/* * 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 #include #include "config.h" #include "core/options.h" #include "audio/format.h" #include "control.h" #include "core/mp_msg.h" struct af_instance; // Number of channels #ifndef AF_NCH #define AF_NCH 8 #endif // Audio data chunk struct mp_audio { 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 }; // 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 */ struct af_info { const char *info; const char *name; const char *author; const char *comment; const int flags; int (*open)(struct af_instance *vf); }; // Linked list of audio filters struct af_instance { struct af_info *info; int (*control)(struct af_instance *af, int cmd, void *arg); void (*uninit)(struct af_instance *af); struct mp_audio * (*play)(struct af_instance *af, struct mp_audio *data); void *setup; // setup data for this specific instance and filter struct mp_audio *data; // configuration for outgoing data stream struct af_instance *next; struct af_instance *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. */ bool auto_inserted; // inserted by af.c, such as conversion filters }; // 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 struct af_cfg { int force; // Initialization type char **list; /* list of names of filters that are added to filter list during first initialization of stream */ }; // Current audio stream struct af_stream { // The first and last filter in the list struct af_instance *first; struct af_instance *last; // Storage for input and output data formats struct mp_audio input; struct mp_audio output; // Configuration for this stream struct af_cfg cfg; struct MPOpts *opts; }; /********************************************* // 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(struct af_stream *s); /** * \brief Uninit and remove all filters from audio filter chain */ void af_uninit(struct af_stream *s); /** * \brief Reinit the filter list from the given filter on downwards * See af.c. * \return AF_OK on success or AF_ERROR on failure */ int af_reinit(struct af_stream *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??). */ struct af_instance *af_add(struct af_stream *s, char *name); /** * \brief Uninit and remove the filter "af" * \param af filter to remove */ void af_remove(struct af_stream *s, struct af_instance *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 */ struct af_instance *af_get(struct af_stream *s, char *name); /** * \brief filter data chunk through the filters in the list * \param data data to play * \return resulting data * \ingroup af_chain */ struct mp_audio *af_play(struct af_stream *s, struct mp_audio *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 */ struct af_instance *af_control_any_rev(struct af_stream *s, int cmd, void *arg); /** * \brief calculate average ratio of filter output lenth to input length * \return the ratio */ double af_calc_filter_multiplier(struct af_stream *s); /** * \brief Calculate the total delay caused by the filters * \return delay in bytes of "missing" output */ double af_calc_delay(struct af_stream *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(struct af_instance *af, struct mp_audio *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, struct mp_audio *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 *af, struct mp_audio *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 struct mp_audio structure * \param data structure to fill * \ingroup af_filter * * Currently only sets bps based on format */ void af_fix_parameters(struct mp_audio *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 */