Mercurial > libavcodec.hg
annotate dca.c @ 7904:47717d5239fa libavcodec
AltiVec detection support for OpenBSD, patch by Brad, brad comstyle com.
author | diego |
---|---|
date | Mon, 22 Sep 2008 05:38:50 +0000 |
parents | 47c118619225 |
children | 8ce423998cca |
rev | line source |
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4599 | 1 /* |
2 * DCA compatible decoder | |
3 * Copyright (C) 2004 Gildas Bazin | |
4 * Copyright (C) 2004 Benjamin Zores | |
5 * Copyright (C) 2006 Benjamin Larsson | |
6 * Copyright (C) 2007 Konstantin Shishkov | |
7 * | |
8 * This file is part of FFmpeg. | |
9 * | |
10 * FFmpeg is free software; you can redistribute it and/or | |
11 * modify it under the terms of the GNU Lesser General Public | |
12 * License as published by the Free Software Foundation; either | |
13 * version 2.1 of the License, or (at your option) any later version. | |
14 * | |
15 * FFmpeg is distributed in the hope that it will be useful, | |
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
18 * Lesser General Public License for more details. | |
19 * | |
20 * You should have received a copy of the GNU Lesser General Public | |
21 * License along with FFmpeg; if not, write to the Free Software | |
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
23 */ | |
24 | |
25 /** | |
26 * @file dca.c | |
27 */ | |
28 | |
29 #include <math.h> | |
30 #include <stddef.h> | |
31 #include <stdio.h> | |
32 | |
33 #include "avcodec.h" | |
34 #include "dsputil.h" | |
35 #include "bitstream.h" | |
36 #include "dcadata.h" | |
37 #include "dcahuff.h" | |
4899 | 38 #include "dca.h" |
4599 | 39 |
40 //#define TRACE | |
41 | |
42 #define DCA_PRIM_CHANNELS_MAX (5) | |
43 #define DCA_SUBBANDS (32) | |
44 #define DCA_ABITS_MAX (32) /* Should be 28 */ | |
45 #define DCA_SUBSUBFAMES_MAX (4) | |
46 #define DCA_LFE_MAX (3) | |
47 | |
48 enum DCAMode { | |
49 DCA_MONO = 0, | |
50 DCA_CHANNEL, | |
51 DCA_STEREO, | |
52 DCA_STEREO_SUMDIFF, | |
53 DCA_STEREO_TOTAL, | |
54 DCA_3F, | |
55 DCA_2F1R, | |
56 DCA_3F1R, | |
57 DCA_2F2R, | |
58 DCA_3F2R, | |
59 DCA_4F2R | |
60 }; | |
61 | |
62 #define DCA_DOLBY 101 /* FIXME */ | |
63 | |
64 #define DCA_CHANNEL_BITS 6 | |
65 #define DCA_CHANNEL_MASK 0x3F | |
66 | |
67 #define DCA_LFE 0x80 | |
68 | |
69 #define HEADER_SIZE 14 | |
70 #define CONVERT_BIAS 384 | |
71 | |
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72 #define DCA_MAX_FRAME_SIZE 16384 |
4599 | 73 |
74 /** Bit allocation */ | |
75 typedef struct { | |
76 int offset; ///< code values offset | |
77 int maxbits[8]; ///< max bits in VLC | |
78 int wrap; ///< wrap for get_vlc2() | |
79 VLC vlc[8]; ///< actual codes | |
80 } BitAlloc; | |
81 | |
82 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select | |
83 static BitAlloc dca_tmode; ///< transition mode VLCs | |
84 static BitAlloc dca_scalefactor; ///< scalefactor VLCs | |
85 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs | |
86 | |
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87 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx) |
4599 | 88 { |
89 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset; | |
90 } | |
91 | |
92 typedef struct { | |
93 AVCodecContext *avctx; | |
94 /* Frame header */ | |
95 int frame_type; ///< type of the current frame | |
96 int samples_deficit; ///< deficit sample count | |
97 int crc_present; ///< crc is present in the bitstream | |
98 int sample_blocks; ///< number of PCM sample blocks | |
99 int frame_size; ///< primary frame byte size | |
100 int amode; ///< audio channels arrangement | |
101 int sample_rate; ///< audio sampling rate | |
102 int bit_rate; ///< transmission bit rate | |
103 | |
104 int downmix; ///< embedded downmix enabled | |
105 int dynrange; ///< embedded dynamic range flag | |
106 int timestamp; ///< embedded time stamp flag | |
107 int aux_data; ///< auxiliary data flag | |
108 int hdcd; ///< source material is mastered in HDCD | |
109 int ext_descr; ///< extension audio descriptor flag | |
110 int ext_coding; ///< extended coding flag | |
111 int aspf; ///< audio sync word insertion flag | |
112 int lfe; ///< low frequency effects flag | |
113 int predictor_history; ///< predictor history flag | |
114 int header_crc; ///< header crc check bytes | |
115 int multirate_inter; ///< multirate interpolator switch | |
116 int version; ///< encoder software revision | |
117 int copy_history; ///< copy history | |
118 int source_pcm_res; ///< source pcm resolution | |
119 int front_sum; ///< front sum/difference flag | |
120 int surround_sum; ///< surround sum/difference flag | |
