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authorStefano Pigozzi <stefano.pigozzi@gmail.com>2012-11-01 13:37:20 +0100
committerStefano Pigozzi <stefano.pigozzi@gmail.com>2012-11-02 19:20:03 +0100
commit11173115bba7747870932e16cee8c5794494bd87 (patch)
tree8223a26ea4491294be73e5002f205a0c946ea188
parent593e433cc8ef99e3efc9ffabe352646abf25ff52 (diff)
downloadmpv-11173115bba7747870932e16cee8c5794494bd87.tar.bz2
mpv-11173115bba7747870932e16cee8c5794494bd87.tar.xz
uncrustify af.h
used `uncrustify -l C -c TOOLS/uncrustify.cfg --no-backup --replace libaf/af.h`
-rw-r--r--libaf/af.h153
1 files changed, 77 insertions, 76 deletions
diff --git a/libaf/af.h b/libaf/af.h
index 865d08aae6..edce49a978 100644
--- a/libaf/af.h
+++ b/libaf/af.h
@@ -38,87 +38,87 @@ struct af_instance;
// Audio data chunk
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
+ 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
};
// Flags used for defining the behavior of an audio filter
-#define AF_FLAGS_REENTRANT 0x00000000
-#define AF_FLAGS_NOT_REENTRANT 0x00000001
+#define AF_FLAGS_REENTRANT 0x00000000
+#define AF_FLAGS_NOT_REENTRANT 0x00000001
/* Audio filter information not specific for current instance, but for
a specific filter */
struct af_info {
- const char *info;
- const char *name;
- const char *author;
- const char *comment;
- const int flags;
- int (*open)(struct af_instance* vf);
+ const char *info;
+ const char *name;
+ const char *author;
+ const char *comment;
+ const int flags;
+ int (*open)(struct af_instance *vf);
};
// Linked list of audio filters
struct af_instance {
- struct af_info* info;
- int (*control)(struct af_instance* af, int cmd, void* arg);
- void (*uninit)(struct af_instance* af);
- struct mp_audio* (*play)(struct af_instance* af, struct mp_audio* data);
- void* setup; // setup data for this specific instance and filter
- struct mp_audio* data; // configuration for outgoing data stream
- struct af_instance* next;
- struct af_instance* prev;
- double delay; /* Delay caused by the filter, in units of bytes read without
- * corresponding output */
- double mul; /* length multiplier: how much does this instance change
- the length of the buffer. */
+ struct af_info *info;
+ int (*control)(struct af_instance *af, int cmd, void *arg);
+ void (*uninit)(struct af_instance *af);
+ struct mp_audio * (*play)(struct af_instance *af, struct mp_audio *data);
+ void *setup; // setup data for this specific instance and filter
+ struct mp_audio *data; // configuration for outgoing data stream
+ struct af_instance *next;
+ struct af_instance *prev;
+ double delay; /* Delay caused by the filter, in units of bytes read without
+ * corresponding output */
+ double mul; /* length multiplier: how much does this instance change
+ the length of the buffer. */
};
// Initialization flags
-extern int* af_cpu_speed;
+extern int *af_cpu_speed;
-#define AF_INIT_AUTO 0x00000000
-#define AF_INIT_SLOW 0x00000001
-#define AF_INIT_FAST 0x00000002
-#define AF_INIT_FORCE 0x00000003
-#define AF_INIT_TYPE_MASK 0x00000003
+#define AF_INIT_AUTO 0x00000000
+#define AF_INIT_SLOW 0x00000001
+#define AF_INIT_FAST 0x00000002
+#define AF_INIT_FORCE 0x00000003
+#define AF_INIT_TYPE_MASK 0x00000003
-#define AF_INIT_INT 0x00000000
-#define AF_INIT_FLOAT 0x00000004
-#define AF_INIT_FORMAT_MASK 0x00000004
+#define AF_INIT_INT 0x00000000
+#define AF_INIT_FLOAT 0x00000004
+#define AF_INIT_FORMAT_MASK 0x00000004
// Default init type
#ifndef AF_INIT_TYPE
-#define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_SLOW)
+#define AF_INIT_TYPE (af_cpu_speed ? *af_cpu_speed : AF_INIT_SLOW)
#endif
// Configuration switches
struct af_cfg {
- int force; // Initialization type
- char** list; /* list of names of filters that are added to filter
- list during first initialization of stream */
+ int force; // Initialization type
+ char **list; /* list of names of filters that are added to filter
+ list during first initialization of stream */
};
// Current audio stream
struct af_stream {
- // The first and last filter in the list
- struct af_instance* first;
- struct af_instance* last;
- // Storage for input and output data formats
- struct mp_audio input;
- struct mp_audio output;
- // Configuration for this stream
- struct af_cfg cfg;
- struct MPOpts *opts;
+ // The first and last filter in the list
+ struct af_instance *first;
+ struct af_instance *last;
+ // Storage for input and output data formats
+ struct mp_audio input;
+ struct mp_audio output;
+ // Configuration for this stream
+ struct af_cfg cfg;
+ struct MPOpts *opts;
};
/*********************************************
-// Return values
-*/
+ // Return values
+ */
#define AF_DETACH 2
#define AF_OK 1
@@ -131,8 +131,8 @@ struct af_stream {
/*********************************************
-// Export functions
-*/
+ // Export functions
+ */
/**
* \defgroup af_chain Audio filter chain functions
@@ -153,19 +153,19 @@ struct af_stream {
* The function is reentrant i.e. if called with an already initialized
* stream the stream will be reinitialized.
