Mercurial > mplayer.hg
view libaf/af.h @ 35041:f674ec17b65b
Fix usage count for passthrough-only filters.
author | reimar |
---|---|
date | Fri, 31 Aug 2012 20:09:30 +0000 |
parents | a93891202051 |
children | 2b9bc3c2933d |
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/* * 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" 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 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 */