annotate libaf/af.h @ 32334:541a774ce8e4

Drop removal of AAC encoder from list of FFmpeg encoders. The original reason to drop this particular encoder, that it failed to build on many platforms, should no longer be the case.

/*
 * 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 "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
{
  const 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;
}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 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 */