/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * $Id$ * * Copyright (C) 2005 Miika Pekkarinen * Copyright (C) 2012 Michael Sevakis * * 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. * ****************************************************************************/ #include "config.h" #include "system.h" #include "dsp.h" #include "dsp_sample_io.h" #if 1 #include #else #undef DEBUGF #define DEBUGF(...) #endif /* The internal format is 32-bit samples, non-interleaved, stereo. This * format is similar to the raw output from several codecs, so no copying is * needed for that case. * * Note that for mono, dst[0] equals dst[1], as there is no point in * processing the same data twice nor should it be done when modifying * samples in-place. * * When conversion is required: * Updates source buffer to point past the samples "consumed" also consuming * that portion of the input buffer and the destination is set to the buffer * of samples for later stages to consume. * * Input operates similarly to how an out-of-place processing stage should * behave. */ extern void dsp_sample_output_init(struct sample_io_data *this); extern void dsp_sample_output_flush(struct sample_io_data *this); /* convert count 16-bit mono to 32-bit mono */ static void sample_input_mono16(struct sample_io_data *this, struct dsp_buffer **buf_p) { struct dsp_buffer *src = *buf_p; struct dsp_buffer *dst = &this->sample_buf; *buf_p = dst; if (dst->remcount > 0) return; /* data still remains */ int count = MIN(src->remcount, SAMPLE_BUF_COUNT); dst->remcount = count; dst->p32[0] = this->sample_buf_arr[0]; dst->p32[1] = this->sample_buf_arr[0]; dst->proc_mask = src->proc_mask; if (count <= 0) return; /* purged sample_buf */ const int16_t *s = src->pin[0]; int32_t *d = dst->p32[0]; const int scale = WORD_SHIFT; dsp_advance_buffer_input(src, count, sizeof (int16_t)); do { *d++ = *s++ << scale; } while (--count > 0); } /* convert count 16-bit interleaved stereo to 32-bit noninterleaved */ static void sample_input_i_stereo16(struct sample_io_data *this, struct dsp_buffer **buf_p) { struct dsp_buffer *src = *buf_p; struct dsp_buffer *dst = &this->sample_buf; *buf_p = dst; if (dst->remcount > 0) return; /* data still remains */ int count = MIN(src->remcount, SAMPLE_BUF_COUNT); dst->remcount = count; dst->p32[0] = this->sample_buf_arr[0]; dst->p32[1] = this->sample_buf_arr[1]; dst->proc_mask = src->proc_mask; if (count <= 0) return; /* purged sample_buf */ const int16_t *s = src->pin[0]; int32_t *dl = dst->p32[0]; int32_t *dr = dst->p32[1]; const int scale = WORD_SHIFT; dsp_advance_buffer_input(src, count, 2*sizeof (int16_t)); do { *dl++ = *s++ << scale; *dr++ = *s++ << scale; } while (--count > 0); } /* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */ static void sample_input_ni_stereo16(struct sample_io_data *this, struct dsp_buffer **buf_p) { struct dsp_buffer *src = *buf_p; struct dsp_buffer *dst = &this->sample_buf; *buf_p = dst; if (dst->remcount > 0) return; /* data still remains */ int count = MIN(src->remcount, SAMPLE_BUF_COUNT); dst->remcount = count; dst->p32[0] = this->sample_buf_arr[0]; dst->p32[1] = this->sample_buf_arr[1]; dst->proc_mask = src->proc_mask; if (count <= 0) return; /* purged sample_buf */ const int16_t *sl = src->pin[0]; const int16_t *sr = src->pin[1]; int32_t *dl = dst->p32[0]; int32_t *dr = dst->p32[1]; const int scale = WORD_SHIFT; dsp_advance_buffer_input(src, count, sizeof (int16_t)); do { *dl++ = *sl++ << scale; *dr++ = *sr++ << scale; } while (--count > 0); } /* convert count 32-bit mono to 32-bit mono */ static void sample_input_mono32(struct sample_io_data *this, struct dsp_buffer **buf_p) { struct dsp_buffer *dst = &this->sample_buf; if (dst->remcount > 0) { *buf_p = dst; return; /* data still remains */ } /* else no buffer switch */ struct dsp_buffer *src = *buf_p; src->p32[1] = src->p32[0]; } /* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */ static void sample_input_i_stereo32(struct sample_io_data *this, struct dsp_buffer **buf_p) { struct dsp_buffer *src = *buf_p; struct dsp_buffer *dst = &this->sample_buf; *buf_p = dst; if (dst->remcount > 0) return; /* data still remains */ int count = MIN(src->remcount, SAMPLE_BUF_COUNT); dst->remcount = count; dst->p32[0] = this->sample_buf_arr[0]; dst->p32[1] = this->sample_buf_arr[1]; dst->proc_mask = src->proc_mask; if (count <= 0) return; /* purged sample_buf */ const int32_t *s = src->pin[0]; int32_t *dl = dst->p32[0]; int32_t *dr = dst->p32[1]; dsp_advance_buffer_input(src, count, 2*sizeof (int32_t)); do { *dl++ = *s++; *dr++ = *s++; } while (--count > 0); } /* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */ static void sample_input_ni_stereo32(struct sample_io_data *this, struct dsp_buffer **buf_p) { struct dsp_buffer *dst = &this->sample_buf; if (dst->remcount > 0) *buf_p = dst; /* data still remains */ /* else no buffer switch */ } /* set the to-native sample conversion function based on dsp sample * parameters */ static void dsp_sample_input_format_change(struct sample_io_data *this, struct dsp_buffer **buf_p) { static const sample_input_fn_type fns[STEREO_NUM_MODES][2] = { [STEREO_INTERLEAVED] = { sample_input_i_stereo16, sample_input_i_stereo32 }, [STEREO_NONINTERLEAVED] = { sample_input_ni_stereo16, sample_input_ni_stereo32 }, [STEREO_MONO] = { sample_input_mono16, sample_input_mono32 }, }; struct dsp_buffer *src = *buf_p; struct dsp_buffer *dst = &this->sample_buf; /* Ack configured format change */ format_change_ack(&this->format); if (dst->remcount > 0) { *buf_p = dst; return; /* data still remains */ } DSP_PRINT_FORMAT(DSP Input, -1, src->format); /* new format - remember it and pass it along */ dst->format = src->format; this->input_samples[0] = fns[this->stereo_mode] [this->sample_depth > NATIVE_DEPTH ? 1 : 0]; this->input_samples[0](this, buf_p); if (*buf_p == dst) /* buffer switch? */ format_change_ack(&src->format); } static void dsp_sample_input_init(struct sample_io_data *this) { this->input_samples[0] = sample_input_ni_stereo32; this->input_samples[1] = dsp_sample_input_format_change; } /* discard the sample buffer */ static void dsp_sample_input_flush(struct sample_io_data *this) { this->sample_buf.remcount = 0; } void dsp_sample_io_configure(struct sample_io_data *this, unsigned int setting, intptr_t value) { switch (setting) { case DSP_INIT: dsp_sample_input_init(this); dsp_sample_output_init(this); break; case DSP_RESET: /* Reset all sample descriptions to default */ format_change_set(&this->format); this->format.num_channels = 2; this->format.frac_bits = WORD_FRACBITS; this->format.output_scale = WORD_FRACBITS + 1 - NATIVE_DEPTH; this->format.frequency = NATIVE_FREQUENCY; this->format.codec_frequency = NATIVE_FREQUENCY; this->sample_depth = NATIVE_DEPTH; this->stereo_mode = STEREO_NONINTERLEAVED; break; case DSP_SET_FREQUENCY: value = value > 0 ? value : NATIVE_FREQUENCY; format_change_set(&this->format); this->format.frequency = value; this->format.codec_frequency = value; break; case DSP_SET_SAMPLE_DEPTH: format_change_set(&this->format); this->format.frac_bits = value <= NATIVE_DEPTH ? WORD_FRACBITS : value; this->format.output_scale = this->format.frac_bits + 1 - NATIVE_DEPTH; this->sample_depth = value; break; case DSP_SET_STEREO_MODE: format_change_set(&this->format); this->format.num_channels = value == STEREO_MONO ? 1 : 2; this->stereo_mode = value; break; case DSP_FLUSH: dsp_sample_input_flush(this); dsp_sample_output_flush(this); break; } }