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author | wm4 <wm4@nowhere> | 2013-11-18 14:11:13 +0100 |
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committer | wm4 <wm4@nowhere> | 2013-11-18 14:21:01 +0100 |
commit | 93852b08f37f630b994d126751b4b9740a13219f (patch) | |
tree | 217fed11845a1c0855e6919a6fe3ec96ff34de4f /audio/filter/af.h | |
parent | 75dd3ec2106701cb865f52966de66c51cb6f9204 (diff) | |
download | mpv-93852b08f37f630b994d126751b4b9740a13219f.tar.bz2 mpv-93852b08f37f630b994d126751b4b9740a13219f.tar.xz |
af: cleanup documentation comments
And by "cleanup", I mean "remove". Actually, only remove the parts that
are redundant and doxygen noise. Move useful parts to the comment above
the function's implementation in the C source file.
Diffstat (limited to 'audio/filter/af.h')
-rw-r--r-- | audio/filter/af.h | 134 |
1 files changed, 2 insertions, 132 deletions
diff --git a/audio/filter/af.h b/audio/filter/af.h index 7852fa09a6..3a56c4c081 100644 --- a/audio/filter/af.h +++ b/audio/filter/af.h @@ -87,10 +87,7 @@ struct af_stream { struct MPOpts *opts; }; -/********************************************* - // Return values - */ - +// Return values #define AF_DETACH 2 #define AF_OK 1 #define AF_TRUE 1 @@ -99,150 +96,23 @@ struct af_stream { #define AF_ERROR -2 #define AF_FATAL -3 - - -/********************************************* - // Export functions - */ - -/** - * \defgroup af_chain Audio filter chain functions - * \{ - * \param s filter chain - */ - struct af_stream *af_new(struct MPOpts *opts); void af_destroy(struct af_stream *s); - -/** - * \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 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, char **args); - -/** - * \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 - * \{ - */ +int af_test_output(struct af_instance *af, struct mp_audio *out); -/** - * \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); #endif /* MPLAYER_AF_H */ |