Prepare new maemo release
[maemo-rb.git] / lib / rbcodec / codecs / adx.c
blob0c67fc8d6e7c9b2feca7686942c18ef7c6580e37
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 >> 3);
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",3)) {
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",3)) {
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 /* is file using encryption */
201 if (buf[0x13]==0x08) {
202 DEBUGF("ADX: error, encrypted ADX not supported\n");
203 return false;
206 if (looping) {
207 DEBUGF("ADX: looped, start: %lx end: %lx\n",start_adr,end_adr);
208 } else {
209 DEBUGF("ADX: not looped\n");
212 /* advance to first frame */
213 DEBUGF("ADX: first frame at %lx\n",chanstart);
214 bufoff = chanstart;
216 /* get in position */
217 ci->seek_buffer(bufoff);
218 ci->set_elapsed(0);
220 /* setup pcm buffer format */
221 ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
222 if (channels == 2) {
223 ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
224 } else if (channels == 1) {
225 ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
226 } else {
227 DEBUGF("ADX CODEC_ERROR: more than 2 channels\n");
228 return CODEC_ERROR;
231 endofstream = 0;
232 loop_count = 0;
233 fade_count = -1; /* disable fade */
234 fade_frames = 1;
236 /* The main decoder loop */
238 while (!endofstream) {
239 enum codec_command_action action = ci->get_command(&param);
241 if (action == CODEC_ACTION_HALT)
242 break;
244 /* do we need to loop? */
245 if (bufoff > end_adr-18*channels && looping) {
246 DEBUGF("ADX: loop!\n");
247 /* check for endless looping */
248 if (ci->loop_track()) {
249 loop_count=0;
250 fade_count = -1; /* disable fade */
251 } else {
252 /* otherwise start fade after LOOP_TIMES loops */
253 loop_count++;
254 if (loop_count >= LOOP_TIMES && fade_count < 0) {
255 /* frames to fade over */
256 fade_frames = FADE_LENGTH*ci->id3->frequency/32/1000;
257 /* volume relative to fade_frames */
258 fade_count = fade_frames;
259 DEBUGF("ADX: fade_frames = %d\n",fade_frames);
262 bufoff = start_adr;
263 ci->seek_buffer(bufoff);
266 /* do we need to seek? */
267 if (action == CODEC_ACTION_SEEK_TIME) {
268 uint32_t newpos;
270 DEBUGF("ADX: seek to %ldms\n", (long)param);
272 endofstream = 0;
273 loop_count = 0;
274 fade_count = -1; /* disable fade */
275 fade_frames = 1;
277 newpos = (((uint64_t)avgbytespersec*param)
278 / (1000LL*18*channels))*(18*channels);
279 bufoff = chanstart + newpos;
280 while (bufoff > end_adr-18*channels) {
281 bufoff-=end_adr-start_adr;
282 loop_count++;
284 ci->seek_buffer(bufoff);
286 ci->set_elapsed(
287 ((end_adr-start_adr)*loop_count + bufoff-chanstart)*
288 1000LL/avgbytespersec);
290 ci->seek_complete();
293 if (bufoff>ci->filesize-channels*18) break; /* End of stream */
295 sampleswritten=0;
297 while (
298 /* Is there data left in the file? */
299 (bufoff <= ci->filesize-(18*channels)) &&
300 /* Is there space in the output buffer? */
301 (sampleswritten <= WAV_CHUNK_SIZE-(32*channels)) &&
302 /* Should we be looping? */
303 ((!looping) || bufoff <= end_adr-18*channels))
305 /* decode first/only channel */
306 int32_t scale;
307 int32_t ch1_0, d;
309 /* fetch a frame */
310 buf = ci->request_buffer(&n, 18);
312 if (!buf || n!=18) {
313 DEBUGF("ADX: couldn't get buffer at %lx\n",
314 bufoff);
315 return CODEC_ERROR;
318 scale = ((buf[0] << 8) | (buf[1])) +1;
320 for (i = 2; i < 18; i++)
322 d = (buf[i] >> 4) & 15;
323 if (d & 8) d-= 16;
324 ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
325 if (ch1_0 > 32767) ch1_0 = 32767;
326 else if (ch1_0 < -32768) ch1_0 = -32768;
327 samples[sampleswritten] = ch1_0;
328 sampleswritten+=channels;
329 ch1_2 = ch1_1; ch1_1 = ch1_0;
331 d = buf[i] & 15;
332 if (d & 8) d -= 16;
333 ch1_0 = d*scale + ((coef1*ch1_1 + coef2*ch1_2) >> 12);
334 if (ch1_0 > 32767) ch1_0 = 32767;
335 else if (ch1_0 < -32768) ch1_0 = -32768;
336 samples[sampleswritten] = ch1_0;
337 sampleswritten+=channels;
338 ch1_2 = ch1_1; ch1_1 = ch1_0;
340 bufoff+=18;
341 ci->advance_buffer(18);
343 if (channels == 2) {
344 /* decode second channel */
345 int32_t scale;
346 int32_t ch2_0, d;
348 buf = ci->request_buffer(&n, 18);
350 if (!buf || n!=18) {
351 DEBUGF("ADX: couldn't get buffer at %lx\n",
352 bufoff);
353 return CODEC_ERROR;
356 scale = ((buf[0] << 8)|(buf[1]))+1;
358 sampleswritten-=63;
360 for (i = 2; i < 18; i++)
362 d = (buf[i] >> 4) & 15;
363 if (d & 8) d-= 16;
364 ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
365 if (ch2_0 > 32767) ch2_0 = 32767;
366 else if (ch2_0 < -32768) ch2_0 = -32768;
367 samples[sampleswritten] = ch2_0;
368 sampleswritten+=2;
369 ch2_2 = ch2_1; ch2_1 = ch2_0;
371 d = buf[i] & 15;
372 if (d & 8) d -= 16;
373 ch2_0 = d*scale + ((coef1*ch2_1 + coef2*ch2_2) >> 12);
374 if (ch2_0 > 32767) ch2_0 = 32767;
375 else if (ch2_0 < -32768) ch2_0 = -32768;
376 samples[sampleswritten] = ch2_0;
377 sampleswritten+=2;
378 ch2_2 = ch2_1; ch2_1 = ch2_0;
380 bufoff+=18;
381 ci->advance_buffer(18);
382 sampleswritten--; /* go back to first channel's next sample */
385 if (fade_count>0) {
386 fade_count--;
387 for (i=0;i<(channels==1?32:64);i++) samples[sampleswritten-i-1]=
388 ((int32_t)samples[sampleswritten-i-1])*fade_count/fade_frames;
389 if (fade_count==0) {endofstream=1; break;}
393 if (channels == 2)
394 sampleswritten >>= 1; /* make samples/channel */
396 ci->pcmbuf_insert(samples, NULL, sampleswritten);
398 ci->set_elapsed(
399 ((end_adr-start_adr)*loop_count + bufoff-chanstart)*
400 1000LL/avgbytespersec);
403 return CODEC_OK;