121 int dialog_norm; ///< dialog normalisation parameter | |
122 | |
123 /* Primary audio coding header */ | |
124 int subframes; ///< number of subframes | |
6463 | 125 int total_channels; ///< number of channels including extensions |
4599 | 126 int prim_channels; ///< number of primary audio channels |
127 int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count | |
128 int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband | |
129 int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index | |
130 int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book | |
131 int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book | |
132 int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select | |
133 int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select | |
134 float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment | |
135 | |
136 /* Primary audio coding side information */ | |
137 int subsubframes; ///< number of subsubframes | |
138 int partial_samples; ///< partial subsubframe samples count | |
139 int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not) | |
140 int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs | |
141 int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index | |
142 int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients) | |
143 int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient) | |
144 int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook | |
145 int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors | |
146 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients | |
147 int dynrange_coef; ///< dynamic range coefficient | |
148 | |
149 int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands | |
150 | |
151 float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX * | |
152 2 /*history */ ]; ///< Low frequency effect data | |
153 int lfe_scale_factor; | |
154 | |
155 /* Subband samples history (for ADPCM) */ | |
156 float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; | |
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157 DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]); |
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158 float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32]; |
7737 | 159 int hist_index[DCA_PRIM_CHANNELS_MAX]; |
4599 | 160 |
161 int output; ///< type of output | |
162 int bias; ///< output bias | |
163 | |
164 DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */ | |
7726 | 165 const float *samples_chanptr[6]; |
4599 | 166 |
167 uint8_t dca_buffer[DCA_MAX_FRAME_SIZE]; | |
168 int dca_buffer_size; ///< how much data is in the dca_buffer | |
169 | |
170 GetBitContext gb; | |
171 /* Current position in DCA frame */ | |
172 int current_subframe; | |
173 int current_subsubframe; | |
174 | |
175 int debug_flag; ///< used for suppressing repeated error messages output | |
176 DSPContext dsp; | |
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177 MDCTContext imdct; |
4599 | 178 } DCAContext; |
179 | |
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180 static av_cold void dca_init_vlcs(void) |
4599 | 181 { |
6350 | 182 static int vlcs_initialized = 0; |
4599 | 183 int i, j; |
184 | |
6350 | 185 if (vlcs_initialized) |
4599 | 186 return; |
187 | |
188 dca_bitalloc_index.offset = 1; | |
5070 | 189 dca_bitalloc_index.wrap = 2; |
4599 | 190 for (i = 0; i < 5; i++) |
191 init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12, | |
192 bitalloc_12_bits[i], 1, 1, | |
193 bitalloc_12_codes[i], 2, 2, 1); | |
194 dca_scalefactor.offset = -64; | |
195 dca_scalefactor.wrap = 2; | |
196 for (i = 0; i < 5; i++) | |
197 init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129, | |
198 scales_bits[i], 1, 1, | |
199 scales_codes[i], 2, 2, 1); | |
200 dca_tmode.offset = 0; | |
201 dca_tmode.wrap = 1; | |
202 for (i = 0; i < 4; i++) | |
203 init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4, | |
204 tmode_bits[i], 1, 1, | |
205 tmode_codes[i], 2, 2, 1); | |
206 | |
207 for(i = 0; i < 10; i++) | |
208 for(j = 0; j < 7; j++){ | |
209 if(!bitalloc_codes[i][j]) break; | |
210 dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i]; | |
211 dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4); | |
212 init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j], | |
213 bitalloc_sizes[i], | |
214 bitalloc_bits[i][j], 1, 1, | |
215 bitalloc_codes[i][j], 2, 2, 1); | |
216 } | |
6350 | 217 vlcs_initialized = 1; |
4599 | 218 } |
219 | |
220 static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) | |
221 { | |
222 while(len--) | |
223 *dst++ = get_bits(gb, bits); | |
