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diff --git a/lib/rbcodec/dsp/dsp.c b/lib/rbcodec/dsp/dsp.c
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1/***************************************************************************
2 * __________ __ ___.
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
7 * \/ \/ \/ \/ \/
8 * $Id$
9 *
10 * Copyright (C) 2005 Miika Pekkarinen
11 *
12 * This program is free software; you can redistribute it and/or
13 * modify it under the terms of the GNU General Public License
14 * as published by the Free Software Foundation; either version 2
15 * of the License, or (at your option) any later version.
16 *
17 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
18 * KIND, either express or implied.
19 *
20 ****************************************************************************/
21#include "config.h"
22#include "system.h"
23#include <sound.h>
24#include "dsp.h"
25#include "dsp-util.h"
26#include "eq.h"
27#include "compressor.h"
28#include "kernel.h"
29#include "settings.h"
30#include "replaygain.h"
31#include "tdspeed.h"
32#include "core_alloc.h"
33#include "fixedpoint.h"
34#include "fracmul.h"
35
36/* Define LOGF_ENABLE to enable logf output in this file */
37/*#define LOGF_ENABLE*/
38#include "logf.h"
39
40/* 16-bit samples are scaled based on these constants. The shift should be
41 * no more than 15.
42 */
43#define WORD_SHIFT 12
44#define WORD_FRACBITS 27
45
46#define NATIVE_DEPTH 16
47#define SMALL_SAMPLE_BUF_COUNT 128 /* Per channel */
48#define DEFAULT_GAIN 0x01000000
49
50/* enums to index conversion properly with stereo mode and other settings */
51enum
52{
53 SAMPLE_INPUT_LE_NATIVE_I_STEREO = STEREO_INTERLEAVED,
54 SAMPLE_INPUT_LE_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED,
55 SAMPLE_INPUT_LE_NATIVE_MONO = STEREO_MONO,
56 SAMPLE_INPUT_GT_NATIVE_I_STEREO = STEREO_INTERLEAVED + STEREO_NUM_MODES,
57 SAMPLE_INPUT_GT_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED + STEREO_NUM_MODES,
58 SAMPLE_INPUT_GT_NATIVE_MONO = STEREO_MONO + STEREO_NUM_MODES,
59 SAMPLE_INPUT_GT_NATIVE_1ST_INDEX = STEREO_NUM_MODES
60};
61
62enum
63{
64 SAMPLE_OUTPUT_MONO = 0,
65 SAMPLE_OUTPUT_STEREO,
66 SAMPLE_OUTPUT_DITHERED_MONO,
67 SAMPLE_OUTPUT_DITHERED_STEREO
68};
69
70/* No asm...yet */
71struct dither_data
72{
73 long error[3]; /* 00h */
74 long random; /* 0ch */
75 /* 10h */
76};
77
78struct crossfeed_data
79{
80 int32_t gain; /* 00h - Direct path gain */
81 int32_t coefs[3]; /* 04h - Coefficients for the shelving filter */
82 int32_t history[4]; /* 10h - Format is x[n - 1], y[n - 1] for both channels */
83 int32_t delay[13][2]; /* 20h */
84 int32_t *index; /* 88h - Current pointer into the delay line */
85 /* 8ch */
86};
87
88/* Current setup is one lowshelf filters three peaking filters and one
89 * highshelf filter. Varying the number of shelving filters make no sense,
90 * but adding peaking filters is possible.
91 */
92struct eq_state
93{
94 char enabled[5]; /* 00h - Flags for active filters */
95 struct eqfilter filters[5]; /* 08h - packing is 4? */
96 /* 10ch */
97};
98
99/* Include header with defines which functions are implemented in assembly
100 code for the target */
101#include <dsp_asm.h>
102
103/* Typedefs keep things much neater in this case */
104typedef void (*sample_input_fn_type)(int count, const char *src[],
105 int32_t *dst[]);
106typedef int (*resample_fn_type)(int count, struct dsp_data *data,
107 const int32_t *src[], int32_t *dst[]);
108typedef void (*sample_output_fn_type)(int count, struct dsp_data *data,
109 const int32_t *src[], int16_t *dst);
110
111/* Single-DSP channel processing in place */
112typedef void (*channels_process_fn_type)(int count, int32_t *buf[]);
113/* DSP local channel processing in place */
114typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data,
115 int32_t *buf[]);
116
117/*
118 ***************************************************************************/
119
120struct dsp_config
121{
122 struct dsp_data data; /* Config members for use in external routines */
123 long codec_frequency; /* Sample rate of data coming from the codec */
124 long frequency; /* Effective sample rate after pitch shift (if any) */
125 int sample_depth;
126 int sample_bytes;
127 int stereo_mode;
128 int32_t tdspeed_percent; /* Speed% * PITCH_SPEED_PRECISION */
129#ifdef HAVE_PITCHSCREEN
130 bool tdspeed_active; /* Timestretch is in use */
131#endif
132#ifdef HAVE_SW_TONE_CONTROLS
133 /* Filter struct for software bass/treble controls */
134 struct eqfilter tone_filter;
135#endif
136 /* Functions that change depending upon settings - NULL if stage is
137 disabled */
138 sample_input_fn_type input_samples;
139 resample_fn_type resample;
140 sample_output_fn_type output_samples;
141 /* These will be NULL for the voice codec and is more economical that
142 way */
143 channels_process_dsp_fn_type apply_gain;
144 channels_process_fn_type apply_crossfeed;
145 channels_process_fn_type eq_process;
146 channels_process_fn_type channels_process;
147 channels_process_dsp_fn_type compressor_process;
148};
149
150/* General DSP config */
151static struct dsp_config dsp_conf[2] IBSS_ATTR; /* 0=A, 1=V */
152/* Dithering */
153static struct dither_data dither_data[2] IBSS_ATTR; /* 0=left, 1=right */
154static long dither_mask IBSS_ATTR;
155static long dither_bias IBSS_ATTR;
156/* Crossfeed */
157struct crossfeed_data crossfeed_data IDATA_ATTR = /* A */
158{
159 .index = (int32_t *)crossfeed_data.delay
160};
161
162/* Equalizer */
163static struct eq_state eq_data; /* A */
164
165/* Software tone controls */
166#ifdef HAVE_SW_TONE_CONTROLS
167static int prescale; /* A/V */
168static int bass; /* A/V */
169static int treble; /* A/V */
170#endif
171
172/* Settings applicable to audio codec only */
173#ifdef HAVE_PITCHSCREEN
174static int32_t pitch_ratio = PITCH_SPEED_100;
175static int big_sample_locks;
176#endif
177static int channels_mode;
178 long dsp_sw_gain;
179 long dsp_sw_cross;
180static bool dither_enabled;
181static long eq_precut;
182static long track_gain;
183static bool new_gain;
184static long album_gain;
185static long track_peak;
186static long album_peak;
187static long replaygain;
188static bool crossfeed_enabled;
189
190#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
191#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])
192
193/* The internal format is 32-bit samples, non-interleaved, stereo. This
194 * format is similar to the raw output from several codecs, so the amount
195 * of copying needed is minimized for that case.
