Mercurial > libavcodec.hg
comparison atrac1.c @ 10157:178274d5fa1d libavcodec
Initial commit of the atrac1 decoder, not hooked up yet
author | banan |
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date | Thu, 10 Sep 2009 18:47:02 +0000 |
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children | e1bb4cf6e659 |
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10156:b49a14edba84 | 10157:178274d5fa1d |
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1 /* | |
2 * Atrac 1 compatible decoder | |
3 * Copyright (c) 2009 Maxim Poliakovski | |
4 * Copyright (c) 2009 Benjamin Larsson | |
5 * | |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 /** | |
24 * @file libavcodec/atrac1.c | |
25 * Atrac 1 compatible decoder. | |
26 * This decoder handles raw ATRAC1 data. | |
27 */ | |
28 | |
29 /* Many thanks to Tim Craig for all the help! */ | |
30 | |
31 #include <math.h> | |
32 #include <stddef.h> | |
33 #include <stdio.h> | |
34 | |
35 #include "avcodec.h" | |
36 #include "get_bits.h" | |
37 #include "dsputil.h" | |
38 | |
39 #include "atrac.h" | |
40 #include "atrac1data.h" | |
41 | |
42 #define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit | |
43 #define AT1_SU_SIZE 212 ///< number of bytes in a sound unit | |
44 #define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit | |
45 #define AT1_FRAME_SIZE AT1_SU_SIZE * 2 | |
46 #define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 | |
47 #define AT1_MAX_CHANNELS 2 | |
48 | |
49 #define AT1_QMF_BANDS 3 | |
50 #define IDX_LOW_BAND 0 | |
51 #define IDX_MID_BAND 1 | |
52 #define IDX_HIGH_BAND 2 | |
53 | |
54 /** | |
55 * Sound unit struct, one unit is used per channel | |
56 */ | |
57 typedef struct { | |
58 int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band | |
59 int num_bfus; ///< number of Block Floating Units | |
60 int idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU | |
61 int idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU | |
62 float* spectrum[2]; | |
63 DECLARE_ALIGNED_16(float,spec1[AT1_SU_SAMPLES]); ///< mdct buffer | |
64 DECLARE_ALIGNED_16(float,spec2[AT1_SU_SAMPLES]); ///< mdct buffer | |
65 DECLARE_ALIGNED_16(float,fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter | |
66 DECLARE_ALIGNED_16(float,snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter | |
67 DECLARE_ALIGNED_16(float,last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter | |
68 } AT1SUCtx; | |
69 | |
70 /** | |
71 * The atrac1 context, holds all needed parameters for decoding | |
72 */ | |
73 typedef struct { | |
74 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit | |
75 DECLARE_ALIGNED_16(float,spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer | |
76 DECLARE_ALIGNED_16(float,short_buf[64]); ///< buffer for the short mode | |
77 DECLARE_ALIGNED_16(float, low[256]); | |
78 DECLARE_ALIGNED_16(float, mid[256]); | |
79 DECLARE_ALIGNED_16(float,high[512]); | |
80 float* bands[3]; | |
81 float out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]; | |
82 MDCTContext mdct_ctx[3]; | |
83 int channels; | |
84 DSPContext dsp; | |
85 } AT1Ctx; | |
86 | |
87 static float *short_window; | |
88 static float *mid_window; | |
89 DECLARE_ALIGNED_16(static float, long_window[256]); | |
90 static float *window_per_band[3]; | |
91 | |
92 /** size of the transform in samples in the long mode for each QMF band */ | |
93 static const uint16_t samples_per_band[3] = {128, 128, 256}; | |
94 static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; | |
95 | |
96 | |
