mirror of https://github.com/mpv-player/mpv
353 lines
9.7 KiB
C
353 lines
9.7 KiB
C
/*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#ifndef MPLAYER_AF_H
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#define MPLAYER_AF_H
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#include <stdio.h>
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#include "config.h"
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#include "options.h"
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#include "libaf/format.h"
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#include "control.h"
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#include "cpudetect.h"
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#include "mp_msg.h"
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struct af_instance_s;
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// Number of channels
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#ifndef AF_NCH
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#define AF_NCH 8
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#endif
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// Audio data chunk
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typedef struct af_data_s
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{
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void* audio; // data buffer
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int len; // buffer length
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int rate; // sample rate
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int nch; // number of channels
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int format; // format
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int bps; // bytes per sample
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} af_data_t;
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// Flags used for defining the behavior of an audio filter
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#define AF_FLAGS_REENTRANT 0x00000000
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#define AF_FLAGS_NOT_REENTRANT 0x00000001
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/* Audio filter information not specific for current instance, but for
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a specific filter */
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typedef struct af_info_s
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{
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const char *info;
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const char *name;
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const char *author;
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const char *comment;
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const int flags;
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int (*open)(struct af_instance_s* vf);
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} af_info_t;
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// Linked list of audio filters
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typedef struct af_instance_s
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{
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af_info_t* info;
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int (*control)(struct af_instance_s* af, int cmd, void* arg);
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void (*uninit)(struct af_instance_s* af);
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af_data_t* (*play)(struct af_instance_s* af, af_data_t* data);
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void* setup; // setup data for this specific instance and filter
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af_data_t* data; // configuration for outgoing data stream
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struct af_instance_s* next;
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struct af_instance_s* prev;
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double delay; /* Delay caused by the filter, in units of bytes read without
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* corresponding output */
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double mul; /* length multiplier: how much does this instance change
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the length of the buffer. */
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}af_instance_t;
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// Initialization flags
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extern int* af_cpu_speed;
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#define AF_INIT_AUTO 0x00000000
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#define AF_INIT_SLOW 0x00000001
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#define AF_INIT_FAST 0x00000002
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#define AF_INIT_FORCE 0x00000003
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#define AF_INIT_TYPE_MASK 0x00000003
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#define AF_INIT_INT 0x00000000
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#define AF_INIT_FLOAT 0x00000004
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#define AF_INIT_FORMAT_MASK 0x00000004
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// Default init type
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#ifndef AF_INIT_TYPE
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#define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_SLOW)
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#endif
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// Configuration switches
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typedef struct af_cfg_s{
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int force; // Initialization type
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char** list; /* list of names of filters that are added to filter
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list during first initialization of stream */
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}af_cfg_t;
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// Current audio stream
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typedef struct af_stream
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{
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// The first and last filter in the list
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af_instance_t* first;
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af_instance_t* last;
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// Storage for input and output data formats
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af_data_t input;
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af_data_t output;
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// Configuration for this stream
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af_cfg_t cfg;
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struct MPOpts *opts;
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}af_stream_t;
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/*********************************************
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// Return values
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*/
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#define AF_DETACH 2
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#define AF_OK 1
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#define AF_TRUE 1
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#define AF_FALSE 0
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#define AF_UNKNOWN -1
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#define AF_ERROR -2
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#define AF_FATAL -3
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/*********************************************
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// Export functions
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*/
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/**
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* \defgroup af_chain Audio filter chain functions
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* \{
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* \param s filter chain
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*/
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/**
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* \brief Initialize the stream "s".
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* \return 0 on success, -1 on failure
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*
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* This function creates a new filter list if necessary, according
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* to the values set in input and output. Input and output should contain
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* the format of the current movie and the format of the preferred output
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* respectively.
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* Filters to convert to the preferred output format are inserted
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* automatically, except when they are set to 0.
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* The function is reentrant i.e. if called with an already initialized
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* stream the stream will be reinitialized.
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*/
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int af_init(af_stream_t* s);
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/**
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* \brief Uninit and remove all filters from audio filter chain
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*/
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void af_uninit(af_stream_t* s);
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/**
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* \brief Reinit the filter list from the given filter on downwards
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* \param Filter instance to begin the reinit from
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* \return AF_OK on success or AF_ERROR on failure
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*/
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int af_reinit(af_stream_t* s, af_instance_t* af);
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/**
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* \brief This function adds the filter "name" to the stream s.
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* \param name name of filter to add
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* \return pointer to the new filter, NULL if insert failed
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*
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* The filter will be inserted somewhere nice in the
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* list of filters (i.e. at the beginning unless the
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* first filter is the format filter (why??).
