Merge branch 'master' into android-test-plugins
[kugel-rb.git] / apps / codecs / adx.c
bloba1b57fce58d965c4ee57ef2957f2b7bb3063a9e9
1 /***************************************************************************
2 * __________ __ ___.
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
7 * \/ \/ \/ \/ \/
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 ****************************************************************************/
22 #include <limits.h>
23 #include "codeclib.h"
24 #include "inttypes.h"
25 #include "math.h"
26 #include "lib/fixedpoint.h"
28 CODEC_HEADER
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 */
34 #define LOOP_TIMES 2
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 static 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(enum codec_entry_call_reason reason)
50 if (reason == CODEC_LOAD) {
51 /* Generic codec initialisation */
52 /* we only render 16 bits */
53 ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
56 return CODEC_OK;
59 /* this is called for each file to process */
60 enum codec_status codec_run(void)
62 int channels;
63 int sampleswritten, i;
64 uint8_t *buf;
65 int32_t ch1_1, ch1_2, ch2_1, ch2_2; /* ADPCM history */
66 size_t n;
67 int endofstream; /* end of stream flag */
68 uint32_t avgbytespersec;
69 int looping; /* looping flag */
70 int loop_count; /* number of loops done so far */
71 int fade_count; /* countdown for fadeout */
72 int fade_frames; /* length of fade in frames */
73 off_t start_adr, end_adr; /* loop points */
74 off_t chanstart, bufoff;
75 /*long coef1=0x7298L,coef2=-0x3350L;*/
76 long coef1, coef2;
77 intptr_t param;
79 DEBUGF("ADX: next_track\n");
80 if (codec_init()) {
81 return CODEC_ERROR;
83 DEBUGF("ADX: after init\n");
85 /* init history */
86 ch1_1=ch1_2=ch2_1=ch2_2=0;
88 codec_set_replaygain(ci->id3);
90 /* Get header */
91 DEBUGF("ADX: request initial buffer\n");
92 ci->seek_buffer(0);
93 buf = ci->request_buffer(&n, 0x38);
94 if (!buf || n < 0x38) {
95 return CODEC_ERROR;
97 bufoff = 0;
98 DEBUGF("ADX: read size = %lx\n",(unsigned long)n);
100 /* Get file header for starting offset, channel count */
102 chanstart = ((buf[2] << 8) | buf[3]) + 4;
103 channels = buf[7];
105 /* useful for seeking and reporting current playback position */
106 avgbytespersec = ci->id3->frequency * 18 * channels / 32;
107 DEBUGF("avgbytespersec=%ld\n",(unsigned long)avgbytespersec);
109 /* calculate filter coefficients */
112 * A simple table of these coefficients would be nice, but
113 * some very odd frequencies are used and if I'm going to
114 * interpolate I might as well just go all the way and
115 * calclate them precisely.
116 * Speed is not an issue as this only needs to be done once per file.
119 const int64_t big28 = 0x10000000LL;
120 const int64_t big32 = 0x100000000LL;
121 int64_t frequency = ci->id3->frequency;
122 int64_t phasemultiple = cutoff*big32/frequency;
124 long z;
125 int64_t a;
126 const int64_t b = (M_SQRT2*big28)-big28;
127 int64_t c;
128 int64_t d;
130 fp_sincos((unsigned long)phasemultiple,&z);
132 a = (M_SQRT2*big28)-(z*big28/LONG_MAX);
135 * In the long passed to fsqrt there are only 4 nonfractional bits,
136 * which is sufficient here, but this is the only reason why I don't
137 * use 32 fractional bits everywhere.
