1 /***************************************************************************
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
9 * Copyright (C) 2006-2008 Adam Gashlin (hcs)
10 * Copyright (C) 2006 Jens Arnold
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.
17 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
18 * KIND, either express or implied.
20 ****************************************************************************/
26 #include "lib/fixedpoint.h"
30 /* Maximum number of bytes to process in one iteration */
31 #define WAV_CHUNK_SIZE (1024*2)
33 /* Number of times to loop looped tracks when repeat is disabled */
36 /* Length of fade-out for looped tracks (milliseconds) */
37 #define FADE_LENGTH 10000L
39 /* Default high pass filter cutoff frequency is 500 Hz.
40 * Others can be set, but the default is nearly always used,
41 * and there is no way to determine if another was used, anyway.
43 const long cutoff
= 500;
45 static int16_t samples
[WAV_CHUNK_SIZE
] IBSS_ATTR
;
47 /* this is the codec entry point */
48 enum codec_status
codec_main(void)
51 int sampleswritten
, i
;
53 int32_t ch1_1
, ch1_2
, ch2_1
, ch2_2
; /* ADPCM history */
55 int endofstream
; /* end of stream flag */
56 uint32_t avgbytespersec
;
57 int looping
; /* looping flag */
58 int loop_count
; /* number of loops done so far */
59 int fade_count
; /* countdown for fadeout */
60 int fade_frames
; /* length of fade in frames */
61 off_t start_adr
, end_adr
; /* loop points */
62 off_t chanstart
, bufoff
;
63 /*long coef1=0x7298L,coef2=-0x3350L;*/
66 /* Generic codec initialisation */
67 /* we only render 16 bits */
68 ci
->configure(DSP_SET_SAMPLE_DEPTH
, 16);
71 DEBUGF("ADX: next_track\n");
75 DEBUGF("ADX: after init\n");
78 ch1_1
=ch1_2
=ch2_1
=ch2_2
=0;
80 /* wait for track info to load */
81 while (!*ci
->taginfo_ready
&& !ci
->stop_codec
)
84 codec_set_replaygain(ci
->id3
);
87 DEBUGF("ADX: request initial buffer\n");
89 buf
= ci
->request_buffer(&n
, 0x38);
90 if (!buf
|| n
< 0x38) {
94 DEBUGF("ADX: read size = %lx\n",(unsigned long)n
);
96 /* Get file header for starting offset, channel count */
98 chanstart
= ((buf
[2] << 8) | buf
[3]) + 4;
101 /* useful for seeking and reporting current playback position */
102 avgbytespersec
= ci
->id3
->frequency
* 18 * channels
/ 32;
103 DEBUGF("avgbytespersec=%ld\n",(unsigned long)avgbytespersec
);
105 /* calculate filter coefficients */
108 * A simple table of these coefficients would be nice, but
109 * some very odd frequencies are used and if I'm going to
110 * interpolate I might as well just go all the way and
111 * calclate them precisely.
112 * Speed is not an issue as this only needs to be done once per file.
115 const int64_t big28
= 0x10000000LL
;
116 const int64_t big32
= 0x100000000LL
;
117 int64_t frequency
= ci
->id3
->frequency
;
118 int64_t phasemultiple
= cutoff
*big32
/frequency
;
122 const int64_t b
= (M_SQRT2
*big28
)-big28
;
126 fp_sincos((unsigned long)phasemultiple
,&z
);
128 a
= (M_SQRT2
*big28
)-(z
*big28
/LONG_MAX
);
131 * In the long passed to fsqrt there are only 4 nonfractional bits,
132 * which is sufficient here, but this is the only reason why I don't
133 * use 32 fractional bits everywhere.
135 d
= fp_sqrt((a
+b
)*(a
-b
)/big28
,28);
138 coef1
= (c
*8192) >> 28;
139 coef2
= (c
*c
/big28
*-4096) >> 28;
140 DEBUGF("ADX: samprate=%ld ",(long)frequency
);
141 DEBUGF("coef1 %04x ",(unsigned int)(coef1
*4));
142 DEBUGF("coef2 %04x\n",(unsigned int)(coef2
*-4));
147 looping
= 0; start_adr
= 0; end_adr
= 0;
148 if (!memcmp(buf
+0x10,"\x01\xF4\x03\x00",4)) {
149 /* Soul Calibur 2 style (type 03) */
150 DEBUGF("ADX: type 03 found\n");
151 /* check if header is too small for loop data */
152 if (chanstart
-6 < 0x2c) looping
=0;
154 looping
= (buf
[0x18]) ||
158 end_adr
= (buf
[0x28]<<24) |
168 )/32*channels
*18+chanstart
;
170 } else if (!memcmp(buf
+0x10,"\x01\xF4\x04\x00",4)) {
171 /* Standard (type 04) */
172 DEBUGF("ADX: type 04 found\n");
173 /* check if header is too small for loop data */
174 if (chanstart
-6 < 0x38) looping
=0;
176 looping
= (buf
[0x24]) ||
180 end_adr