224 } | |
225 | |
226 static int dca_parse_frame_header(DCAContext * s) | |
227 { | |
228 int i, j; | |
229 static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; | |
230 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; | |
231 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; | |
232 | |
233 s->bias = CONVERT_BIAS; | |
234 | |
235 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
236 | |
237 /* Sync code */ | |
238 get_bits(&s->gb, 32); | |
239 | |
240 /* Frame header */ | |
241 s->frame_type = get_bits(&s->gb, 1); | |
242 s->samples_deficit = get_bits(&s->gb, 5) + 1; | |
243 s->crc_present = get_bits(&s->gb, 1); | |
244 s->sample_blocks = get_bits(&s->gb, 7) + 1; | |
245 s->frame_size = get_bits(&s->gb, 14) + 1; | |
246 if (s->frame_size < 95) | |
247 return -1; | |
248 s->amode = get_bits(&s->gb, 6); | |
249 s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)]; | |
250 if (!s->sample_rate) | |
251 return -1; | |
252 s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)]; | |
253 if (!s->bit_rate) | |
254 return -1; | |
255 | |
256 s->downmix = get_bits(&s->gb, 1); | |
257 s->dynrange = get_bits(&s->gb, 1); | |
258 s->timestamp = get_bits(&s->gb, 1); | |
259 s->aux_data = get_bits(&s->gb, 1); | |
260 s->hdcd = get_bits(&s->gb, 1); | |
261 s->ext_descr = get_bits(&s->gb, 3); | |
262 s->ext_coding = get_bits(&s->gb, 1); | |
263 s->aspf = get_bits(&s->gb, 1); | |
264 s->lfe = get_bits(&s->gb, 2); | |
265 s->predictor_history = get_bits(&s->gb, 1); | |
266 | |
267 /* TODO: check CRC */ | |
268 if (s->crc_present) | |
269 s->header_crc = get_bits(&s->gb, 16); | |
270 | |
271 s->multirate_inter = get_bits(&s->gb, 1); | |
272 s->version = get_bits(&s->gb, 4); | |
273 s->copy_history = get_bits(&s->gb, 2); | |
274 s->source_pcm_res = get_bits(&s->gb, 3); | |
275 s->front_sum = get_bits(&s->gb, 1); | |
276 s->surround_sum = get_bits(&s->gb, 1); | |
277 s->dialog_norm = get_bits(&s->gb, 4); | |
278 | |
279 /* FIXME: channels mixing levels */ | |
4893 | 280 s->output = s->amode; |
281 if(s->lfe) s->output |= DCA_LFE; | |
4599 | 282 |
283 #ifdef TRACE | |
284 av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); | |
285 av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit); | |
286 av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present); | |
287 av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n", | |
288 s->sample_blocks, s->sample_blocks * 32); | |
289 av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size); | |
290 av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n", | |
291 s->amode, dca_channels[s->amode]); | |
292 av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i (%i Hz)\n", | |
293 s->sample_rate, dca_sample_rates[s->sample_rate]); | |
294 av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i (%i bits/s)\n", | |
295 s->bit_rate, dca_bit_rates[s->bit_rate]); | |
296 av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix); | |
297 av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange); | |
298 av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp); | |
299 av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data); | |
300 av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd); | |
301 av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr); | |
302 av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding); | |
303 av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf); | |
304 av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe); | |
305 av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n", | |
306 s->predictor_history); | |
307 av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc); | |
308 av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n", | |
309 s->multirate_inter); | |
310 av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version); | |
311 av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history); | |
312 av_log(s->avctx, AV_LOG_DEBUG, | |
313 "source pcm resolution: %i (%i bits/sample)\n", | |
314 s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]); | |
315 av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum); | |
316 av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum); | |
317 av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm); | |
318 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
319 #endif | |
320 | |
321 /* Primary audio coding header */ | |
322 s->subframes = get_bits(&s->gb, 4) + 1; | |
6463 | 323 s->total_channels = get_bits(&s->gb, 3) + 1; |
324 s->prim_channels = s->total_channels; | |
325 if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) | |
