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;inr;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;inr;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;idata->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;idata->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);