97 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, int rev_spec) | |
98 { | |
99 MDCTContext* mdct_context; | |
100 int transf_size = 1 << nbits; | |
101 | |
102 mdct_context = &q->mdct_ctx[nbits - 5 - (nbits>6)]; | |
103 | |
104 if (rev_spec) { | |
105 int i; | |
106 for (i=0 ; i<transf_size/2 ; i++) | |
107 FFSWAP(float, spec[i], spec[transf_size-1-i]); | |
108 } | |
109 ff_imdct_half(mdct_context,out,spec); | |
110 } | |
111 | |
112 | |
113 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) | |
114 { | |
115 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; | |
116 unsigned int start_pos, ref_pos=0, pos = 0; | |
117 | |
118 for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) { | |
119 band_samples = samples_per_band[band_num]; | |
120 log2_block_count = su->log2_block_count[band_num]; | |
121 | |
122 /* number of mdct blocks in the current QMF band: 1 - for long mode */ | |
123 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ | |
124 num_blocks = 1 << log2_block_count; | |
125 | |
126 /* mdct block size in samples: 128 (long mode, low & mid bands), */ | |
127 /* 256 (long mode, high band) and 32 (short mode, all bands) */ | |
128 block_size = band_samples >> log2_block_count; | |
129 | |
130 /* calc transform size in bits according to the block_size_mode */ | |
131 nbits = mdct_long_nbits[band_num] - log2_block_count; | |
132 | |
133 if (nbits!=5 && nbits!=7 && nbits!=8) | |
134 return -1; | |
135 | |
136 if (num_blocks == 1) { | |
137 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num); | |
138 pos += block_size; // move to the next mdct block in the spectrum | |
139 } else { | |
140 /* calc start position for the 1st short block: 96(128) or 112(256) */ | |
141 start_pos = (band_samples * (num_blocks - 1)) >> (log2_block_count + 1); | |
142 memset(&su->spectrum[0][ref_pos], 0, sizeof(float) * (band_samples * 2)); | |
143 | |
144 for (; num_blocks!=0 ; num_blocks--) { | |
145 /* use hardcoded nbits for the short mode */ | |
146 at1_imdct(q, &q->spec[pos], q->short_buf, 5, band_num); | |
147 | |
148 /* overlap and window between short blocks */ | |
149 q->dsp.vector_fmul_window(&su->spectrum[0][ref_pos+start_pos], | |
150 &su->spectrum[0][ref_pos+start_pos],q->short_buf,short_window, 0, 16); | |
151 start_pos += 32; // use hardcoded block_size | |
152 pos += 32; | |
153 } | |
154 } | |
155 | |
156 /* overlap and window with the previous frame and output the result */ | |
157 q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos+band_samples/2], | |
158 &su->spectrum[0][ref_pos], window_per_band[band_num], 0, band_samples/2); | |
159 | |
160 ref_pos += band_samples; | |
161 } | |
162 | |
163 /* Swap buffers so the mdct overlap works */ | |
164 FFSWAP(float*, su->spectrum[0], su->spectrum[1]); | |
165 | |
166 return 0; | |
167 } | |
168 | |
169 | |
170 static int at1_parse_block_size_mode(GetBitContext* gb, int log2_block_count[AT1_QMF_BANDS]) | |
171 { | |
172 int log2_block_count_tmp, i; | |
173 | |
174 for(i=0 ; i<2 ; i++) { | |
175 /* low and mid band */ | |
176 log2_block_count_tmp = get_bits(gb, 2); | |
177 if (log2_block_count_tmp & 1) | |
178 return -1; | |
179 log2_block_count[i] = 2 - log2_block_count_tmp; | |
180 } | |
181 | |
182 /* high band */ | |
183 log2_block_count_tmp = get_bits(gb, 2); | |
184 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) | |
185 return -1; | |
186 log2_block_count[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; | |
187 | |
188 skip_bits(gb, 2); | |
189 return 0; | |
190 } | |
191 | |
192 | |
