Mercurial > libavcodec.hg
annotate atrac1.c @ 10218:84a9a55135f4 libavcodec
Add forgotten cleanup function in atrac1.
author | banan |
---|---|
date | Mon, 21 Sep 2009 21:00:18 +0000 |
parents | 35f1814a6496 |
children | 47f2b03e0c62 |
rev | line source |
---|---|
10157 | 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 float* spectrum[2]; | |
10197 | 61 DECLARE_ALIGNED_16(float, spec1[AT1_SU_SAMPLES]); ///< mdct buffer |
62 DECLARE_ALIGNED_16(float, spec2[AT1_SU_SAMPLES]); ///< mdct buffer | |
63 DECLARE_ALIGNED_16(float, fst_qmf_delay[46]); ///< delay line for the 1st stacked QMF filter | |
64 DECLARE_ALIGNED_16(float, snd_qmf_delay[46]); ///< delay line for the 2nd stacked QMF filter | |
65 DECLARE_ALIGNED_16(float, last_qmf_delay[256+23]); ///< delay line for the last stacked QMF filter | |
10157 | 66 } AT1SUCtx; |
67 | |
68 /** | |
69 * The atrac1 context, holds all needed parameters for decoding | |
70 */ | |
71 typedef struct { | |
72 AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit | |
10197 | 73 DECLARE_ALIGNED_16(float, spec[AT1_SU_SAMPLES]); ///< the mdct spectrum buffer |
10185 | 74 |
10197 | 75 DECLARE_ALIGNED_16(float, low[256]); |
76 DECLARE_ALIGNED_16(float, mid[256]); | |
77 DECLARE_ALIGNED_16(float, high[512]); | |
10157 | 78 float* bands[3]; |
10197 | 79 DECLARE_ALIGNED_16(float, out_samples[AT1_MAX_CHANNELS][AT1_SU_SAMPLES]); |
10199 | 80 FFTContext mdct_ctx[3]; |
10157 | 81 int channels; |
82 DSPContext dsp; | |
83 } AT1Ctx; | |
84 | |
10185 | 85 DECLARE_ALIGNED_16(static float, short_window[32]); |
10157 | 86 |
87 /** size of the transform in samples in the long mode for each QMF band */ | |
88 static const uint16_t samples_per_band[3] = {128, 128, 256}; | |
89 static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; | |
90 | |
91 | |
10170 | 92 static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, |
93 int rev_spec) | |
10157 | 94 { |
10216 | 95 FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; |
10157 | 96 int transf_size = 1 << nbits; |
97 | |
98 if (rev_spec) { | |
99 int i; | |
10197 | 100 for (i = 0; i < transf_size / 2; i++) |
10170 | 101 FFSWAP(float, spec[i], spec[transf_size - 1 - i]); |
10157 | 102 } |
10170 | 103 ff_imdct_half(mdct_context, out, spec); |
10157 | 104 } |
105 | |
106 | |
107 static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) | |
108 { | |
10197 | 109 int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; |
10198
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110 unsigned int start_pos, ref_pos = 0, pos = 0; |
10157 | 111 |
10197 | 112 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
10157 | 113 band_samples = samples_per_band[band_num]; |
114 log2_block_count = su->log2_block_count[band_num]; | |
115 | |
116 /* number of mdct blocks in the current QMF band: 1 - for long mode */ | |
117 /* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ | |
118 num_blocks = 1 << log2_block_count; | |
119 | |
120 /* mdct block size in samples: 128 (long mode, low & mid bands), */ | |
121 /* 256 (long mode, high band) and 32 (short mode, all bands) */ | |
122 block_size = band_samples >> log2_block_count; | |
123 | |
124 /* calc transform size in bits according to the block_size_mode */ | |
125 nbits = mdct_long_nbits[band_num] - log2_block_count; | |
126 | |
10197 | 127 if (nbits != 5 && nbits != 7 && nbits != 8) |
10157 | 128 return -1; |
129 | |
130 if (num_blocks == 1) { | |
10189 | 131 /* long blocks */ |
10157 | 132 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos], nbits, band_num); |
133 pos += block_size; // move to the next mdct block in the spectrum | |
10185 | 134 |
135 /* overlap and window long blocks */ | |
10197 | 136 q->dsp.vector_fmul_window(q->bands[band_num], &su->spectrum[1][ref_pos + band_samples - 16], |
137 &su->spectrum[0][ref_pos], short_window, 0, 16); | |
138 memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); | |
10157 | 139 } else { |
10189 | 140 /* short blocks */ |
10185 | 141 float *prev_buf; |
10189 | 142 start_pos = 0; |
10197 | 143 prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; |
144 for (; num_blocks != 0; num_blocks--) { | |
145 at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], 5, band_num); | |
10157 | 146 |