196 */
197
198#define RESAMPLE_RATIO 4 /* Enough for 11,025 Hz -> 44,100 Hz */
199#define SMALL_RESAMPLE_BUF_COUNT (SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO)
200#define BIG_SAMPLE_BUF_COUNT SMALL_RESAMPLE_BUF_COUNT
201#define BIG_RESAMPLE_BUF_COUNT (BIG_SAMPLE_BUF_COUNT * RESAMPLE_RATIO)
202
203static int32_t small_sample_buf[2][SMALL_SAMPLE_BUF_COUNT] IBSS_ATTR;
204static int32_t small_resample_buf[2][SMALL_RESAMPLE_BUF_COUNT] IBSS_ATTR;
205
206#ifdef HAVE_PITCHSCREEN
207static int32_t (* big_sample_buf)[BIG_SAMPLE_BUF_COUNT] = NULL;
208static int32_t (* big_resample_buf)[BIG_RESAMPLE_BUF_COUNT] = NULL;
209#endif
210
211static int sample_buf_count = SMALL_SAMPLE_BUF_COUNT;
212static int32_t *sample_buf[2] = { small_sample_buf[0], small_sample_buf[1] };
213static int resample_buf_count = SMALL_RESAMPLE_BUF_COUNT;
214static int32_t *resample_buf[2] = { small_resample_buf[0], small_resample_buf[1] };
215
216#ifdef HAVE_PITCHSCREEN
217int32_t sound_get_pitch(void)
218{
219 return pitch_ratio;
220}
221
222void sound_set_pitch(int32_t percent)
223{
224 pitch_ratio = percent;
225 dsp_configure(&AUDIO_DSP, DSP_SWITCH_FREQUENCY,
226 AUDIO_DSP.codec_frequency);
227}
228
229static void tdspeed_set_pointers( bool time_stretch_active )
230{
231 if( time_stretch_active )
232 {
233 sample_buf_count = BIG_SAMPLE_BUF_COUNT;
234 resample_buf_count = BIG_RESAMPLE_BUF_COUNT;
235 sample_buf[0] = big_sample_buf[0];
236 sample_buf[1] = big_sample_buf[1];
237 resample_buf[0] = big_resample_buf[0];
238 resample_buf[1] = big_resample_buf[1];
239 }
240 else
241 {
242 sample_buf_count = SMALL_SAMPLE_BUF_COUNT;
243 resample_buf_count = SMALL_RESAMPLE_BUF_COUNT;
244 sample_buf[0] = small_sample_buf[0];
245 sample_buf[1] = small_sample_buf[1];
246 resample_buf[0] = small_resample_buf[0];
247 resample_buf[1] = small_resample_buf[1];
248 }
249}
250
251static void tdspeed_setup(struct dsp_config *dspc)
252{
253 /* Assume timestretch will not be used */
254 dspc->tdspeed_active = false;
255
256 tdspeed_set_pointers( false );
257
258 if (!dsp_timestretch_available())
259 return; /* Timestretch not enabled or buffer not allocated */
260
261 if (dspc->tdspeed_percent == 0)
262 dspc->tdspeed_percent = PITCH_SPEED_100;
263
264 if (!tdspeed_config(
265 dspc->codec_frequency == 0 ? NATIVE_FREQUENCY : dspc->codec_frequency,
266 dspc->stereo_mode != STEREO_MONO,
267 dspc->tdspeed_percent))
268 return; /* Timestretch not possible or needed with these parameters */
269
270 /* Timestretch is to be used */
271 dspc->tdspeed_active = true;
272
273 tdspeed_set_pointers( true );
274}
275
276
277static int move_callback(int handle, void* current, void* new)
278{
279 (void)handle;(void)current;
280
281 if ( big_sample_locks > 0 )
282 return BUFLIB_CB_CANNOT_MOVE;
283
284 big_sample_buf = new;
285
286 /* no allocation without timestretch enabled */
287 tdspeed_set_pointers( true );
288 return BUFLIB_CB_OK;
289}
290
291static void lock_sample_buf( bool lock )
292{
293 if ( lock )
294 big_sample_locks++;
295 else
296 big_sample_locks--;
297}
298
299static struct buflib_callbacks ops = {
300 .move_callback = move_callback,
301 .shrink_callback = NULL,
302};
303
304
305void dsp_timestretch_enable(bool enabled)
306{
307 /* Hook to set up timestretch buffer on first call to settings_apply() */
308 static int handle = -1;
309 if (enabled)
310 {
311 if (big_sample_buf)
312 return; /* already allocated and enabled */
313
314 /* Set up timestretch buffers */
315 big_sample_buf = &small_resample_buf[0];
316 handle = core_alloc_ex("resample buf",
317 2 * BIG_RESAMPLE_BUF_COUNT * sizeof(int32_t),
318 &ops);
319 big_sample_locks = 0;
320 enabled = handle >= 0;
321
322 if (enabled)
323 {
324 /* success, now setup tdspeed */
325 big_resample_buf = core_get_data(handle);
326
327 tdspeed_init();
328 tdspeed_setup(&AUDIO_DSP);
329 }
330 }
331
332 if (!enabled)
333 {
334 dsp_set_timestretch(PITCH_SPEED_100);
335 tdspeed_finish();
336
337 if (handle >= 0)
338 core_free(handle);
339
340 handle = -1;
341 big_sample_buf = NULL;
342 }
343}
344
345void dsp_set_timestretch(int32_t percent)
346{
347 AUDIO_DSP.tdspeed_percent = percent;
348 tdspeed_setup(&AUDIO_DSP);
349}
350
351int32_t dsp_get_timestretch()
352{
353 return AUDIO_DSP.tdspeed_percent;
354}
355
356bool dsp_timestretch_available()
357{
358 return (global_settings.timestretch_enabled && big_sample_buf);
359}
360#endif /* HAVE_PITCHSCREEN */
361
362/* Convert count samples to the internal format, if needed. Updates src
363 * to point past the samples "consumed" and dst is set to point to the
364 * samples to consume. Note that for mono, dst[0] equals dst[1], as there
365 * is no point in processing the same data twice.
