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author | Stefano Pigozzi <stefano.pigozzi@gmail.com> | 2012-11-01 12:00:00 +0100 |
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committer | Stefano Pigozzi <stefano.pigozzi@gmail.com> | 2012-11-02 19:19:28 +0100 |
commit | 0374ddb79d4e20aa73ef91496beb2c0811c34ca7 (patch) | |
tree | c9066550112a2b905c2e87351c10bd484ebe04ba | |
parent | e0aef8cf1275cc988007fcb2a099cdd9f92fa374 (diff) | |
download | mpv-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.c | 24 | ||||
-rw-r--r-- | libaf/af.h | 25 | ||||
-rw-r--r-- | libaf/af_bs2b.c | 12 | ||||
-rw-r--r-- | libaf/af_center.c | 12 | ||||
-rw-r--r-- | libaf/af_channels.c | 24 | ||||
-rw-r--r-- | libaf/af_delay.c | 14 | ||||
-rw-r--r-- | libaf/af_dummy.c | 6 | ||||
-rw-r--r-- | libaf/af_equalizer.c | 10 | ||||
-rw-r--r-- | libaf/af_export.c | 12 | ||||
-rw-r--r-- | libaf/af_extrastereo.c | 16 | ||||
-rw-r--r-- | libaf/af_format.c | 36 | ||||
-rw-r--r-- | libaf/af_hrtf.c | 10 | ||||
-rw-r--r-- | libaf/af_karaoke.c | 12 | ||||
-rw-r--r-- | libaf/af_ladspa.c | 10 | ||||
-rw-r--r-- | libaf/af_lavcac3enc.c | 10 | ||||
-rw-r--r-- | libaf/af_lavcresample.c | 8 | ||||
-rw-r--r-- | libaf/af_pan.c | 22 | ||||
-rw-r--r-- | libaf/af_resample.c | 14 | ||||
-rw-r--r-- | libaf/af_scaletempo.c | 12 | ||||
-rw-r--r-- | libaf/af_sinesuppress.c | 16 | ||||
-rw-r--r-- | libaf/af_sub.c | 12 | ||||
-rw-r--r-- | libaf/af_surround.c | 16 | ||||
-rw-r--r-- | libaf/af_sweep.c | 6 | ||||
-rw-r--r-- | libaf/af_tools.c | 4 | ||||
-rw-r--r-- | libaf/af_volnorm.c | 20 | ||||
-rw-r--r-- | libaf/af_volume.c | 14 | ||||
-rw-r--r-- | libaf/control.h | 2 | ||||
-rw-r--r-- | libmpcodecs/dec_audio.c | 4 |
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; |