193 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, float spec[AT1_SU_SAMPLES]) | |
194 { | |
195 int bits_used, band_num, bfu_num, i; | |
196 | |
197 /* parse the info byte (2nd byte) telling how much BFUs were coded */ | |
198 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; | |
199 | |
200 /* calc number of consumed bits: | |
201 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) | |
202 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */ | |
203 bits_used = su->num_bfus * 10 + 32 + | |
204 bfu_amount_tab2[get_bits(gb, 2)] + | |
205 (bfu_amount_tab3[get_bits(gb, 3)] << 1); | |
206 | |
207 /* get word length index (idwl) for each BFU */ | |
208 for (i=0 ; i<su->num_bfus ; i++) | |
209 su->idwls[i] = get_bits(gb, 4); | |
210 | |
211 /* get scalefactor index (idsf) for each BFU */ | |
212 for (i=0 ; i<su->num_bfus ; i++) | |
213 su->idsfs[i] = get_bits(gb, 6); | |
214 | |
215 /* zero idwl/idsf for empty BFUs */ | |
216 for (i = su->num_bfus; i < AT1_MAX_BFU; i++) | |
217 su->idwls[i] = su->idsfs[i] = 0; | |
218 | |
219 /* read in the spectral data and reconstruct MDCT spectrum of this channel */ | |
220 for (band_num=0 ; band_num<AT1_QMF_BANDS ; band_num++) { | |
221 for (bfu_num=bfu_bands_t[band_num] ; bfu_num<bfu_bands_t[band_num+1] ; bfu_num++) { | |
222 int pos; | |
223 | |
224 int num_specs = specs_per_bfu[bfu_num]; | |
225 int word_len = !!su->idwls[bfu_num] + su->idwls[bfu_num]; | |
226 float scale_factor = sf_table[su->idsfs[bfu_num]]; | |
227 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ | |
228 | |
229 /* check for bitstream overflow */ | |
230 if (bits_used > AT1_SU_MAX_BITS) | |
231 return -1; | |
232 | |
233 /* get the position of the 1st spec according to the block size mode */ | |
234 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; | |
235 | |
236 if (word_len) { | |
237 float max_quant = 1.0/(float)((1 << (word_len - 1)) - 1); | |
238 | |
239 for (i=0 ; i<num_specs ; i++) { | |
240 /* read in a quantized spec and convert it to | |
241 * signed int and then inverse quantization | |
242 */ | |
243 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; | |
244 } | |
245 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */ | |
246 memset(&spec[pos], 0, num_specs*sizeof(float)); | |
247 } | |
248 } | |
249 } | |
250 | |
251 return 0; | |
252 } | |
253 | |
254 | |
255 void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) | |
256 { | |
257 float temp[256]; | |
258 float iqmf_temp[512 + 46]; | |
259 | |
260 /* combine low and middle bands */ | |
261 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); | |
262 | |
263 /* delay the signal of the high band by 23 samples */ | |
264 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float)*23); | |
265 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float)*256); | |
266 | |
267 /* combine (low + middle) and high bands */ | |
268 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); | |
269 } | |
270 | |
271 | |
272 static int atrac1_decode_frame(AVCodecContext *avctx, | |
273 void *data, int *data_size, | |
274 AVPacket *avpkt) | |
275 { | |
276 const uint8_t *buf = avpkt->data; | |
277 int buf_size = avpkt->size; | |
278 AT1Ctx *q = avctx->priv_data; | |
279 int ch, ret, i; | |
280 GetBitContext gb; | |
281 float* samples = data; | |
282 | |
283 | |
284 if (buf_size < 212 * q->channels) { | |
285 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); | |
286 return -1; | |
287 } | |
288 | |
289 for (ch=0 ; ch<q->channels ; ch++) { | |
290 AT1SUCtx* su = &q->SUs[ch]; | |