366 */
367
368/* convert count 16-bit mono to 32-bit mono */
369static void sample_input_lte_native_mono(
370 int count, const char *src[], int32_t *dst[])
371{
372 const int16_t *s = (int16_t *) src[0];
373 const int16_t * const send = s + count;
374 int32_t *d = dst[0] = dst[1] = sample_buf[0];
375 int scale = WORD_SHIFT;
376
377 while (s < send)
378 {
379 *d++ = *s++ << scale;
380 }
381
382 src[0] = (char *)s;
383}
384
385/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
386static void sample_input_lte_native_i_stereo(
387 int count, const char *src[], int32_t *dst[])
388{
389 const int32_t *s = (int32_t *) src[0];
390 const int32_t * const send = s + count;
391 int32_t *dl = dst[0] = sample_buf[0];
392 int32_t *dr = dst[1] = sample_buf[1];
393 int scale = WORD_SHIFT;
394
395 while (s < send)
396 {
397 int32_t slr = *s++;
398#ifdef ROCKBOX_LITTLE_ENDIAN
399 *dl++ = (slr >> 16) << scale;
400 *dr++ = (int32_t)(int16_t)slr << scale;
401#else /* ROCKBOX_BIG_ENDIAN */
402 *dl++ = (int32_t)(int16_t)slr << scale;
403 *dr++ = (slr >> 16) << scale;
404#endif
405 }
406
407 src[0] = (char *)s;
408}
409
410/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
411static void sample_input_lte_native_ni_stereo(
412 int count, const char *src[], int32_t *dst[])
413{
414 const int16_t *sl = (int16_t *) src[0];
415 const int16_t *sr = (int16_t *) src[1];
416 const int16_t * const slend = sl + count;
417 int32_t *dl = dst[0] = sample_buf[0];
418 int32_t *dr = dst[1] = sample_buf[1];
419 int scale = WORD_SHIFT;
420
421 while (sl < slend)
422 {
423 *dl++ = *sl++ << scale;
424 *dr++ = *sr++ << scale;
425 }
426
427 src[0] = (char *)sl;
428 src[1] = (char *)sr;
429}
430
431/* convert count 32-bit mono to 32-bit mono */
432static void sample_input_gt_native_mono(
433 int count, const char *src[], int32_t *dst[])
434{
435 dst[0] = dst[1] = (int32_t *)src[0];
436 src[0] = (char *)(dst[0] + count);
437}
438
439/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
440static void sample_input_gt_native_i_stereo(
441 int count, const char *src[], int32_t *dst[])
442{
443 const int32_t *s = (int32_t *)src[0];
444 const int32_t * const send = s + 2*count;
445 int32_t *dl = dst[0] = sample_buf[0];
446 int32_t *dr = dst[1] = sample_buf[1];
447
448 while (s < send)
449 {
450 *dl++ = *s++;
451 *dr++ = *s++;
452 }
453
454 src[0] = (char *)send;
455}
456
457/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
458static void sample_input_gt_native_ni_stereo(
459 int count, const char *src[], int32_t *dst[])
460{
461 dst[0] = (int32_t *)src[0];
462 dst[1] = (int32_t *)src[1];
463 src[0] = (char *)(dst[0] + count);
464 src[1] = (char *)(dst[1] + count);
465}
466
467/**
468 * sample_input_new_format()
469 *
470 * set the to-native sample conversion function based on dsp sample parameters
471 *
472 * !DSPPARAMSYNC
473 * needs syncing with changes to the following dsp parameters:
474 * * dsp->stereo_mode (A/V)
475 * * dsp->sample_depth (A/V)
476 */
477static void sample_input_new_format(struct dsp_config *dsp)
478{
479 static const sample_input_fn_type sample_input_functions[] =
480 {
481 [SAMPLE_INPUT_LE_NATIVE_I_STEREO] = sample_input_lte_native_i_stereo,
482 [SAMPLE_INPUT_LE_NATIVE_NI_STEREO] = sample_input_lte_native_ni_stereo,
483 [SAMPLE_INPUT_LE_NATIVE_MONO] = sample_input_lte_native_mono,
484 [SAMPLE_INPUT_GT_NATIVE_I_STEREO] = sample_input_gt_native_i_stereo,
485 [SAMPLE_INPUT_GT_NATIVE_NI_STEREO] = sample_input_gt_native_ni_stereo,
486 [SAMPLE_INPUT_GT_NATIVE_MONO] = sample_input_gt_native_mono,
487 };
488
489 int convert = dsp->stereo_mode;
490
491 if (dsp->sample_depth > NATIVE_DEPTH)
492 convert += SAMPLE_INPUT_GT_NATIVE_1ST_INDEX;
493
494 dsp->input_samples = sample_input_functions[convert];
495}
496
497
498#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
499/* write mono internal format to output format */
500static void sample_output_mono(int count, struct dsp_data *data,
501 const int32_t *src[], int16_t *dst)
502{
503 const int32_t *s0 = src[0];
504 const int scale = data->output_scale;
505 const int dc_bias = 1 << (scale - 1);
506
507 while (count-- > 0)
508 {
509 int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale);
510 *dst++ = lr;
511 *dst++ = lr;
512 }
513}
514#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO */
515
516/* write stereo internal format to output format */
517#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
518static void sample_output_stereo(int count, struct dsp_data *data,
519 const int32_t *src[], int16_t *dst)
520{
521 const int32_t *s0 = src[0];
522 const int32_t *s1 = src[1];
523 const int scale = data->output_scale;
524 const int dc_bias = 1 << (scale - 1);
525
526 while (count-- > 0)
527 {
528 *dst++ = clip_sample_16((*s0++ + dc_bias) >> scale);
529 *dst++ = clip_sample_16((*s1++ + dc_bias) >> scale);
530 }
531}
532#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO */
533
534/**
535 * The "dither" code to convert the 24-bit samples produced by libmad was
536 * taken from the coolplayer project - coolplayer.sourceforge.net
537 *
538 * This function handles mono and stereo outputs.
