/libaf/af_surround.c
index 28f69a586e..012c1da9f9 100644
--- a/libaf/af_surround.c
+++ b/libaf/af_surround.c
@@ -92,8 +92,8 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
switch(cmd){
case AF_CONTROL_REINIT:{
float fc;
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch*2;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch*2;
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
@@ -123,10 +123,10 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// printf("%i\n",s->wi);
s->ri = 0;
- if((af->data->format != ((af_data_t*)arg)->format) ||
- (af->data->bps != ((af_data_t*)arg)->bps)){
- ((af_data_t*)arg)->format = af->data->format;
- ((af_data_t*)arg)->bps = af->data->bps;
+ if((af->data->format != ((struct mp_audio*)arg)->format) ||
+ (af->data->bps != ((struct mp_audio*)arg)->bps)){
+ ((struct mp_audio*)arg)->format = af->data->format;
+ ((struct mp_audio*)arg)->bps = af->data->bps;
return AF_FALSE;
}
return AF_OK;
@@ -167,7 +167,7 @@ static float steering_matrix[][12] = {
//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data){
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data){
af_surround_t* s = (af_surround_t*)af->setup;
float* m = steering_matrix[0];
float* in = data->audio; // Input audio data
@@ -254,7 +254,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=2;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_surround_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_sweep.c b/libaf/af_sweep.c
index 3280125be1..f61e846619 100644
--- a/libaf/af_sweep.c
+++ b/libaf/af_sweep.c
@@ -37,7 +37,7 @@ typedef struct af_sweep_s{
static int control(struct af_instance_s* af, int cmd, void* arg)
{
af_sweept* s = (af_sweept*)af->setup;
- af_data_t *data= (af_data_t*)arg;
+ struct mp_audio *data= (struct mp_audio*)arg;
switch(cmd){
case AF_CONTROL_REINIT:
@@ -65,7 +65,7 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
af_sweept *s = af->setup;
int i, j;
@@ -88,7 +88,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_sweept));
return AF_OK;
}
diff --git a/libaf/af_tools.c b/libaf/af_tools.c
index 82e0940fb2..8652474963 100644
--- a/libaf/af_tools.c
+++ b/libaf/af_tools.c
@@ -85,13 +85,13 @@ int af_to_ms(int n, int* in, float* out, int rate)
}
/* Helper function for testing the output format */
-int af_test_output(struct af_instance_s* af, af_data_t* out)
+int af_test_output(struct af_instance_s* af, struct mp_audio* out)
{
if((af->data->format != out->format) ||
(af->data->bps != out->bps) ||
(af->data->rate != out->rate) ||
(af->data->nch != out->nch)){
- memcpy(out,af->data,sizeof(af_data_t));
+ memcpy(out,af->data,sizeof(struct mp_audio));
return AF_FALSE;
}
return AF_OK;
diff --git a/libaf/af_volnorm.c b/libaf/af_volnorm.c
index f7698e784c..80a9d31471 100644
--- a/libaf/af_volnorm.c
+++ b/libaf/af_volnorm.c
@@ -87,17 +87,17 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
- if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){
+ if(((struct mp_audio*)arg)->format == (AF_FORMAT_S16_NE)){
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
}else{
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
}
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
case AF_CONTROL_COMMAND_LINE:{
int i = 0;
float target = DEFAULT_TARGET;
@@ -120,7 +120,7 @@ static void uninit(struct af_instance_s* af)
free(af->setup);
}
-static void method1_int16(af_volnorm_t *s, af_data_t *c)
+static void method1_int16(af_volnorm_t *s, struct mp_audio *c)
{
register int i = 0;
int16_t *data = (int16_t*)c->audio; // Audio data
@@ -162,7 +162,7 @@ static void method1_int16(af_volnorm_t *s, af_data_t *c)
s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
}
-static void method1_float(af_volnorm_t *s, af_data_t *c)
+static void method1_float(af_volnorm_t *s, struct mp_audio *c)
{
register int i = 0;
float *data = (float*)c->audio; // Audio data