147 /* overlap and window between short blocks */ | |
10189 | 148 q->dsp.vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, |
10197 | 149 &su->spectrum[0][ref_pos + start_pos], short_window, 0, 16); |
10185 | 150 |
10197 | 151 prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; |
10157 | 152 start_pos += 32; // use hardcoded block_size |
153 pos += 32; | |
154 } | |
155 } | |
156 ref_pos += band_samples; | |
157 } | |
158 | |
159 /* Swap buffers so the mdct overlap works */ | |
160 FFSWAP(float*, su->spectrum[0], su->spectrum[1]); | |
161 | |
162 return 0; | |
163 } | |
164 | |
10170 | 165 /** |
166 * Parse the block size mode byte | |
167 */ | |
10157 | 168 |
10170 | 169 static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) |
10157 | 170 { |
171 int log2_block_count_tmp, i; | |
172 | |
10197 | 173 for (i = 0; i < 2; i++) { |
10157 | 174 /* low and mid band */ |
175 log2_block_count_tmp = get_bits(gb, 2); | |
176 if (log2_block_count_tmp & 1) | |
177 return -1; | |
10170 | 178 log2_block_cnt[i] = 2 - log2_block_count_tmp; |
10157 | 179 } |
180 | |
181 /* high band */ | |
182 log2_block_count_tmp = get_bits(gb, 2); | |
183 if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) | |
184 return -1; | |
10170 | 185 log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; |
10157 | 186 |
187 skip_bits(gb, 2); | |
188 return 0; | |
189 } | |
190 | |
191 | |
10170 | 192 static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, |
193 float spec[AT1_SU_SAMPLES]) | |
10157 | 194 { |
195 int bits_used, band_num, bfu_num, i; | |
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196 uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU |
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197 uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU |
10157 | 198 |
199 /* parse the info byte (2nd byte) telling how much BFUs were coded */ | |
200 su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; | |
201 | |
202 /* calc number of consumed bits: | |
203 num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) | |
204 + info_byte_copy(8bits) + log2_block_count_copy(8bits) */ | |
205 bits_used = su->num_bfus * 10 + 32 + | |
206 bfu_amount_tab2[get_bits(gb, 2)] + | |
207 (bfu_amount_tab3[get_bits(gb, 3)] << 1); | |
208 | |
209 /* get word length index (idwl) for each BFU */ | |
10197 | 210 for (i = 0; i < su->num_bfus; i++) |
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211 idwls[i] = get_bits(gb, 4); |
10157 | 212 |
213 /* get scalefactor index (idsf) for each BFU */ | |
10197 | 214 for (i = 0; i < su->num_bfus; i++) |
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215 idsfs[i] = get_bits(gb, 6); |
10157 | 216 |
217 /* zero idwl/idsf for empty BFUs */ | |
218 for (i = su->num_bfus; i < AT1_MAX_BFU; i++) | |
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219 idwls[i] = idsfs[i] = 0; |
10157 | 220 |
221 /* read in the spectral data and reconstruct MDCT spectrum of this channel */ | |
10197 | 222 for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
223 for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { | |
10157 | 224 int pos; |
225 | |
226 int num_specs = specs_per_bfu[bfu_num]; | |
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227 int word_len = !!idwls[bfu_num] + idwls[bfu_num]; |
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228 float scale_factor = sf_table[idsfs[bfu_num]]; |
10217 | 229 bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ |
10157 | 230 |
231 /* check for bitstream overflow */ | |
232 if (bits_used > AT1_SU_MAX_BITS) | |
233 return -1; | |
234 | |
235 /* get the position of the 1st spec according to the block size mode */ | |
236 pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; | |
237 | |
238 if (word_len) { | |
10170 | 239 float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); |
10157 | 240 |
10197 | 241 for (i = 0; i < num_specs; i++) { |
10157 | 242 /* read in a quantized spec and convert it to |
243 * signed int and then inverse quantization | |
244 */ | |
245 spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; | |
246 } | |
247 } else { /* word_len = 0 -> empty BFU, zero all specs in the emty BFU */ | |
10197 | 248 memset(&spec[pos], 0, num_specs * sizeof(float)); |
10157 | 249 } |