539 */
540static void sample_output_dithered(int count, struct dsp_data *data,
541 const int32_t *src[], int16_t *dst)
542{
543 const int32_t mask = dither_mask;
544 const int32_t bias = dither_bias;
545 const int scale = data->output_scale;
546 const int32_t min = data->clip_min;
547 const int32_t max = data->clip_max;
548 const int32_t range = max - min;
549 int ch;
550 int16_t *d;
551
552 for (ch = 0; ch < data->num_channels; ch++)
553 {
554 struct dither_data * const dither = &dither_data[ch];
555 const int32_t *s = src[ch];
556 int i;
557
558 for (i = 0, d = &dst[ch]; i < count; i++, s++, d += 2)
559 {
560 int32_t output, sample;
561 int32_t random;
562
563 /* Noise shape and bias (for correct rounding later) */
564 sample = *s;
565 sample += dither->error[0] - dither->error[1] + dither->error[2];
566 dither->error[2] = dither->error[1];
567 dither->error[1] = dither->error[0]/2;
568
569 output = sample + bias;
570
571 /* Dither, highpass triangle PDF */
572 random = dither->random*0x0019660dL + 0x3c6ef35fL;
573 output += (random & mask) - (dither->random & mask);
574 dither->random = random;
575
576 /* Round sample to output range */
577 output &= ~mask;
578
579 /* Error feedback */
580 dither->error[0] = sample - output;
581
582 /* Clip */
583 if ((uint32_t)(output - min) > (uint32_t)range)
584 {
585 int32_t c = min;
586 if (output > min)
587 c += range;
588 output = c;
589 }
590
591 /* Quantize and store */
592 *d = output >> scale;
593 }
594 }
595
596 if (data->num_channels == 2)
597 return;
598
599 /* Have to duplicate left samples into the right channel since
600 pcm buffer and hardware is interleaved stereo */
601 d = &dst[0];
602
603 while (count-- > 0)
604 {
605 int16_t s = *d++;
606 *d++ = s;
607 }
608}
609
610/**
611 * sample_output_new_format()
612 *
613 * set the from-native to ouput sample conversion routine
614 *
615 * !DSPPARAMSYNC
616 * needs syncing with changes to the following dsp parameters:
617 * * dsp->stereo_mode (A/V)
618 * * dither_enabled (A)
619 */
620static void sample_output_new_format(struct dsp_config *dsp)
621{
622 static const sample_output_fn_type sample_output_functions[] =
623 {
624 sample_output_mono,
625 sample_output_stereo,
626 sample_output_dithered,
627 sample_output_dithered
628 };
629
630 int out = dsp->data.num_channels - 1;
631
632 if (dsp == &AUDIO_DSP && dither_enabled)
633 out += 2;
634
635 dsp->output_samples = sample_output_functions[out];
636}
637
638/**
639 * Linear interpolation resampling that introduces a one sample delay because
640 * of our inability to look into the future at the end of a frame.
641 */
642#ifndef DSP_HAVE_ASM_RESAMPLING
643static int dsp_downsample(int count, struct dsp_data *data,
644 const int32_t *src[], int32_t *dst[])
645{
646 int ch = data->num_channels - 1;
647 uint32_t delta = data->resample_data.delta;
648 uint32_t phase, pos;
649 int32_t *d;
650
651 /* Rolled channel loop actually showed slightly faster. */
652 do
653 {
654 /* Just initialize things and not worry too much about the relatively
655 * uncommon case of not being able to spit out a sample for the frame.
656 */
657 const int32_t *s = src[ch];
658 int32_t last = data->resample_data.last_sample[ch];
659
660 data->resample_data.last_sample[ch] = s[count - 1];
661 d = dst[ch];
662 phase = data->resample_data.phase;
663 pos = phase >> 16;
664
665 /* Do we need last sample of previous frame for interpolation? */
666 if (pos > 0)
667 last = s[pos - 1];
668
669 while (pos < (uint32_t)count)
670 {
671 *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
672 phase += delta;
673 pos = phase >> 16;
674 last = s[pos - 1];
675 }
676 }
677 while (--ch >= 0);
678
679 /* Wrap phase accumulator back to start of next frame. */
680 data->resample_data.phase = phase - (count << 16);
681 return d - dst[0];
682}
683
684static int dsp_upsample(int count, struct dsp_data *data,
685 const int32_t *src[], int32_t *dst[])
686{
687 int ch = data->num_channels - 1;
688 uint32_t delta = data->resample_data.delta;
689 uint32_t phase, pos;
690 int32_t *d;
691
692 /* Rolled channel loop actually showed slightly faster. */
693 do
694 {
695 /* Should always be able to output a sample for a ratio up to RESAMPLE_RATIO */
696 const int32_t *s = src[ch];
697 int32_t last = data->resample_data.last_sample[ch];
698
699 data->resample_data.last_sample[ch] = s[count - 1];
700 d = dst[ch];
701 phase = data->resample_data.phase;
702 pos = phase >> 16;
703
704 while (pos == 0)
705 {
706 *d++ = last + FRACMUL((phase & 0xffff) << 15, s[0] - last);
707 phase += delta;
708 pos = phase >> 16;
709 }
710
711 while (pos < (uint32_t)count)
712 {
713 last = s[pos - 1];
714 *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
715 phase += delta;
716 pos = phase >> 16;
717 }
718 }
719 while (--ch >= 0);
720
721 /* Wrap phase accumulator back to start of next frame. */
722 data->resample_data.phase = phase & 0xffff;
723 return d - dst[0];
724}
725#endif /* DSP_HAVE_ASM_RESAMPLING */
726
727static void resampler_new_delta(struct dsp_config *dsp)
728{
729 dsp->data.resample_data.delta = (unsigned long)
730 dsp->frequency * 65536LL / NATIVE_FREQUENCY;
731
732 if (dsp->frequency == NATIVE_FREQUENCY)
733 {
734 /* NOTE: If fully glitch-free transistions from no resampling to
735 resampling are desired, last_sample history should be maintained
736 even when not resampling. */
737 dsp->resample = NULL;
738 dsp->data.resample_data.phase = 0;
739 dsp->data.resample_data.last_sample[0] = 0;
740 dsp->data.resample_data.last_sample[1] = 0;
741 }
742 else if (dsp->frequency < NATIVE_FREQUENCY)
743 dsp->resample = dsp_upsample;
744 else
745 dsp->resample = dsp_downsample;
746}
747
748/* Resample count stereo samples. Updates the src array, if resampling is
749 * done, to refer to the resampled data. Returns number of stereo samples
750 * for further processing.
751 */
752static inline int resample(struct dsp_config *dsp, int count, int32_t *src[])
753{
754 int32_t *dst[2] =
755 {
756 resample_buf[0],
757 resample_buf[1]
758 };
759 lock_sample_buf( true );
760 count = dsp->resample(count, &dsp->data, (const int32_t **)src, dst);
761
762 src[0] = dst[0];
763 src[1] = dst[dsp->data.num_channels - 1];
764 lock_sample_buf( false );
765 return count;
766}
767
768static void dither_init(struct dsp_config *dsp)
769{
770 memset(dither_data, 0, sizeof (dither_data));
771 dither_bias = (1L << (dsp->data.frac_bits - NATIVE_DEPTH));
772 dither_mask = (1L << (dsp->data.frac_bits + 1 - NATIVE_DEPTH)) - 1;
773}
774
775void dsp_dither_enable(bool enable)
776{
777 struct dsp_config *dsp = &AUDIO_DSP;
778 dither_enabled = enable;
779 sample_output_new_format(dsp);
780}
781
782/* Applies crossfeed to the stereo signal in src.
783 * Crossfeed is a process where listening over speakers is simulated. This
784 * is good for old hard panned stereo records, which might be quite fatiguing
785 * to listen to on headphones with no crossfeed.