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*/
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af_instance_t* af_add(af_stream_t* s, char* name);
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/**
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* \brief Uninit and remove the filter "af"
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* \param af filter to remove
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*/
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void af_remove(af_stream_t* s, af_instance_t* af);
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/**
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* \brief find filter in chain by name
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* \param name name of the filter to find
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* \return first filter with right name or NULL if not found
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*
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* This function is used for finding already initialized filters
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*/
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af_instance_t* af_get(af_stream_t* s, char* name);
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/**
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* \brief filter data chunk through the filters in the list
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* \param data data to play
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* \return resulting data
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* \ingroup af_chain
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*/
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af_data_t* af_play(af_stream_t* s, af_data_t* data);
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/**
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* \brief send control to all filters, starting with the last until
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* one accepts the command with AF_OK.
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* \param cmd filter control command
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* \param arg argument for filter command
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* \return the accepting filter or NULL if none was found
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*/
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af_instance_t *af_control_any_rev (af_stream_t* s, int cmd, void* arg);
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/**
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* \brief calculate average ratio of filter output lenth to input length
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* \return the ratio
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*/
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double af_calc_filter_multiplier(af_stream_t* s);
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/**
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* \brief Calculate the total delay caused by the filters
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* \return delay in bytes of "missing" output
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*/
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double af_calc_delay(af_stream_t* s);
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/** \} */ // end of af_chain group
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// Helper functions and macros used inside the audio filters
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/**
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* \defgroup af_filter Audio filter helper functions
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* \{
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*/
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/* Helper function called by the macro with the same name only to be
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called from inside filters */
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int af_resize_local_buffer(af_instance_t* af, af_data_t* data);
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/* Helper function used to calculate the exact buffer length needed
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when buffers are resized. The returned length is >= than what is
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needed */
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int af_lencalc(double mul, af_data_t* data);
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/**
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* \brief convert dB to gain value
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* \param n number of values to convert
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* \param in [in] values in dB, <= -200 will become 0 gain
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* \param out [out] gain values
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* \param k input values are divided by this
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* \param mi minimum dB value, input will be clamped to this
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* \param ma maximum dB value, input will be clamped to this
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* \return AF_ERROR on error, AF_OK otherwise
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*/
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int af_from_dB(int n, float* in, float* out, float k, float mi, float ma);
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/**
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* \brief convert gain value to dB
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* \param n number of values to convert
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* \param in [in] gain values, 0 wil become -200 dB
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* \param out [out] values in dB
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* \param k output values will be multiplied by this
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* \return AF_ERROR on error, AF_OK otherwise
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*/
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int af_to_dB(int n, float* in, float* out, float k);
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/**
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* \brief convert milliseconds to sample time
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* \param n number of values to convert
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* \param in [in] values in milliseconds
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* \param out [out] sample time values
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* \param rate sample rate
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* \param mi minimum ms value, input will be clamped to this
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* \param ma maximum ms value, input will be clamped to this
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* \return AF_ERROR on error, AF_OK otherwise
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*/
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int af_from_ms(int n, float* in, int* out, int rate, float mi, float ma);
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/**
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* \brief convert sample time to milliseconds
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* \param n number of values to convert
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* \param in [in] sample time values
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* \param out [out] values in milliseconds
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* \param rate sample rate
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* \return AF_ERROR on error, AF_OK otherwise
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*/
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int af_to_ms(int n, int* in, float* out, int rate);
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/**
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* \brief test if output format matches
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* \param af audio filter
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* \param out needed format, will be overwritten by available
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* format if they do not match
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* \return AF_FALSE if formats do not match, AF_OK if they match
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*
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* compares the format, bps, rate and nch values of af->data with out
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*/
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int af_test_output(struct af_instance_s* af, af_data_t* out);
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/**
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* \brief soft clipping function using sin()
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* \param a input value
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* \return clipped value
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*/
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float af_softclip(float a);
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/** \} */ // end of af_filter group, but more functions of this group below
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/** Print a list of all available audio filters */
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void af_help(void);
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/**
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* \brief fill the missing parameters in the af_data_t structure
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* \param data structure to fill
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* \ingroup af_filter
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*
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* Currently only sets bps based on format
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*/
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void af_fix_parameters(af_data_t *data);
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/** Memory reallocation macro: if a local buffer is used (i.e. if the
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filter doesn't operate on the incoming buffer this macro must be
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called to ensure the buffer is big enough.
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* \ingroup af_filter
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*/
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#define RESIZE_LOCAL_BUFFER(a,d)\
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((a->data->len < af_lencalc(a->mul,d))?af_resize_local_buffer(a,d):AF_OK)
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/* Some other useful macro definitions*/
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#ifndef min
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#define min(a,b)(((a)>(b))?(b):(a))
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#endif
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#ifndef max
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#define max(a,b)(((a)>(b))?(a):(b))
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#endif
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#ifndef clamp
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#define clamp(a,min,max) (((a)>(max))?(max):(((a)<(min))?(min):(a)))
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#endif
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#ifndef sign
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#define sign(a) (((a)>0)?(1):(-1))
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#endif
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#ifndef lrnd
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#define lrnd(a,b) ((b)((a)>=0.0?(a)+0.5:(a)-0.5))
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#endif
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#endif /* MPLAYER_AF_H */
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