250 } | |
251 } | |
252 | |
253 return 0; | |
254 } | |
255 | |
256 | |
257 void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) | |
258 { | |
10197 | 259 float temp[256]; |
260 float iqmf_temp[512 + 46]; | |
10157 | 261 |
262 /* combine low and middle bands */ | |
263 atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); | |
264 | |
265 /* delay the signal of the high band by 23 samples */ | |
10197 | 266 memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 23); |
267 memcpy(&su->last_qmf_delay[23], q->bands[2], sizeof(float) * 256); | |
10157 | 268 |
269 /* combine (low + middle) and high bands */ | |
270 atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); | |
271 } | |
272 | |
273 | |
10170 | 274 static int atrac1_decode_frame(AVCodecContext *avctx, void *data, |
275 int *data_size, AVPacket *avpkt) | |
10157 | 276 { |
277 const uint8_t *buf = avpkt->data; | |
10170 | 278 int buf_size = avpkt->size; |
279 AT1Ctx *q = avctx->priv_data; | |
10157 | 280 int ch, ret, i; |
281 GetBitContext gb; | |
282 float* samples = data; | |
283 | |
284 | |
285 if (buf_size < 212 * q->channels) { | |
286 av_log(q,AV_LOG_ERROR,"Not enought data to decode!\n"); | |
287 return -1; | |
288 } | |
289 | |
10197 | 290 for (ch = 0; ch < q->channels; ch++) { |
10157 | 291 AT1SUCtx* su = &q->SUs[ch]; |
292 | |
10197 | 293 init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
10157 | 294 |
295 /* parse block_size_mode, 1st byte */ | |
10170 | 296 ret = at1_parse_bsm(&gb, su->log2_block_count); |
10157 | 297 if (ret < 0) |
298 return ret; | |
299 | |
300 ret = at1_unpack_dequant(&gb, su, q->spec); | |
301 if (ret < 0) | |
302 return ret; | |
303 | |
304 ret = at1_imdct_block(su, q); | |
305 if (ret < 0) | |
306 return ret; | |
307 at1_subband_synthesis(q, su, q->out_samples[ch]); | |
308 } | |
309 | |
310 /* round, convert to 16bit and interleave */ | |
311 if (q->channels == 1) { | |
312 /* mono */ | |
10197 | 313 q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15), |
314 32700.0 / (1 << 15), AT1_SU_SAMPLES); | |
10157 | 315 } else { |
316 /* stereo */ | |
317 for (i = 0; i < AT1_SU_SAMPLES; i++) { | |
10197 | 318 samples[i * 2] = av_clipf(q->out_samples[0][i], |
319 -32700.0 / (1 << 15), | |
320 32700.0 / (1 << 15)); | |
321 samples[i * 2 + 1] = av_clipf(q->out_samples[1][i], | |
322 -32700.0 / (1 << 15), | |
323 32700.0 / (1 << 15)); | |
10157 | 324 } |
325 } | |
326 | |
327 *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples); | |
328 return avctx->block_align; | |
329 } | |
330 | |
331 | |
332 static av_cold int atrac1_decode_init(AVCodecContext *avctx) | |
333 { | |
334 AT1Ctx *q = avctx->priv_data; | |
335 | |
336 avctx->sample_fmt = SAMPLE_FMT_FLT; | |
337 | |
338 q->channels = avctx->channels; | |
339 | |
340 /* Init the mdct transforms */ | |
10197 | 341 ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15)); |
342 ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15)); | |
343 ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)); | |
10185 | 344 |
345 ff_sine_window_init(short_window, 32); | |
10157 | 346 |
347 atrac_generate_tables(); | |
348 | |
349 dsputil_init(&q->dsp, avctx); | |
350 | |
351 q->bands[0] = q->low; | |
352 q->bands[1] = q->mid; | |
353 q->bands[2] = q->high; | |
354 | |
355 /* Prepare the mdct overlap buffers */ | |
356 q->SUs[0].spectrum[0] = q->SUs[0].spec1; | |
357 q->SUs[0].spectrum[1] = q->SUs[0].spec2; | |
358 q->SUs[1].spectrum[0] = q->SUs[1].spec1; | |
359 q->SUs[1].spectrum[1] = q->SUs[1].spec2; | |
360 | |
361 return 0; | |
362 } | |
363 | |
10218 | 364 |
365 static av_cold int atrac1_decode_end(AVCodecContext * avctx) { | |
366 AT1Ctx *q = avctx->priv_data; | |
367 | |
368 ff_mdct_end(&q->mdct_ctx[0]); | |
369 ff_mdct_end(&q->mdct_ctx[1]); | |
370 ff_mdct_end(&q->mdct_ctx[2]); | |
371 return 0; | |
372 } | |
373 | |
374 | |
10157 | 375 AVCodec atrac1_decoder = { |
376 .name = "atrac1", | |
377 .type = CODEC_TYPE_AUDIO, | |
378 .id = CODEC_ID_ATRAC1, | |
379 .priv_data_size = sizeof(AT1Ctx), | |
380 .init = atrac1_decode_init, | |
10218 | 381 .close = atrac1_decode_end, |
10157 | 382 .decode = atrac1_decode_frame, |
383 .long_name = NULL_IF_CONFIG_SMALL("Atrac 1 (Adaptive TRansform Acoustic Coding)"), | |
384 }; |