326 s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */ | |
4599 | 327 |
328 | |
329 for (i = 0; i < s->prim_channels; i++) { | |
330 s->subband_activity[i] = get_bits(&s->gb, 5) + 2; | |
331 if (s->subband_activity[i] > DCA_SUBBANDS) | |
332 s->subband_activity[i] = DCA_SUBBANDS; | |
333 } | |
334 for (i = 0; i < s->prim_channels; i++) { | |
335 s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; | |
336 if (s->vq_start_subband[i] > DCA_SUBBANDS) | |
337 s->vq_start_subband[i] = DCA_SUBBANDS; | |
338 } | |
339 get_array(&s->gb, s->joint_intensity, s->prim_channels, 3); | |
340 get_array(&s->gb, s->transient_huffman, s->prim_channels, 2); | |
341 get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3); | |
342 get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3); | |
343 | |
344 /* Get codebooks quantization indexes */ | |
345 memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); | |
346 for (j = 1; j < 11; j++) | |
347 for (i = 0; i < s->prim_channels; i++) | |
348 s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); | |
349 | |
350 /* Get scale factor adjustment */ | |
351 for (j = 0; j < 11; j++) | |
352 for (i = 0; i < s->prim_channels; i++) | |
353 s->scalefactor_adj[i][j] = 1; | |
354 | |
355 for (j = 1; j < 11; j++) | |
356 for (i = 0; i < s->prim_channels; i++) | |
357 if (s->quant_index_huffman[i][j] < thr[j]) | |
358 s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; | |
359 | |
360 if (s->crc_present) { | |
361 /* Audio header CRC check */ | |
362 get_bits(&s->gb, 16); | |
363 } | |
364 | |
365 s->current_subframe = 0; | |
366 s->current_subsubframe = 0; | |
367 | |
368 #ifdef TRACE | |
369 av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes); | |
370 av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels); | |
371 for(i = 0; i < s->prim_channels; i++){ | |
372 av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]); | |
373 av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]); | |
374 av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]); | |
375 av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]); | |
376 av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]); | |
377 av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]); | |
378 av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:"); | |
379 for (j = 0; j < 11; j++) | |
380 av_log(s->avctx, AV_LOG_DEBUG, " %i", | |
381 s->quant_index_huffman[i][j]); | |
382 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
383 av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:"); | |
384 for (j = 0; j < 11; j++) | |
385 av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]); | |
386 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
387 } | |
388 #endif | |
389 | |
390 return 0; | |
391 } | |
392 | |
393 | |
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394 static inline int get_scale(GetBitContext *gb, int level, int value) |
4599 | 395 { |
396 if (level < 5) { | |
397 /* huffman encoded */ | |
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398 value += get_bitalloc(gb, &dca_scalefactor, level); |
4599 | 399 } else if(level < 8) |
400 value = get_bits(gb, level + 1); | |
401 return value; | |
402 } | |
403 | |
404 static int dca_subframe_header(DCAContext * s) | |
405 { | |
406 /* Primary audio coding side information */ | |
407 int j, k; | |
408 | |
409 s->subsubframes = get_bits(&s->gb, 2) + 1; | |
410 s->partial_samples = get_bits(&s->gb, 3); | |
411 for (j = 0; j < s->prim_channels; j++) { | |
412 for (k = 0; k < s->subband_activity[j]; k++) | |
413 s->prediction_mode[j][k] = get_bits(&s->gb, 1); | |
414 } | |
415 | |
416 /* Get prediction codebook */ | |
417 for (j = 0; j < s->prim_channels; j++) { | |
418 for (k = 0; k < s->subband_activity[j]; k++) { | |
419 if (s->prediction_mode[j][k] > 0) { | |
420 /* (Prediction coefficient VQ address) */ | |
421 s->prediction_vq[j][k] = get_bits(&s->gb, 12); | |
422 } | |
423 } | |
424 } | |
425 | |
426 /* Bit allocation index */ | |
427 for (j = 0; j < s->prim_channels; j++) { | |
428 for (k = 0; k < s->vq_start_subband[j]; k++) { | |
429 if (s->bitalloc_huffman[j] == 6) | |
430 s->bitalloc[j][k] = get_bits(&s->gb, 5); | |
431 else if (s->bitalloc_huffman[j] == 5) | |
432 s->bitalloc[j][k] = get_bits(&s->gb, 4); | |
6463 | 433 else if (s->bitalloc_huffman[j] == 7) { |
434 av_log(s->avctx, AV_LOG_ERROR, | |
435 "Invalid bit allocation index\n"); | |
436 return -1; | |
437 } else { | |
4599 | 438 s->bitalloc[j][k] = |
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439 get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); |