786 */
787#ifndef DSP_HAVE_ASM_CROSSFEED
788static void apply_crossfeed(int count, int32_t *buf[])
789{
790 int32_t *hist_l = &crossfeed_data.history[0];
791 int32_t *hist_r = &crossfeed_data.history[2];
792 int32_t *delay = &crossfeed_data.delay[0][0];
793 int32_t *coefs = &crossfeed_data.coefs[0];
794 int32_t gain = crossfeed_data.gain;
795 int32_t *di = crossfeed_data.index;
796
797 int32_t acc;
798 int32_t left, right;
799 int i;
800
801 for (i = 0; i < count; i++)
802 {
803 left = buf[0][i];
804 right = buf[1][i];
805
806 /* Filter delayed sample from left speaker */
807 acc = FRACMUL(*di, coefs[0]);
808 acc += FRACMUL(hist_l[0], coefs[1]);
809 acc += FRACMUL(hist_l[1], coefs[2]);
810 /* Save filter history for left speaker */
811 hist_l[1] = acc;
812 hist_l[0] = *di;
813 *di++ = left;
814 /* Filter delayed sample from right speaker */
815 acc = FRACMUL(*di, coefs[0]);
816 acc += FRACMUL(hist_r[0], coefs[1]);
817 acc += FRACMUL(hist_r[1], coefs[2]);
818 /* Save filter history for right speaker */
819 hist_r[1] = acc;
820 hist_r[0] = *di;
821 *di++ = right;
822 /* Now add the attenuated direct sound and write to outputs */
823 buf[0][i] = FRACMUL(left, gain) + hist_r[1];
824 buf[1][i] = FRACMUL(right, gain) + hist_l[1];
825
826 /* Wrap delay line index if bigger than delay line size */
827 if (di >= delay + 13*2)
828 di = delay;
829 }
830 /* Write back local copies of data we've modified */
831 crossfeed_data.index = di;
832}
833#endif /* DSP_HAVE_ASM_CROSSFEED */
834
835/**
836 * dsp_set_crossfeed(bool enable)
837 *
838 * !DSPPARAMSYNC
839 * needs syncing with changes to the following dsp parameters:
840 * * dsp->stereo_mode (A)
841 */
842void dsp_set_crossfeed(bool enable)
843{
844 crossfeed_enabled = enable;
845 AUDIO_DSP.apply_crossfeed = (enable && AUDIO_DSP.data.num_channels > 1)
846 ? apply_crossfeed : NULL;
847}
848
849void dsp_set_crossfeed_direct_gain(int gain)
850{
851 crossfeed_data.gain = get_replaygain_int(gain * 10) << 7;
852 /* If gain is negative, the calculation overflowed and we need to clamp */
853 if (crossfeed_data.gain < 0)
854 crossfeed_data.gain = 0x7fffffff;
855}
856
857/* Both gains should be below 0 dB */
858void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
859{
860 int32_t *c = crossfeed_data.coefs;
861 long scaler = get_replaygain_int(lf_gain * 10) << 7;
862
863 cutoff = 0xffffffff/NATIVE_FREQUENCY*cutoff;
864 hf_gain -= lf_gain;
865 /* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
866 * point instead of shelf midpoint. This is for compatibility with the old
867 * crossfeed shelf filter and should be removed if crossfeed settings are
868 * ever made incompatible for any other good reason.
869 */
870 cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
871 filter_shelf_coefs(cutoff, hf_gain, false, c);
872 /* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
873 * over 1 and can do this safely
874 */
875 c[0] = FRACMUL_SHL(c[0], scaler, 4);
876 c[1] = FRACMUL_SHL(c[1], scaler, 4);
877 c[2] <<= 4;
878}
879
880/* Apply a constant gain to the samples (e.g., for ReplayGain).
881 * Note that this must be called before the resampler.
882 */
883#ifndef DSP_HAVE_ASM_APPLY_GAIN
884static void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
885{
886 const int32_t gain = data->gain;
887 int ch;
888
889 for (ch = 0; ch < data->num_channels; ch++)
890 {
891 int32_t *d = buf[ch];
892 int i;
893
894 for (i = 0; i < count; i++)
895 d[i] = FRACMUL_SHL(d[i], gain, 8);
896 }
897}
898#endif /* DSP_HAVE_ASM_APPLY_GAIN */
899
900/* Combine all gains to a global gain. */
901static void set_gain(struct dsp_config *dsp)
902{
903 /* gains are in S7.24 format */
904 dsp->data.gain = DEFAULT_GAIN;
905
906 /* Replay gain not relevant to voice */
907 if (dsp == &AUDIO_DSP && replaygain)
908 {
909 dsp->data.gain = replaygain;
910 }
911
912 if (dsp->eq_process && eq_precut)
913 {
914 dsp->data.gain = fp_mul(dsp->data.gain, eq_precut, 24);
915 }
916
917#ifdef HAVE_SW_VOLUME_CONTROL
918 if (global_settings.volume < SW_VOLUME_MAX ||
919 global_settings.volume > SW_VOLUME_MIN)
920 {
921 int vol_gain = get_replaygain_int(global_settings.volume * 100);
922 dsp->data.gain = (long) (((int64_t) dsp->data.gain * vol_gain) >> 24);
923 }
924#endif
925
926 if (dsp->data.gain == DEFAULT_GAIN)
927 {
928 dsp->data.gain = 0;
929 }
930 else
931 {
932 dsp->data.gain >>= 1; /* convert gain to S8.23 format */
933 }
934
935 dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
936}
937
938/**
939 * Update the amount to cut the audio before applying the equalizer.
940 *
941 * @param precut to apply in decibels (multiplied by 10)
942 */
943void dsp_set_eq_precut(int precut)
944{
945 eq_precut = get_replaygain_int(precut * -10);
946 set_gain(&AUDIO_DSP);
947}
948
949/**
950 * Synchronize the equalizer filter coefficients with the global settings.
951 *
952 * @param band the equalizer band to synchronize
953 */
954void dsp_set_eq_coefs(int band, int cutoff, int q, int gain)
955{
956 /* Convert user settings to format required by coef generator functions */
957 cutoff = 0xffffffff / NATIVE_FREQUENCY * cutoff;
958
959 if (q == 0)
960 q = 1;
961
962 /* NOTE: The coef functions assume the EMAC unit is in fractional mode,
963 which it should be, since we're executed from the main thread. */
964
965 /* Assume a band is disabled if the gain is zero */
966 if (gain == 0)
967 {
968 eq_data.enabled[band] = 0;
969 }
970 else
971 {
972 if (band == 0)
973 eq_ls_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
974 else if (band == 4)
975 eq_hs_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
976 else
977 eq_pk_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
978
979 eq_data.enabled[band] = 1;
980 }
981}
982
983/* Apply EQ filters to those bands that have got it switched on. */
984static void eq_process(int count, int32_t *buf[])
985{
986 static const int shifts[] =
987 {
988 EQ_SHELF_SHIFT, /* low shelf */
989 EQ_PEAK_SHIFT, /* peaking */
990 EQ_PEAK_SHIFT, /* peaking */
991 EQ_PEAK_SHIFT, /* peaking */
992 EQ_SHELF_SHIFT, /* high shelf */
993 };
994 unsigned int channels = AUDIO_DSP.data.num_channels;
995 int i;
996
997 /* filter configuration currently is 1 low shelf filter, 3 band peaking
998 filters and 1 high shelf filter, in that order. we need to know this
999 so we can choose the correct shift factor.