= (buf
[0x34]<<24) |
189 )/32*channels
*18+chanstart
;
192 DEBUGF("ADX: error, couldn't determine ADX type\n");
197 DEBUGF("ADX: looped, start: %lx end: %lx\n",start_adr
,end_adr
);
199 DEBUGF("ADX: not looped\n");
202 /* advance to first frame */
203 DEBUGF("ADX: first frame at %lx\n",chanstart
);
206 /* get in position */
207 ci
->seek_buffer(bufoff
);
210 /* setup pcm buffer format */
211 ci
->configure(DSP_SWITCH_FREQUENCY
, ci
->id3
->frequency
);
213 ci
->configure(DSP_SET_STEREO_MODE
, STEREO_INTERLEAVED
);
214 } else if (channels
== 1) {
215 ci
->configure(DSP_SET_STEREO_MODE
, STEREO_MONO
);
217 DEBUGF("ADX CODEC_ERROR: more than 2 channels\n");
223 fade_count
= -1; /* disable fade */
226 /* The main decoder loop */
228 while (!endofstream
) {
230 if (ci
->stop_codec
|| ci
->new_track
) {
234 /* do we need to loop? */
235 if (bufoff
> end_adr
-18*channels
&& looping
) {
236 DEBUGF("ADX: loop!\n");
237 /* check for endless looping */
238 if (ci
->global_settings
->repeat_mode
==REPEAT_ONE
) {
240 fade_count
= -1; /* disable fade */
242 /* otherwise start fade after LOOP_TIMES loops */
244 if (loop_count
>= LOOP_TIMES
&& fade_count
< 0) {
245 /* frames to fade over */
246 fade_frames
= FADE_LENGTH
*ci
->id3
->frequency
/32/1000;
247 /* volume relative to fade_frames */
248 fade_count
= fade_frames
;
249 DEBUGF("ADX: fade_frames = %d\n",fade_frames
);
253 ci
->seek_buffer(bufoff
);
256 /* do we need to seek? */
260 DEBUGF("ADX: seek to %ldms\n",ci
->seek_time
);
264 fade_count
= -1; /* disable fade */
267 newpos
= (((uint64_t)avgbytespersec
*(ci
->seek_time
- 1))
268 / (1000LL*18*channels
))*(18*channels
);
269 bufoff
= chanstart
+ newpos
;
270 while (bufoff
> end_adr
-18*channels
) {
271 bufoff
-=end_adr
-start_adr
;
274 ci
->seek_buffer(bufoff
);
278 if (bufoff
>ci
->filesize
-channels
*18) break; /* End of stream */
283 /* Is there data left in the file? */
284 (bufoff
<= ci
->filesize
-(18*channels
)) &&
285 /* Is there space in the output buffer? */
286 (sampleswritten
<= WAV_CHUNK_SIZE
-(32*channels
)) &&
287 /* Should we be looping? */
288 ((!looping
) || bufoff
<= end_adr
-18*channels
))
290 /* decode first/only channel */
295 buf
= ci
->request_buffer(&n
, 18);
298 DEBUGF("ADX: couldn't get buffer at %lx\n",
303 scale
= ((buf
[0] << 8) | (buf
[1])) +1;
305 for (i
= 2; i
< 18; i
++)
307 d
= (buf
[i
] >> 4) & 15;
309 ch1_0
= d
*scale
+ ((coef1
*ch1_1
+ coef2
*ch1_2
) >> 12);
310 if (ch1_0
> 32767) ch1_0
= 32767;
311 else if (ch1_0
< -32768) ch1_0
= -32768;
312 samples
[sampleswritten
] = ch1_0
;
313 sampleswritten
+=channels
;
314 ch1_2
= ch1_1
; ch1_1
= ch1_0
;
318 ch1_0
= d
*scale
+ ((coef1
*ch1_1
+ coef2
*ch1_2
) >> 12);
319 if (ch1_0
> 32767) ch1_0
= 32767;
320 else if (ch1_0
< -32768) ch1_0
= -32768;
321 samples
[sampleswritten
] = ch1_0
;
322 sampleswritten
+=channels
;
323 ch1_2
= ch1_1
; ch1_1
= ch1_0
;
326 ci
->advance_buffer(18);
329 /* decode second channel */
333 buf
= ci
->request_buffer(&n
, 18);
336 DEBUGF("ADX: couldn't get buffer at %lx\n",
341 scale
= ((buf
[0] << 8)|(buf
[1]))+1;
345 for (i
= 2; i
< 18; i
++)
347 d
= (buf
[i
] >> 4) & 15;
349 ch2_0
= d
*scale
+ ((coef1
*ch2_1
+ coef2
*ch2_2
) >> 12);
350 if (ch2_0
> 32767) ch2_0
= 32767;
351 else if (ch2_0
< -32768) ch2_0
= -32768;
352 samples
[sampleswritten
] = ch2_0
;
354 ch2_2
= ch2_1
; ch2_1
= ch2_0
;
358 ch2_0
= d
*scale
+ ((coef1
*ch2_1
+ coef2
*ch2_2
) >> 12);
359 if (ch2_0
> 32767) ch2_0
= 32767;
360 else if (ch2_0
< -32768) ch2_0
= -32768;
361 samples
[sampleswritten
] = ch2_0
;
363 ch2_2
= ch2_1
; ch2_1
= ch2_0
;
366 ci
->advance_buffer(18);
367 sampleswritten
--; /* go back to first channel's next sample */
372 for (i
=0;i
<(channels
==1?32:64);i
++) samples
[sampleswritten
-i
-1]=
373 ((int32_t)samples
[sampleswritten
-i
-1])*fade_count
/fade_frames
;
374 if (fade_count
==0) {endofstream
=1; break;}
379 sampleswritten
>>= 1; /* make samples/channel */
381 ci
->pcmbuf_insert(samples
, NULL
, sampleswritten
);
384 ((end_adr
-start_adr
)*loop_count
+ bufoff
-chanstart
)*
385 1000LL/avgbytespersec
);
388 if (ci
->request_next_track())