4599 | 440 } |
441 | |
442 if (s->bitalloc[j][k] > 26) { | |
443 // av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n", | |
444 // j, k, s->bitalloc[j][k]); | |
445 return -1; | |
446 } | |
447 } | |
448 } | |
449 | |
450 /* Transition mode */ | |
451 for (j = 0; j < s->prim_channels; j++) { | |
452 for (k = 0; k < s->subband_activity[j]; k++) { | |
453 s->transition_mode[j][k] = 0; | |
454 if (s->subsubframes > 1 && | |
455 k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { | |
456 s->transition_mode[j][k] = | |
457 get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); | |
458 } | |
459 } | |
460 } | |
461 | |
462 for (j = 0; j < s->prim_channels; j++) { | |
6214 | 463 const uint32_t *scale_table; |
4599 | 464 int scale_sum; |
465 | |
466 memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); | |
467 | |
468 if (s->scalefactor_huffman[j] == 6) | |
6214 | 469 scale_table = scale_factor_quant7; |
4599 | 470 else |
6214 | 471 scale_table = scale_factor_quant6; |
4599 | 472 |
473 /* When huffman coded, only the difference is encoded */ | |
474 scale_sum = 0; | |
475 | |
476 for (k = 0; k < s->subband_activity[j]; k++) { | |
477 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { | |
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478 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 479 s->scale_factor[j][k][0] = scale_table[scale_sum]; |
480 } | |
481 | |
482 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { | |
483 /* Get second scale factor */ | |
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484 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 485 s->scale_factor[j][k][1] = scale_table[scale_sum]; |
486 } | |
487 } | |
488 } | |
489 | |
490 /* Joint subband scale factor codebook select */ | |
491 for (j = 0; j < s->prim_channels; j++) { | |
492 /* Transmitted only if joint subband coding enabled */ | |
493 if (s->joint_intensity[j] > 0) | |
494 s->joint_huff[j] = get_bits(&s->gb, 3); | |
495 } | |
496 | |
497 /* Scale factors for joint subband coding */ | |
498 for (j = 0; j < s->prim_channels; j++) { | |
499 int source_channel; | |
500 | |
501 /* Transmitted only if joint subband coding enabled */ | |
502 if (s->joint_intensity[j] > 0) { | |
503 int scale = 0; | |
504 source_channel = s->joint_intensity[j] - 1; | |
505 | |
506 /* When huffman coded, only the difference is encoded | |
507 * (is this valid as well for joint scales ???) */ | |
508 | |
509 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { | |
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510 scale = get_scale(&s->gb, s->joint_huff[j], 0); |
4599 | 511 scale += 64; /* bias */ |
512 s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ | |
513 } | |
514 | |
515 if (!s->debug_flag & 0x02) { | |
516 av_log(s->avctx, AV_LOG_DEBUG, | |
517 "Joint stereo coding not supported\n"); | |
518 s->debug_flag |= 0x02; | |
519 } | |
520 } | |
521 } | |
522 | |
523 /* Stereo downmix coefficients */ | |
4894 | 524 if (s->prim_channels > 2) { |
525 if(s->downmix) { | |
4895 | 526 for (j = 0; j < s->prim_channels; j++) { |
527 s->downmix_coef[j][0] = get_bits(&s->gb, 7); | |
528 s->downmix_coef[j][1] = get_bits(&s->gb, 7); | |
529 } | |
4894 | 530 } else { |
531 int am = s->amode & DCA_CHANNEL_MASK; | |
532 for (j = 0; j < s->prim_channels; j++) { | |
533 s->downmix_coef[j][0] = dca_default_coeffs[am][j][0]; | |
534 s->downmix_coef[j][1] = dca_default_coeffs[am][j][1]; | |
535 } | |
536 } | |
4599 | 537 } |
538 | |
539 /* Dynamic range coefficient */ | |
540 if (s->dynrange) | |
541 s->dynrange_coef = get_bits(&s->gb, 8); | |
542 | |
543 /* Side information CRC check word */ | |
544 if (s->crc_present) { | |
545 get_bits(&s->gb, 16); | |
546 } | |
547 | |
548 /* | |
549 * Primary audio data arrays | |
550 */ | |
551 | |
552 /* VQ encoded high frequency subbands */ | |
553 for (j = 0; j < s->prim_channels; j++) | |
554 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
555 /* 1 vector -> 32 samples */ | |
556 s->high_freq_vq[j][k] = get_bits(&s->gb, 10); | |
557 | |
558 /* Low frequency effect data */ | |
559 if (s->lfe) { | |
560 /* LFE samples */ | |
561 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
562 float lfe_scale; | |
563 | |
564 for (j = lfe_samples; j < lfe_samples * 2; j++) { | |
565 /* Signed 8 bits int */ | |
566 s->lfe_data[j] = get_sbits(&s->gb, 8); | |
567 } | |
568 | |
569 /* Scale factor index */ | |
570 s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)]; | |
571 | |
572 /* Quantization step size * scale factor */ | |
573 lfe_scale = 0.035 * s->lfe_scale_factor; | |