1000 */
1001 for (i = 0; i < 5; i++)
1002 {
1003 if (!eq_data.enabled[i])
1004 continue;
1005 eq_filter(buf, &eq_data.filters[i], count, channels, shifts[i]);
1006 }
1007}
1008
1009/**
1010 * Use to enable the equalizer.
1011 *
1012 * @param enable true to enable the equalizer
1013 */
1014void dsp_set_eq(bool enable)
1015{
1016 AUDIO_DSP.eq_process = enable ? eq_process : NULL;
1017 set_gain(&AUDIO_DSP);
1018}
1019
1020static void dsp_set_stereo_width(int value)
1021{
1022 long width, straight, cross;
1023
1024 width = value * 0x7fffff / 100;
1025
1026 if (value <= 100)
1027 {
1028 straight = (0x7fffff + width) / 2;
1029 cross = straight - width;
1030 }
1031 else
1032 {
1033 /* straight = (1 + width) / (2 * width) */
1034 straight = ((int64_t)(0x7fffff + width) << 22) / width;
1035 cross = straight - 0x7fffff;
1036 }
1037
1038 dsp_sw_gain = straight << 8;
1039 dsp_sw_cross = cross << 8;
1040}
1041
1042/**
1043 * Implements the different channel configurations and stereo width.
1044 */
1045
1046/* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for
1047 * completeness. */
1048#if 0
1049static void channels_process_sound_chan_stereo(int count, int32_t *buf[])
1050{
1051 /* The channels are each just themselves */
1052 (void)count; (void)buf;
1053}
1054#endif
1055
1056#ifndef DSP_HAVE_ASM_SOUND_CHAN_MONO
1057static void channels_process_sound_chan_mono(int count, int32_t *buf[])
1058{
1059 int32_t *sl = buf[0], *sr = buf[1];
1060
1061 while (count-- > 0)
1062 {
1063 int32_t lr = *sl/2 + *sr/2;
1064 *sl++ = lr;
1065 *sr++ = lr;
1066 }
1067}
1068#endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */
1069
1070#ifndef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
1071static void channels_process_sound_chan_custom(int count, int32_t *buf[])
1072{
1073 const int32_t gain = dsp_sw_gain;
1074 const int32_t cross = dsp_sw_cross;
1075 int32_t *sl = buf[0], *sr = buf[1];
1076
1077 while (count-- > 0)
1078 {
1079 int32_t l = *sl;
1080 int32_t r = *sr;
1081 *sl++ = FRACMUL(l, gain) + FRACMUL(r, cross);
1082 *sr++ = FRACMUL(r, gain) + FRACMUL(l, cross);
1083 }
1084}
1085#endif /* DSP_HAVE_ASM_SOUND_CHAN_CUSTOM */
1086
1087static void channels_process_sound_chan_mono_left(int count, int32_t *buf[])
1088{
1089 /* Just copy over the other channel */
1090 memcpy(buf[1], buf[0], count * sizeof (*buf));
1091}
1092
1093static void channels_process_sound_chan_mono_right(int count, int32_t *buf[])
1094{
1095 /* Just copy over the other channel */
1096 memcpy(buf[0], buf[1], count * sizeof (*buf));
1097}
1098
1099#ifndef DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
1100static void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
1101{
1102 int32_t *sl = buf[0], *sr = buf[1];
1103
1104 while (count-- > 0)
1105 {
1106 int32_t ch = *sl/2 - *sr/2;
1107 *sl++ = ch;
1108 *sr++ = -ch;
1109 }
1110}
1111#endif /* DSP_HAVE_ASM_SOUND_CHAN_KARAOKE */
1112
1113static void dsp_set_channel_config(int value)
1114{
1115 static const channels_process_fn_type channels_process_functions[] =
1116 {
1117 /* SOUND_CHAN_STEREO = All-purpose index for no channel processing */
1118 [SOUND_CHAN_STEREO] = NULL,
1119 [SOUND_CHAN_MONO] = channels_process_sound_chan_mono,
1120 [SOUND_CHAN_CUSTOM] = channels_process_sound_chan_custom,
1121 [SOUND_CHAN_MONO_LEFT] = channels_process_sound_chan_mono_left,
1122 [SOUND_CHAN_MONO_RIGHT] = channels_process_sound_chan_mono_right,
1123 [SOUND_CHAN_KARAOKE] = channels_process_sound_chan_karaoke,
1124 };
1125
1126 if ((unsigned)value >= ARRAYLEN(channels_process_functions) ||
1127 AUDIO_DSP.stereo_mode == STEREO_MONO)
1128 {
1129 value = SOUND_CHAN_STEREO;
1130 }
1131
1132 /* This doesn't apply to voice */
1133 channels_mode = value;
1134 AUDIO_DSP.channels_process = channels_process_functions[value];
1135}
1136
1137#if CONFIG_CODEC == SWCODEC
1138
1139#ifdef HAVE_SW_TONE_CONTROLS
1140static void set_tone_controls(void)
1141{
1142 filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
1143 0xffffffff/NATIVE_FREQUENCY*3500,
1144 bass, treble, -prescale,
1145 AUDIO_DSP.tone_filter.coefs);
1146 /* Sync the voice dsp coefficients */
1147 memcpy(&VOICE_DSP.tone_filter.coefs, AUDIO_DSP.tone_filter.coefs,
1148 sizeof (VOICE_DSP.tone_filter.coefs));
1149}
1150#endif
1151
1152/* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
1153 * code directly.
1154 */
1155int dsp_callback(int msg, intptr_t param)
1156{
1157 switch (msg)
1158 {
1159#ifdef HAVE_SW_TONE_CONTROLS
1160 case DSP_CALLBACK_SET_PRESCALE:
1161 prescale = param;
1162 set_tone_controls();
1163 break;
1164 /* prescaler is always set after calling any of these, so we wait with
1165 * calculating coefs until the above case is hit.
1166 */
1167 case DSP_CALLBACK_SET_BASS:
1168 bass = param;
1169 break;
1170 case DSP_CALLBACK_SET_TREBLE:
1171 treble = param;
1172 break;
1173#ifdef HAVE_SW_VOLUME_CONTROL
1174 case DSP_CALLBACK_SET_SW_VOLUME:
1175 set_gain(&AUDIO_DSP);
1176 break;
1177#endif
1178#endif
1179 case DSP_CALLBACK_SET_CHANNEL_CONFIG:
1180 dsp_set_channel_config(param);
1181 break;
1182 case DSP_CALLBACK_SET_STEREO_WIDTH:
1183 dsp_set_stereo_width(param);
1184 break;
1185 default:
1186 break;
1187 }
1188 return 0;
1189}
1190#endif
1191
1192/* Process and convert src audio to dst based on the DSP configuration,
1193 * reading count number of audio samples. dst is assumed to be large
1194 * enough; use dsp_output_count() to get the required number. src is an
1195 * array of pointers; for mono and interleaved stereo, it contains one
1196 * pointer to the start of the audio data and the other is ignored; for
1197 * non-interleaved stereo, it contains two pointers, one for each audio
1198 * channel. Returns number of bytes written to dst.