139 d = fp_sqrt((a+b)*(a-b)/big28,28);
140 c = (a-d)*big28/b;
142 coef1 = (c*8192) >> 28;
143 coef2 = (c*c/big28*-4096) >> 28;
144 DEBUGF("ADX: samprate=%ld ",(long)frequency);
145 DEBUGF("coef1 %04x ",(unsigned int)(coef1*4));
146 DEBUGF("coef2 %04x\n",(unsigned int)(coef2*-4));
149 /* Get loop data */
151 looping = 0; start_adr = 0; end_adr = 0;
152 if (!memcmp(buf+0x10,"\x01\xF4\x03\x00",4)) {
153 /* Soul Calibur 2 style (type 03) */
154 DEBUGF("ADX: type 03 found\n");
155 /* check if header is too small for loop data */
156 if (chanstart-6 < 0x2c) looping=0;
157 else {
158 looping = (buf[0x18]) ||
159 (buf[0x19]) ||
160 (buf[0x1a]) ||
161 (buf[0x1b]);
162 end_adr = (buf[0x28]<<24) |
163 (buf[0x29]<<16) |
164 (buf[0x2a]<<8) |
165 (buf[0x2b]);
167 start_adr = (
168 (buf[0x1c]<<24) |
169 (buf[0x1d]<<16) |
170 (buf[0x1e]<<8) |
171 (buf[0x1f])
172 )/32*channels*18+chanstart;
174 } else if (!memcmp(buf+0x10,"\x01\xF4\x04\x00",4)) {
175 /* Standard (type 04) */
176 DEBUGF("ADX: type 04 found\n");
177 /* check if header is too small for loop data */
178 if (chanstart-6 < 0x38) looping=0;
179 else {
180 looping = (buf[0x24]) ||
181 (buf[0x25]) ||
182 (buf[0x26]) ||
183 (buf[0x27]);
184 end_adr = (buf[0x34]<<24) |
185 (buf[0x35]<<16) |
186 (buf[0x36]<<8) |
187 buf[0x37];
188 start_adr = (
189 (buf[0x28]<<24) |
190 (buf[0x29]<<16) |
191 (buf[0x2a]<<8) |
192 (buf[0x2b])
193 )/32*channels*18+chanstart;
195 } else {
196 DEBUGF("ADX: error, couldn't determine ADX type\n");
197 return CODEC_ERROR;
200 if (looping) {
201 DEBUGF("ADX: looped, start: %lx end: %lx\n",start_adr,end_adr);
202 } else {
203 DEBUGF("ADX: not looped\n");
206 /* advance to first frame */
207 DEBUGF("ADX: first frame at %lx\n",chanstart);
208 bufoff = chanstart;
210 /* get in position */
211 ci->seek_buffer(bufoff);
214 /* setup pcm buffer format */
215 ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
216 if (channels == 2) {
217 ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
218 } else if (channels == 1) {
219 ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
220 } else {
221 DEBUGF("ADX CODEC_ERROR: more than 2 channels\n");
222 return CODEC_ERROR;
225 endofstream = 0;
226 loop_count = 0;
227 fade_count = -1; /* disable fade */
228 fade_frames = 1;
230 /* The main decoder loop */
232 while (!endofstream) {
233 enum codec_command_action action = ci->get_command(&param);
235 if (action == CODEC_ACTION_HALT)
236 break;
238 /* do we need to loop? */
239 if (bufoff > end_adr-18*channels && looping) {
240 DEBUGF("ADX: loop!\n");
241 /* check for endless looping */
242 if (ci->global_settings->repeat_mode==REPEAT_ONE) {
243 loop_count=0;
244 fade_count = -1; /* disable fade */
245 } else {
246 /* otherwise start fade after LOOP_TIMES loops */
247 loop_count++;
248 if (loop_count >= LOOP_TIMES && fade_count < 0) {
249 /* frames to fade over */
250 fade_frames = FADE_LENGTH*ci->id3->frequency/32/1000;
251 /* volume relative to fade_frames */
252 fade_count = fade_frames;
253 DEBUGF("ADX: fade_frames = %d\n",fade_frames);
256 bufoff = start_adr;
257 ci->seek_buffer(bufoff);
260 /* do we need to seek? */
261 if (action == CODEC_ACTION_SEEK_TIME) {
262 uint32_t newpos;