*/
-int af_init(struct af_stream* s);
+int af_init(struct af_stream *s);
/**
* \brief Uninit and remove all filters from audio filter chain
*/
-void af_uninit(struct af_stream* s);
+void af_uninit(struct af_stream *s);
/**
* \brief Reinit the filter list from the given filter on downwards
* \param Filter instance to begin the reinit from
* \return AF_OK on success or AF_ERROR on failure
*/
-int af_reinit(struct af_stream* s, struct af_instance* af);
+int af_reinit(struct af_stream *s, struct af_instance *af);
/**
* \brief This function adds the filter "name" to the stream s.
@@ -176,13 +176,13 @@ int af_reinit(struct af_stream* s, struct af_instance* af);
* list of filters (i.e. at the beginning unless the
* first filter is the format filter (why??).
*/
-struct af_instance* af_add(struct af_stream* s, char* name);
+struct af_instance *af_add(struct af_stream *s, char *name);
/**
* \brief Uninit and remove the filter "af"
* \param af filter to remove
*/
-void af_remove(struct af_stream* s, struct af_instance* af);
+void af_remove(struct af_stream *s, struct af_instance *af);
/**
* \brief find filter in chain by name
@@ -191,7 +191,7 @@ void af_remove(struct af_stream* s, struct af_instance* af);
*
* This function is used for finding already initialized filters
*/
-struct af_instance* af_get(struct af_stream* s, char* name);
+struct af_instance *af_get(struct af_stream *s, char *name);
/**
* \brief filter data chunk through the filters in the list
@@ -199,7 +199,7 @@ struct af_instance* af_get(struct af_stream* s, char* name);
* \return resulting data
* \ingroup af_chain
*/
-struct mp_audio* af_play(struct af_stream* s, struct mp_audio* data);
+struct mp_audio *af_play(struct af_stream *s, struct mp_audio *data);
/**
* \brief send control to all filters, starting with the last until
@@ -208,19 +208,19 @@ struct mp_audio* af_play(struct af_stream* s, struct mp_audio* data);
* \param arg argument for filter command
* \return the accepting filter or NULL if none was found
*/
-struct af_instance *af_control_any_rev (struct af_stream* s, int cmd, void* arg);
+struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg);
/**
* \brief calculate average ratio of filter output lenth to input length
* \return the ratio
*/
-double af_calc_filter_multiplier(struct af_stream* s);
+double af_calc_filter_multiplier(struct af_stream *s);
/**
* \brief Calculate the total delay caused by the filters
* \return delay in bytes of "missing" output
*/
-double af_calc_delay(struct af_stream* s);
+double af_calc_delay(struct af_stream *s);
/** \} */ // end of af_chain group
@@ -233,12 +233,12 @@ double af_calc_delay(struct af_stream* s);
/* Helper function called by the macro with the same name only to be
called from inside filters */
-int af_resize_local_buffer(struct af_instance* af, struct mp_audio* data);
+int af_resize_local_buffer(struct af_instance *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, struct mp_audio* data);
+int af_lencalc(double mul, struct mp_audio *data);
/**
* \brief convert dB to gain value
@@ -250,7 +250,7 @@ int af_lencalc(double mul, struct mp_audio* data);
* \param ma maximum dB value, input will be clamped to this
* \return AF_ERROR on error, AF_OK otherwise
*/
-int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
+int af_from_dB(int n, float *in, float *out, float k, float mi, float ma);
/**
* \brief convert gain value to dB
@@ -260,7 +260,7 @@ int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
* \param k output values will be multiplied by this
* \return AF_ERROR on error, AF_OK otherwise
*/
-int af_to_dB(int n, float* in, float* out, float k);
+int af_to_dB(int n, float *in, float *out, float k);
/**
* \brief convert milliseconds to sample time
@@ -272,7 +272,7 @@ int af_to_dB(int n, float* in, float* out, float k);
* \param ma maximum ms value, input will be clamped to this
* \return AF_ERROR on error, AF_OK otherwise
*/
-int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
+int af_from_ms(int n, float *in, int *out, int rate, float mi, float ma);
/**
* \brief convert sample time to milliseconds
@@ -282,7 +282,7 @@ int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
* \param rate sample rate
* \return AF_ERROR on error, AF_OK otherwise
*/
-int af_to_ms(int n, int* in, float* out, int rate);
+int af_to_ms(int n, int *in, float *out, int rate);
/**
* \brief test if output format matches
@@ -293,7 +293,7 @@ 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* af, struct mp_audio* out);
+int af_test_output(struct af_instance *af, struct mp_audio *out);
/**
* \brief soft clipping function using sin()
@@ -321,28 +321,29 @@ void af_fix_parameters(struct mp_audio *data);
called to ensure the buffer is big enough.
* \ingroup af_filter
*/
-#define RESIZE_LOCAL_BUFFER(a,d)\
-((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK)
+#define RESIZE_LOCAL_BUFFER(a, d) \
+ ((a->data->len < \
+ af_lencalc(a->mul, d)) ? af_resize_local_buffer(a, d) : AF_OK)
/* Some other useful macro definitions*/
#ifndef min
-#define min(a,b)(((a)>(b))?(b):(a))
+#define min(a, b)(((a) > (b)) ? (b) : (a))
#endif
#ifndef max
-#define max(a,b)(((a)>(b))?(a):(b))
+#define max(a, b)(((a) > (b)) ? (a) : (b))
#endif
#ifndef clamp
-#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
+#define clamp(a, min, max) (((a) > (max)) ? (max) : (((a) < (min)) ? (min) : (a)))
#endif
#ifndef sign
-#define sign(a) (((a)>0)?(1):(-1))
+#define sign(a) (((a) > 0) ? (1) : (-1))
#endif
#ifndef lrnd
-#define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
+#define lrnd(a, b) ((b)((a) >= 0.0 ? (a) + 0.5 : (a) - 0.5))
#endif
#endif /* MPLAYER_AF_H */