1199 */
1200int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
1201{
1202 static int32_t *tmp[2]; /* tdspeed_doit() needs it static */
1203 static long last_yield;
1204 long tick;
1205 int written = 0;
1206
1207#if defined(CPU_COLDFIRE)
1208 /* set emac unit for dsp processing, and save old macsr, we're running in
1209 codec thread context at this point, so can't clobber it */
1210 unsigned long old_macsr = coldfire_get_macsr();
1211 coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
1212#endif
1213
1214 if (new_gain)
1215 dsp_set_replaygain(); /* Gain has changed */
1216
1217 /* Perform at least one yield before starting */
1218 last_yield = current_tick;
1219 yield();
1220
1221 /* Testing function pointers for NULL is preferred since the pointer
1222 will be preloaded to be used for the call if not. */
1223 while (count > 0)
1224 {
1225 int samples = MIN(sample_buf_count, count);
1226 count -= samples;
1227
1228 dsp->input_samples(samples, src, tmp);
1229
1230#ifdef HAVE_PITCHSCREEN
1231 if (dsp->tdspeed_active)
1232 samples = tdspeed_doit(tmp, samples);
1233#endif
1234
1235 int chunk_offset = 0;
1236 while (samples > 0)
1237 {
1238 int32_t *t2[2];
1239 t2[0] = tmp[0]+chunk_offset;
1240 t2[1] = tmp[1]+chunk_offset;
1241
1242 int chunk = MIN(sample_buf_count, samples);
1243 chunk_offset += chunk;
1244 samples -= chunk;
1245
1246 if (dsp->apply_gain)
1247 dsp->apply_gain(chunk, &dsp->data, t2);
1248
1249 if (dsp->resample && (chunk = resample(dsp, chunk, t2)) <= 0)
1250 break; /* I'm pretty sure we're downsampling here */
1251
1252 if (dsp->apply_crossfeed)
1253 dsp->apply_crossfeed(chunk, t2);
1254
1255 if (dsp->eq_process)
1256 dsp->eq_process(chunk, t2);
1257
1258#ifdef HAVE_SW_TONE_CONTROLS
1259 if ((bass | treble) != 0)
1260 eq_filter(t2, &dsp->tone_filter, chunk,
1261 dsp->data.num_channels, FILTER_BISHELF_SHIFT);
1262#endif
1263
1264 if (dsp->channels_process)
1265 dsp->channels_process(chunk, t2);
1266
1267 if (dsp->compressor_process)
1268 dsp->compressor_process(chunk, &dsp->data, t2);
1269
1270 dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);
1271
1272 written += chunk;
1273 dst += chunk * sizeof (int16_t) * 2;
1274
1275 /* yield at least once each tick */
1276 tick = current_tick;
1277 if (TIME_AFTER(tick, last_yield))
1278 {
1279 last_yield = tick;
1280 yield();
1281 }
1282 }
1283 }
1284
1285#if defined(CPU_COLDFIRE)
1286 /* set old macsr again */
1287 coldfire_set_macsr(old_macsr);
1288#endif
1289 return written;
1290}
1291
1292/* Given count number of input samples, calculate the maximum number of
1293 * samples of output data that would be generated (the calculation is not
1294 * entirely exact and rounds upwards to be on the safe side; during
1295 * resampling, the number of samples generated depends on the current state
1296 * of the resampler).
1297 */
1298/* dsp_input_size MUST be called afterwards */
1299int dsp_output_count(struct dsp_config *dsp, int count)
1300{
1301#ifdef HAVE_PITCHSCREEN
1302 if (dsp->tdspeed_active)
1303 count = tdspeed_est_output_size();
1304#endif
1305 if (dsp->resample)
1306 {
1307 count = (int)(((unsigned long)count * NATIVE_FREQUENCY
1308 + (dsp->frequency - 1)) / dsp->frequency);
1309 }
1310
1311 /* Now we have the resampled sample count which must not exceed
1312 * resample_buf_count to avoid resample buffer overflow. One
1313 * must call dsp_input_count() to get the correct input sample
1314 * count.
1315 */
1316 if (count > resample_buf_count)
1317 count = resample_buf_count;
1318
1319 return count;
1320}
1321
1322/* Given count output samples, calculate number of input samples
1323 * that would be consumed in order to fill the output buffer.
1324 */
1325int dsp_input_count(struct dsp_config *dsp, int count)
1326{
1327 /* count is now the number of resampled input samples. Convert to
1328 original input samples. */
1329 if (dsp->resample)
1330 {
1331 /* Use the real resampling delta =
1332 * dsp->frequency * 65536 / NATIVE_FREQUENCY, and
1333 * round towards zero to avoid buffer overflows. */
1334 count = (int)(((unsigned long)count *
1335 dsp->data.resample_data.delta) >> 16);
1336 }
1337
1338#ifdef HAVE_PITCHSCREEN
1339 if (dsp->tdspeed_active)
1340 count = tdspeed_est_input_size(count);
1341#endif
1342
1343 return count;
1344}
1345
1346static void dsp_set_gain_var(long *var, long value)
1347{
1348 *var = value;
1349 new_gain = true;
1350}
1351
1352static void dsp_update_functions(struct dsp_config *dsp)
1353{
1354 sample_input_new_format(dsp);
1355 sample_output_new_format(dsp);
1356 if (dsp == &AUDIO_DSP)
1357 dsp_set_crossfeed(crossfeed_enabled);
1358}
1359
1360intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
1361{
1362 switch (setting)
1363 {
1364 case DSP_MYDSP:
1365 switch (value)
1366 {
1367 case CODEC_IDX_AUDIO:
1368 return (intptr_t)&AUDIO_DSP;
1369 case CODEC_IDX_VOICE:
1370 return (intptr_t)&VOICE_DSP;
1371 default:
1372 return (intptr_t)NULL;
1373 }
1374
1375 case DSP_SET_FREQUENCY:
1376 memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data));
1377 /* Fall through!!! */
1378 case DSP_SWITCH_FREQUENCY:
1379 dsp->codec_frequency = (value == 0) ? NATIVE_FREQUENCY : value;
1380 /* Account for playback speed adjustment when setting dsp->frequency
1381 if we're called from the main audio thread. Voice UI thread should
1382 not need this feature.