291 | |
292 init_get_bits(&gb, &buf[212*ch], 212*8); | |
293 | |
294 /* parse block_size_mode, 1st byte */ | |
295 ret = at1_parse_block_size_mode(&gb, su->log2_block_count); | |
296 if (ret < 0) | |
297 return ret; | |
298 | |
299 ret = at1_unpack_dequant(&gb, su, q->spec); | |
300 if (ret < 0) | |
301 return ret; | |
302 | |
303 ret = at1_imdct_block(su, q); | |
304 if (ret < 0) | |
305 return ret; | |
306 at1_subband_synthesis(q, su, q->out_samples[ch]); | |
307 } | |
308 | |
309 /* round, convert to 16bit and interleave */ | |
310 if (q->channels == 1) { | |
311 /* mono */ | |
312 q->dsp.vector_clipf(samples, q->out_samples[0], -32700./(1<<15), 32700./(1<<15), AT1_SU_SAMPLES); | |
313 } else { | |
314 /* stereo */ | |
315 for (i = 0; i < AT1_SU_SAMPLES; i++) { | |
316 samples[i*2] = av_clipf(q->out_samples[0][i], -32700./(1<<15), 32700./(1<<15)); | |
317 samples[i*2+1] = av_clipf(q->out_samples[1][i], -32700./(1<<15), 32700./(1<<15)); | |
318 } | |
319 } | |
320 | |
321 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples); | |
322 return avctx->block_align; | |
323 } | |
324 | |
325 | |
326 static av_cold void init_mdct_windows(void) | |
327 { | |
328 int i; | |
329 | |
330 /** The mid and long windows uses the same sine window splitted | |
331 * in the middle and wrapped into zero/one regions as follows: | |
332 * | |
333 * region of "ones" | |
334 * ------------- | |
335 * / | |
336 * / 1st half | |
337 * / of the sine | |
338 * / window | |
339 * ---------/ | |
340 * zero region | |
341 * | |
342 * The mid and short windows are subsets of the long window. | |
343 */ | |
344 | |
345 /* Build "zero" region */ | |
346 memset(long_window, 0, sizeof(long_window)); | |
347 /* Build sine window region */ | |
348 short_window = &long_window[112]; | |
349 ff_sine_window_init(short_window,32); | |
350 /* Build "ones" region */ | |
351 for (i = 0; i < 112; i++) | |
352 long_window[144 + i] = 1.0f; | |
353 /* Save the mid window subset start */ | |
354 mid_window = &long_window[64]; | |
355 | |
356 /* Prepare the window table */ | |
357 window_per_band[0] = mid_window; | |
358 window_per_band[1] = mid_window; | |
359 window_per_band[2] = long_window; | |
360 } | |
361 | |
362 static av_cold int atrac1_decode_init(AVCodecContext *avctx) | |
363 { | |
364 AT1Ctx *q = avctx->priv_data; | |
365 | |
366 avctx->sample_fmt = SAMPLE_FMT_FLT; | |
367 | |
368 q->channels = avctx->channels; | |
369 | |
370 /* Init the mdct transforms */ | |
371 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1<<15)); | |
372 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1<<15)); | |
373 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1<<15)); | |
374 init_mdct_windows(); | |
375 | |
376 atrac_generate_tables(); | |
377 | |
378 dsputil_init(&q->dsp, avctx); | |
379 | |
380 q->bands[0] = q->low; | |
381 q->bands[1] = q->mid; | |
382 q->bands[2] = q->high; | |
383 | |
384 /* Prepare the mdct overlap buffers */ | |
385 q->SUs[0].spectrum[0] = q->SUs[0].spec1; | |
386 q->SUs[0].spectrum[1] = q->SUs[0].spec2; | |
387 q->SUs[1].spectrum[0] = q->SUs[1].spec1; | |
388 q->SUs[1].spectrum[1] = q->SUs[1].spec2; | |
389 | |
390 return 0; | |
391 } | |
392 | |
393 AVCodec atrac1_decoder = { | |
394 .name = "atrac1", | |
395 .type = CODEC_TYPE_AUDIO, | |
396 .id = CODEC_ID_ATRAC1, | |
397 .priv_data_size = sizeof(AT1Ctx), | |
398 .init = atrac1_decode_init, | |
399 .close = NULL, | |
400 .decode = atrac1_decode_frame, | |
401 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), | |
402 }; |