/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2005 Miika Pekkarinen * * This program 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. * * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ #ifndef _DSP_H #define _DSP_H struct dsp_config; enum dsp_ids { CODEC_IDX_AUDIO, CODEC_IDX_VOICE, DSP_COUNT, }; enum dsp_settings { DSP_INIT, DSP_RESET, DSP_SET_FREQUENCY, DSP_SET_SAMPLE_DEPTH, DSP_SET_STEREO_MODE, DSP_FLUSH, DSP_SET_PITCH, DSP_SET_OUT_FREQUENCY, DSP_GET_OUT_FREQUENCY, DSP_PROC_INIT, DSP_PROC_CLOSE, DSP_PROC_NEW_FORMAT, DSP_PROC_SETTING, /* stage-specific should be this + id */ }; enum dsp_stereo_modes { STEREO_INTERLEAVED, STEREO_NONINTERLEAVED, STEREO_MONO, STEREO_NUM_MODES, }; /* Format into for the buffer */ struct sample_format { uint8_t version; /* 00h: format version number (never == 0, 0 is used to detect format calls for individual stages, such as when they are newly enabled) */ uint8_t num_channels; /* 01h: number of channels of data */ uint8_t frac_bits; /* 02h: number of fractional bits */ uint8_t output_scale; /* 03h: output scaling shift */ int32_t frequency; /* 04h: pitch-adjusted sample rate */ int32_t codec_frequency; /* 08h: codec-specifed sample rate */ /* 0ch */ }; /* Used by ASM routines - keep field order or else fix the functions */ struct dsp_buffer { int32_t remcount; /* 00h: Samples in buffer (In, Int, Out) */ union { const void *pin[2]; /* 04h: Channel pointers (In) */ int32_t *p32[2]; /* 04h: Channel pointers (Int) */ int16_t *p16out; /* 04h: DSP output buffer (Out) */ }; union { uint32_t proc_mask; /* 0Ch: In-place effects already appled to buffer in order to avoid double-processing. Set to zero on new buffer before passing to DSP. */ int bufcount; /* 0Ch: Buffer length/dest buffer remaining Basically, pay no attention unless it's *your* new buffer and is used internally or is specifically the final output buffer. */ }; struct sample_format format; /* 10h: Buffer format data */ /* 1ch */ }; /* Remove samples from input buffer (In). Sample size is specified. Provided to dsp_process(). */ static inline void dsp_advance_buffer_input(struct dsp_buffer *buf, int by_count, size_t size_each) { buf->remcount -= by_count; buf->pin[0] += by_count * size_each; buf->pin[1] += by_count * size_each; } /* Add samples to output buffer and update remaining space (Out). Provided to dsp_process() */ static inline void dsp_advance_buffer_output(struct dsp_buffer *buf, int by_count) { buf->bufcount -= by_count; buf->remcount += by_count; buf->p16out += 2 * by_count; /* Interleaved stereo */ } /* Remove samples from internal input buffer (In, Int). Provided to dsp_process() or by another processing stage. */ static inline void dsp_advance_buffer32(struct dsp_buffer *buf, int by_count) { buf->remcount -= by_count; buf->p32[0] += by_count; buf->p32[1] += by_count; } /* Get DSP pointer */ struct dsp_config *dsp_get_config(unsigned int dsp_id); /* Get DSP id */ unsigned int dsp_get_id(const struct dsp_config *dsp); /** General DSP processing **/ /* Process the given buffer - see implementation in dsp.c for more */ void dsp_process(struct dsp_config *dsp, struct dsp_buffer *src, struct dsp_buffer *dst); /* Change DSP settings */ intptr_t dsp_configure(struct dsp_config *dsp, unsigned int setting, intptr_t value); /* One-time startup init that must come before settings reset/apply */ void dsp_init(void) INIT_ATTR; #endif /* _DSP_H */