574 | |
575 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
576 s->lfe_data[j] *= lfe_scale; | |
577 } | |
578 | |
579 #ifdef TRACE | |
580 av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes); | |
581 av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", | |
582 s->partial_samples); | |
583 for (j = 0; j < s->prim_channels; j++) { | |
584 av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:"); | |
585 for (k = 0; k < s->subband_activity[j]; k++) | |
586 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]); | |
587 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
588 } | |
589 for (j = 0; j < s->prim_channels; j++) { | |
590 for (k = 0; k < s->subband_activity[j]; k++) | |
591 av_log(s->avctx, AV_LOG_DEBUG, | |
592 "prediction coefs: %f, %f, %f, %f\n", | |
593 (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192, | |
594 (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192, | |
595 (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192, | |
596 (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192); | |
597 } | |
598 for (j = 0; j < s->prim_channels; j++) { | |
599 av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: "); | |
600 for (k = 0; k < s->vq_start_subband[j]; k++) | |
601 av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]); | |
602 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
603 } | |
604 for (j = 0; j < s->prim_channels; j++) { | |
605 av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:"); | |
606 for (k = 0; k < s->subband_activity[j]; k++) | |
607 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]); | |
608 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
609 } | |
610 for (j = 0; j < s->prim_channels; j++) { | |
611 av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:"); | |
612 for (k = 0; k < s->subband_activity[j]; k++) { | |
613 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) | |
614 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]); | |
615 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) | |
616 av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]); | |
617 } | |
618 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
619 } | |
620 for (j = 0; j < s->prim_channels; j++) { | |
621 if (s->joint_intensity[j] > 0) { | |
5069 | 622 int source_channel = s->joint_intensity[j] - 1; |
4599 | 623 av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n"); |
624 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) | |
625 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]); | |
626 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
627 } | |
628 } | |
629 if (s->prim_channels > 2 && s->downmix) { | |
630 av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); | |
631 for (j = 0; j < s->prim_channels; j++) { | |
632 av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]); | |
633 av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]); | |
634 } | |
635 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
636 } | |
637 for (j = 0; j < s->prim_channels; j++) | |
638 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
639 av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]); | |
640 if(s->lfe){ | |
5069 | 641 int lfe_samples = 2 * s->lfe * s->subsubframes; |
4599 | 642 av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n"); |
643 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
644 av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]); | |
645 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
646 } | |
647 #endif | |
648 | |
649 return 0; | |
650 } | |
651 | |
652 static void qmf_32_subbands(DCAContext * s, int chans, | |
653 float samples_in[32][8], float *samples_out, | |
654 float scale, float bias) | |
655 { | |
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656 const float *prCoeff; |
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657 int i, j; |
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658 DECLARE_ALIGNED_16(float, raXin[32]); |
4599 | 659 |
7737 | 660 int hist_index= s->hist_index[chans]; |
4599 | 661 float *subband_fir_hist2 = s->subband_fir_noidea[chans]; |
662 | |
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663 int subindex; |
4599 | 664 |
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665 scale *= sqrt(1/8.0); |
4599 | 666 |
667 /* Select filter */ | |
668 if (!s->multirate_inter) /* Non-perfect reconstruction */ | |
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669 prCoeff = fir_32bands_nonperfect; |
4599 | 670 else /* Perfect reconstruction */ |