1383 */
1384#ifdef HAVE_PITCHSCREEN
1385 if (dsp == &AUDIO_DSP)
1386 dsp->frequency = pitch_ratio * dsp->codec_frequency / PITCH_SPEED_100;
1387 else
1388#endif
1389 dsp->frequency = dsp->codec_frequency;
1390
1391 resampler_new_delta(dsp);
1392#ifdef HAVE_PITCHSCREEN
1393 tdspeed_setup(dsp);
1394#endif
1395 break;
1396
1397 case DSP_SET_SAMPLE_DEPTH:
1398 dsp->sample_depth = value;
1399
1400 if (dsp->sample_depth <= NATIVE_DEPTH)
1401 {
1402 dsp->data.frac_bits = WORD_FRACBITS;
1403 dsp->sample_bytes = sizeof (int16_t); /* samples are 16 bits */
1404 dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
1405 dsp->data.clip_min = -((1 << WORD_FRACBITS));
1406 }
1407 else
1408 {
1409 dsp->data.frac_bits = value;
1410 dsp->sample_bytes = sizeof (int32_t); /* samples are 32 bits */
1411 dsp->data.clip_max = (1 << value) - 1;
1412 dsp->data.clip_min = -(1 << value);
1413 }
1414
1415 dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH;
1416 sample_input_new_format(dsp);
1417 dither_init(dsp);
1418 break;
1419
1420 case DSP_SET_STEREO_MODE:
1421 dsp->stereo_mode = value;
1422 dsp->data.num_channels = value == STEREO_MONO ? 1 : 2;
1423 dsp_update_functions(dsp);
1424#ifdef HAVE_PITCHSCREEN
1425 tdspeed_setup(dsp);
1426#endif
1427 break;
1428
1429 case DSP_RESET:
1430 dsp->stereo_mode = STEREO_NONINTERLEAVED;
1431 dsp->data.num_channels = 2;
1432 dsp->sample_depth = NATIVE_DEPTH;
1433 dsp->data.frac_bits = WORD_FRACBITS;
1434 dsp->sample_bytes = sizeof (int16_t);
1435 dsp->data.output_scale = dsp->data.frac_bits + 1 - NATIVE_DEPTH;
1436 dsp->data.clip_max = ((1 << WORD_FRACBITS) - 1);
1437 dsp->data.clip_min = -((1 << WORD_FRACBITS));
1438 dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;
1439
1440 if (dsp == &AUDIO_DSP)
1441 {
1442 track_gain = 0;
1443 album_gain = 0;
1444 track_peak = 0;
1445 album_peak = 0;
1446 new_gain = true;
1447 }
1448
1449 dsp_update_functions(dsp);
1450 resampler_new_delta(dsp);
1451#ifdef HAVE_PITCHSCREEN
1452 tdspeed_setup(dsp);
1453#endif
1454 if (dsp == &AUDIO_DSP)
1455 compressor_reset();
1456 break;
1457
1458 case DSP_FLUSH:
1459 memset(&dsp->data.resample_data, 0,
1460 sizeof (dsp->data.resample_data));
1461 resampler_new_delta(dsp);
1462 dither_init(dsp);
1463#ifdef HAVE_PITCHSCREEN
1464 tdspeed_setup(dsp);
1465#endif
1466 if (dsp == &AUDIO_DSP)
1467 compressor_reset();
1468 break;
1469
1470 case DSP_SET_TRACK_GAIN:
1471 if (dsp == &AUDIO_DSP)
1472 dsp_set_gain_var(&track_gain, value);
1473 break;
1474
1475 case DSP_SET_ALBUM_GAIN:
1476 if (dsp == &AUDIO_DSP)
1477 dsp_set_gain_var(&album_gain, value);
1478 break;
1479
1480 case DSP_SET_TRACK_PEAK:
1481 if (dsp == &AUDIO_DSP)
1482 dsp_set_gain_var(&track_peak, value);
1483 break;
1484
1485 case DSP_SET_ALBUM_PEAK:
1486 if (dsp == &AUDIO_DSP)
1487 dsp_set_gain_var(&album_peak, value);
1488 break;
1489
1490 default:
1491 return 0;
1492 }
1493
1494 return 1;
1495}
1496
1497int get_replaygain_mode(bool have_track_gain, bool have_album_gain)
1498{
1499 int type;
1500
1501 bool track = ((global_settings.replaygain_type == REPLAYGAIN_TRACK)
1502 || ((global_settings.replaygain_type == REPLAYGAIN_SHUFFLE)
1503 && global_settings.playlist_shuffle));
1504
1505 type = (!track && have_album_gain) ? REPLAYGAIN_ALBUM
1506 : have_track_gain ? REPLAYGAIN_TRACK : -1;
1507
1508 return type;
1509}
1510
1511void dsp_set_replaygain(void)
1512{
1513 long gain = 0;
1514
1515 new_gain = false;
1516
1517 if ((global_settings.replaygain_type != REPLAYGAIN_OFF) ||
1518 global_settings.replaygain_noclip)
1519 {
1520 bool track_mode = get_replaygain_mode(track_gain != 0,
1521 album_gain != 0) == REPLAYGAIN_TRACK;
1522 long peak = (track_mode || !album_peak) ? track_peak : album_peak;
1523
1524 if (global_settings.replaygain_type != REPLAYGAIN_OFF)
1525 {
1526 gain = (track_mode || !album_gain) ? track_gain : album_gain;
1527
1528 if (global_settings.replaygain_preamp)
1529 {
1530 long preamp = get_replaygain_int(
1531 global_settings.replaygain_preamp * 10);
1532
1533 gain = (long) (((int64_t) gain * preamp) >> 24);
1534 }
1535 }
1536
1537 if (gain == 0)
1538 {
1539 /* So that noclip can work even with no gain information. */
1540 gain = DEFAULT_GAIN;
1541 }
1542
1543 if (global_settings.replaygain_noclip && (peak != 0)
1544 && ((((int64_t) gain * peak) >> 24) >= DEFAULT_GAIN))
1545 {
1546 gain = (((int64_t) DEFAULT_GAIN << 24) / peak);
1547 }
1548
1549 if (gain == DEFAULT_GAIN)
1550 {
1551 /* Nothing to do, disable processing. */
1552 gain = 0;
1553 }
1554 }
1555
1556 /* Store in S7.24 format to simplify calculations. */
1557 replaygain = gain;
1558 set_gain(&AUDIO_DSP);
1559}
1560
1561/** SET COMPRESSOR
1562 * Called by the menu system to configure the compressor process */
1563void dsp_set_compressor(const struct compressor_settings *settings)
1564{
1565 /* enable/disable the compressor */
1566 AUDIO_DSP.compressor_process = compressor_update(settings) ?
1567 compressor_process : NULL;
1568}
generated by cgit v1.2.3 (git 2.39.1) at 2024-10-06 07:27:48 +0000