@@ -199,7 +199,7 @@ static void method1_float(af_volnorm_t *s, af_data_t *c)
s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
}
-static void method2_int16(af_volnorm_t *s, af_data_t *c)
+static void method2_int16(af_volnorm_t *s, struct mp_audio *c)
{
register int i = 0;
int16_t *data = (int16_t*)c->audio; // Audio data
@@ -249,7 +249,7 @@ static void method2_int16(af_volnorm_t *s, af_data_t *c)
s->idx = (s->idx + 1) % NSAMPLES;
}
-static void method2_float(af_volnorm_t *s, af_data_t *c)
+static void method2_float(af_volnorm_t *s, struct mp_audio *c)
{
register int i = 0;
float *data = (float*)c->audio; // Audio data
@@ -296,7 +296,7 @@ static void method2_float(af_volnorm_t *s, af_data_t *c)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
af_volnorm_t *s = af->setup;
@@ -324,7 +324,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_volnorm_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/af_volume.c b/libaf/af_volume.c
index 4e6a3b40f6..8ce60b5eac 100644
--- a/libaf/af_volume.c
+++ b/libaf/af_volume.c
@@ -66,10 +66,10 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Sanity check
if(!arg) return AF_ERROR;
- af->data->rate = ((af_data_t*)arg)->rate;
- af->data->nch = ((af_data_t*)arg)->nch;
+ af->data->rate = ((struct mp_audio*)arg)->rate;
+ af->data->nch = ((struct mp_audio*)arg)->nch;
- if(s->fast && (((af_data_t*)arg)->format != (AF_FORMAT_FLOAT_NE))){
+ if(s->fast && (((struct mp_audio*)arg)->format != (AF_FORMAT_FLOAT_NE))){
af->data->format = AF_FORMAT_S16_NE;
af->data->bps = 2;
}
@@ -82,7 +82,7 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
af->data->format = AF_FORMAT_FLOAT_NE;
af->data->bps = 4;
}
- return af_test_output(af,(af_data_t*)arg);
+ return af_test_output(af,(struct mp_audio*)arg);
case AF_CONTROL_COMMAND_LINE:{
float v=0.0;
float vol[AF_NCH];
@@ -138,9 +138,9 @@ static void uninit(struct af_instance_s* af)
}
// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+static struct mp_audio* play(struct af_instance_s* af, struct mp_audio* data)
{
- af_data_t* c = data; // Current working data
+ struct mp_audio* c = data; // Current working data
af_volume_t* s = (af_volume_t*)af->setup; // Setup for this instance
register int nch = c->nch; // Number of channels
register int i = 0;
@@ -203,7 +203,7 @@ static int af_open(af_instance_t* af){
af->uninit=uninit;
af->play=play;
af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
+ af->data=calloc(1,sizeof(struct mp_audio));
af->setup=calloc(1,sizeof(af_volume_t));
if(af->data == NULL || af->setup == NULL)
return AF_ERROR;
diff --git a/libaf/control.h b/libaf/control.h
index b99d50bcb4..323b9a3924 100644
--- a/libaf/control.h
+++ b/libaf/control.h
@@ -89,7 +89,7 @@ typedef struct af_control_ext_s{
// MANDATORY CALLS
/* Reinitialize filter. The optional argument contains the new
- configuration in form of a af_data_t struct. If the filter does not
+ configuration in form of a struct mp_audio struct. If the filter does not
support the new format the struct should be changed and AF_FALSE
should be returned. If the incoming and outgoing data streams are
identical the filter can return AF_DETACH. This will remove the
diff --git a/libmpcodecs/dec_audio.c b/libmpcodecs/dec_audio.c
index ad0bf336bc..3a3cb51417 100644
--- a/libmpcodecs/dec_audio.c
+++ b/libmpcodecs/dec_audio.c
@@ -353,7 +353,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
}
// Filter
- af_data_t filter_input = {
+ struct mp_audio filter_input = {
.audio = sh->a_buffer,
.len = len,
.rate = sh->samplerate,
@@ -361,7 +361,7 @@ static int filter_n_bytes(sh_audio_t *sh, struct bstr *outbuf, int len)
.format = sh->sample_format
};
af_fix_parameters(&filter_input);
- af_data_t *filter_output = af_play(sh->afilter, &filter_input);
+ struct mp_audio *filter_output = af_play(sh->afilter, &filter_input);
if (!filter_output)
return -1;
set_min_out_buffer_size(outbuf, outbuf->len + filter_output->len);