264 DEBUGF("ADX: seek to %ldms\n", (long)param);
266 endofstream = 0;
267 loop_count = 0;
268 fade_count = -1; /* disable fade */
269 fade_frames = 1;
271 newpos = (((uint64_t)avgbytespersec*param)
272 / (1000LL*18*channels))*(18*channels);
273 bufoff = chanstart + newpos;
274 while (bufoff > end_adr-18*channels) {
275 bufoff-=end_adr-start_adr;
276 loop_count++;
278 ci->seek_buffer(bufoff);
279 ci->seek_complete();
282 if (bufoff>ci->filesize-channels*18) break; /* End of stream */
284 sampleswritten=0;
286 while (
287 /* Is there data left in the file? */
288 (bufoff <= ci->filesize-(18*channels)) &&
289 /* Is there space in the output buffer? */
290 (sampleswritten <= WAV_CHUNK_SIZE-(32*channels)) &&
291 /* Should we be looping? */
292 ((!looping) || bufoff <= end_adr-18*channels))
294 /* decode first/only channel */
295 int32_t scale;
296 int32_t ch1_0, d;
298 /* fetch a frame */
299 buf = ci->request_buffer(&n, 18);
301 if (!buf || n!=18) {
302 DEBUGF("ADX: couldn't get buffer at %lx\n",
303 bufoff);
304 return CODEC_ERROR;
307 scale = ((buf[0] << 8) | (buf[1])) +1;
309 for (i = 2; i < 18; i++)
311 d = (buf[i] >> 4) & 15;
312 if (d & 8) d-= 16;
313 ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
314 if (ch1_0 > 32767) ch1_0 = 32767;
315 else if (ch1_0 < -32768) ch1_0 = -32768;
316 samples[sampleswritten] = ch1_0;
317 sampleswritten+=channels;
318 ch1_2 = ch1_1; ch1_1 = ch1_0;
320 d = buf[i] & 15;
321 if (d & 8) d -= 16;
322 ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
323 if (ch1_0 > 32767) ch1_0 = 32767;
324 else if (ch1_0 < -32768) ch1_0 = -32768;
325 samples[sampleswritten] = ch1_0;
326 sampleswritten+=channels;
327 ch1_2 = ch1_1; ch1_1 = ch1_0;
329 bufoff+=18;
330 ci->advance_buffer(18);
332 if (channels == 2) {
333 /* decode second channel */
334 int32_t scale;
335 int32_t ch2_0, d;
337 buf = ci->request_buffer(&n, 18);
339 if (!buf || n!=18) {
340 DEBUGF("ADX: couldn't get buffer at %lx\n",
341 bufoff);
342 return CODEC_ERROR;
345 scale = ((buf[0] << 8)|(buf[1]))+1;
347 sampleswritten-=63;
349 for (i = 2; i < 18; i++)
351 d = (buf[i] >> 4) & 15;
352 if (d & 8) d-= 16;
353 ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
354 if (ch2_0 > 32767) ch2_0 = 32767;
355 else if (ch2_0 < -32768) ch2_0 = -32768;
356 samples[sampleswritten] = ch2_0;
357 sampleswritten+=2;
358 ch2_2 = ch2_1; ch2_1 = ch2_0;
360 d = buf[i] & 15;
361 if (d & 8) d -= 16;
362 ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
363 if (ch2_0 > 32767) ch2_0 = 32767;
364 else if (ch2_0 < -32768) ch2_0 = -32768;
365 samples[sampleswritten] = ch2_0;
366 sampleswritten+=2;
367 ch2_2 = ch2_1; ch2_1 = ch2_0;
369 bufoff+=18;
370 ci->advance_buffer(18);
371 sampleswritten--; /* go back to first channel's next sample */
374 if (fade_count>0) {
375 fade_count--;
376 for (i=0;i<(channels==1?32:64);i++) samples[sampleswritten-i-1]=
377 ((int32_t)samples[sampleswritten-i-1])*fade_count/fade_frames;
378 if (fade_count==0) {endofstream=1; break;}
382 if (channels == 2)
383 sampleswritten >>= 1; /* make samples/channel */
385 ci->pcmbuf_insert(samples, NULL, sampleswritten);
387 ci->set_elapsed(
388 ((end_adr-start_adr)*loop_count + bufoff-chanstart)*
389 1000LL/avgbytespersec);
392 return CODEC_OK;