mirror of
https://github.com/mpv-player/mpv
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1844e1dd0c
Rewrite decode_audio to better deal with filters that handle input in large blocks. It now always places output in sh_audio->a_out_buffer (which was always given as a parameter before) and reallocates the buffer if needed. After the changes filters can return arbitrarily large blocks of data without some of it being lost. The new version also allows simplifying some code. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24920 b3059339-0415-0410-9bf9-f77b7e298cf2
357 lines
9.7 KiB
C
357 lines
9.7 KiB
C
#ifndef __af_h__
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#define __af_h__
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#include <stdio.h>
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#include "af_mp.h"
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#include "config.h"
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#include "control.h"
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#include "af_format.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 6
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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 [ms]
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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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#if defined(HAVE_SSE) || defined(HAVE_3DNOW)
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#define AF_INIT_TYPE (af_cpu_speed?*af_cpu_speed:AF_INIT_FAST)
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#else
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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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#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_s
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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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}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 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 seconds
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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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/* Error messages */
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typedef struct af_msg_cfg_s
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{
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int level; /* Message level for debug and error messages max = 2
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min = -2 default = 0 */
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FILE* err; // Stream to print error messages to
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FILE* msg; // Stream to print information messages to
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}af_msg_cfg_t;
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extern af_msg_cfg_t af_msg_cfg; // Message
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//! \addtogroup af_filter
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//! \{
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#define AF_MSG_FATAL -3 ///< Fatal error exit immediately
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#define AF_MSG_ERROR -2 ///< Error return gracefully
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#define AF_MSG_WARN -1 ///< Print warning but do not exit (can be suppressed)
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#define AF_MSG_INFO 0 ///< Important information
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#define AF_MSG_VERBOSE 1 ///< Print this if verbose is enabled
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#define AF_MSG_DEBUG0 2 ///< Print if very verbose
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#define AF_MSG_DEBUG1 3 ///< Print if very very verbose
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//! Macro for printing error messages
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#ifndef af_msg
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#define af_msg(lev, args... ) \
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(((lev)<AF_MSG_WARN)?(fprintf(af_msg_cfg.err?af_msg_cfg.err:stderr, ## args )): \
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(((lev)<=af_msg_cfg.level)?(fprintf(af_msg_cfg.msg?af_msg_cfg.msg:stdout, ## args )):0))
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#endif
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//! \}
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#endif /* __af_h__ */
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