mpv/libaf/af.h

353 lines
9.7 KiB
C

/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#ifndef MPLAYER_AF_H
#define MPLAYER_AF_H
#include <stdio.h>
#include "config.h"
#include "options.h"
#include "af_format.h"
#include "control.h"
#include "cpudetect.h"
#include "mp_msg.h"
struct af_instance_s;
// Number of channels
#ifndef AF_NCH
#define AF_NCH 8
#endif
// Audio data chunk
typedef struct af_data_s
{
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
#define AF_FLAGS_REENTRANT 0x00000000
#define AF_FLAGS_NOT_REENTRANT 0x00000001
/* Audio filter information not specific for current instance, but for
a specific filter */
typedef struct af_info_s
{
const char *info;
const char *name;
const char *author;
const char *comment;
const int flags;
int (*open)(struct af_instance_s* vf);
} af_info_t;
// Linked list of audio filters
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);
void* setup; // setup data for this specific instance and filter
af_data_t* 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
* corresponding output */
double mul; /* length multiplier: how much does this instance change
the length of the buffer. */
}af_instance_t;
// Initialization flags
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_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)
#endif
// Configuration switches
typedef struct af_cfg_s{
int force; // Initialization type
char** list; /* list of names of filters that are added to filter
list during first initialization of stream */
}af_cfg_t;
// Current audio stream
typedef struct af_stream
{
// The first and last filter in the list
af_instance_t* first;
af_instance_t* last;
// Storage for input and output data formats
af_data_t input;
af_data_t output;
// Configuration for this stream
af_cfg_t cfg;
struct MPOpts *opts;
}af_stream_t;
/*********************************************
// Return values
*/
#define AF_DETACH 2
#define AF_OK 1
#define AF_TRUE 1
#define AF_FALSE 0
#define AF_UNKNOWN -1
#define AF_ERROR -2
#define AF_FATAL -3
/*********************************************
// Export functions
*/
/**
* \defgroup af_chain Audio filter chain functions
* \{
* \param s filter chain
*/
/**
* \brief Initialize the stream "s".
* \return 0 on success, -1 on failure
*
* This function creates a new filter list if necessary, according
* to the values set in input and output. Input and output should contain
* the format of the current movie and the format of the preferred output
* respectively.
* Filters to convert to the preferred output format are inserted
* automatically, except when they are set to 0.
* The function is reentrant i.e. if called with an already initialized
* stream the stream will be reinitialized.
*/
int af_init(af_stream_t* s);
/**
* \brief Uninit and remove all filters from audio filter chain
*/
void af_uninit(af_stream_t* 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(af_stream_t* s, af_instance_t* af);
/**
* \brief This function adds the filter "name" to the stream s.
* \param name name of filter to add
* \return pointer to the new filter, NULL if insert failed
*
* The filter will be inserted somewhere nice in the
* list of filters (i.e. at the beginning unless the
* first filter is the format filter (why??).
*/
af_instance_t* af_add(af_stream_t* s, char* name);
/**
* \brief Uninit and remove the filter "af"
* \param af filter to remove
*/
void af_remove(af_stream_t* s, af_instance_t* af);
/**
* \brief find filter in chain by name
* \param name name of the filter to find
* \return first filter with right name or NULL if not found
*
* This function is used for finding already initialized filters
*/
af_instance_t* af_get(af_stream_t* s, char* name);
/**
* \brief filter data chunk through the filters in the list
* \param data data to play
* \return resulting data
* \ingroup af_chain
*/
af_data_t* af_play(af_stream_t* s, af_data_t* data);
/**
* \brief send control to all filters, starting with the last until
* one accepts the command with AF_OK.
* \param cmd filter control command
* \param arg argument for filter command
* \return the accepting filter or NULL if none was found
*/
af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg);
/**
* \brief calculate average ratio of filter output lenth to input length
* \return the ratio
*/
double af_calc_filter_multiplier(af_stream_t* s);
/**
* \brief Calculate the total delay caused by the filters
* \return delay in bytes of "missing" output
*/
double af_calc_delay(af_stream_t* s);
/** \} */ // end of af_chain group
// Helper functions and macros used inside the audio filters
/**
* \defgroup af_filter Audio filter helper functions
* \{
*/
/* 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);
/* 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);
/**
* \brief convert dB to gain value
* \param n number of values to convert
* \param in [in] values in dB, <= -200 will become 0 gain
* \param out [out] gain values
* \param k input values are divided by this
* \param mi minimum dB value, input will be clamped to this
* \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);
/**
* \brief convert gain value to dB
* \param n number of values to convert
* \param in [in] gain values, 0 wil become -200 dB
* \param out [out] values in dB
* \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);
/**
* \brief convert milliseconds to sample time
* \param n number of values to convert
* \param in [in] values in milliseconds
* \param out [out] sample time values
* \param rate sample rate
* \param mi minimum ms value, input will be clamped to this
* \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);
/**
* \brief convert sample time to milliseconds
* \param n number of values to convert
* \param in [in] sample time values
* \param out [out] values in milliseconds
* \param rate sample rate
* \return AF_ERROR on error, AF_OK otherwise
*/
int af_to_ms(int n, int* in, float* out, int rate);
/**
* \brief test if output format matches
* \param af audio filter
* \param out needed format, will be overwritten by available
* format if they do not match
* \return AF_FALSE if formats do not match, AF_OK if they match
*
* 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);
/**
* \brief soft clipping function using sin()
* \param a input value
* \return clipped value
*/
float af_softclip(float a);
/** \} */ // end of af_filter group, but more functions of this group below
/** Print a list of all available audio filters */
void af_help(void);
/**
* \brief fill the missing parameters in the af_data_t structure
* \param data structure to fill
* \ingroup af_filter
*
* Currently only sets bps based on format
*/
void af_fix_parameters(af_data_t *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
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)
/* Some other useful macro definitions*/
#ifndef min
#define min(a,b)(((a)>(b))?(b):(a))
#endif
#ifndef max
#define max(a,b)(((a)>(b))?(a):(b))
#endif
#ifndef clamp
#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
#endif
#ifndef sign
#define sign(a) (((a)>0)?(1):(-1))
#endif
#ifndef lrnd
#define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
#endif
#endif /* MPLAYER_AF_H */