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671 prCoeff = fir_32bands_perfect; |
4599 | 672 |
673 /* Reconstructed channel sample index */ | |
674 for (subindex = 0; subindex < 8; subindex++) { | |
7737 | 675 float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index; |
4599 | 676 /* Load in one sample from each subband and clear inactive subbands */ |
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677 for (i = 0; i < s->subband_activity[chans]; i++){ |
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678 if((i-1)&2) raXin[i] = -samples_in[i][subindex]; |
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679 else raXin[i] = samples_in[i][subindex]; |
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680 } |
4599 | 681 for (; i < 32; i++) |
682 raXin[i] = 0.0; | |
683 | |
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684 ff_imdct_half(&s->imdct, subband_fir_hist, raXin); |
4599 | 685 |
686 /* Multiply by filter coefficients */ | |
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687 for (i = 0; i < 16; i++){ |
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688 float a= subband_fir_hist2[i ]; |
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689 float b= subband_fir_hist2[i+16]; |
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690 float c= 0; |
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691 float d= 0; |
7737 | 692 for (j = 0; j < 512-hist_index; j += 64){ |
7738
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693 a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]); |
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694 b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]); |
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695 c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]); |
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696 d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]); |
4599 | 697 } |
7737 | 698 for ( ; j < 512; j += 64){ |
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699 a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]); |
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700 b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]); |
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701 c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]); |
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702 d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]); |
7737 | 703 } |
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704 samples_out[i ] = a * scale + bias; |
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705 samples_out[i+16] = b * scale + bias; |
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706 subband_fir_hist2[i ] = c; |
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707 subband_fir_hist2[i+16] = d; |
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708 } |
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709 samples_out+= 32; |
4599 | 710 |
7737 | 711 hist_index = (hist_index-32)&511; |
4599 | 712 } |
7737 | 713 s->hist_index[chans]= hist_index; |
4599 | 714 } |
715 | |
716 static void lfe_interpolation_fir(int decimation_select, | |
717 int num_deci_sample, float *samples_in, | |
718 float *samples_out, float scale, | |
719 float bias) | |
720 { | |
721 /* samples_in: An array holding decimated samples. | |
722 * Samples in current subframe starts from samples_in[0], | |
723 * while samples_in[-1], samples_in[-2], ..., stores samples | |
724 * from last subframe as history. | |
725 * | |
726 * samples_out: An array holding interpolated samples | |
727 */ | |
728 | |
729 int decifactor, k, j; | |
730 const float *prCoeff; | |
731 | |
732 int interp_index = 0; /* Index to the interpolated samples */ | |
733 int deciindex; | |
734 | |
735 /* Select decimation filter */ | |
736 if (decimation_select == 1) { | |
737 decifactor = 128; | |
738 prCoeff = lfe_fir_128; | |
739 } else { | |
740 decifactor = 64; | |
741 prCoeff = lfe_fir_64; | |
742 } | |
743 /* Interpolation */ | |
744 for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { | |
745 /* One decimated sample generates decifactor interpolated ones */ | |
746 for (k = 0; k < decifactor; k++) { | |
747 float rTmp = 0.0; | |
748 //FIXME the coeffs are symetric, fix that | |
749 for (j = 0; j < 512 / decifactor; j++) | |
750 rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor]; | |
751 samples_out[interp_index++] = rTmp / scale + bias; | |
752 } | |
753 } | |
754 } | |
755 | |
756 /* downmixing routines */ | |
4894 | 757 #define MIX_REAR1(samples, si1, rs, coef) \ |
758 samples[i] += samples[si1] * coef[rs][0]; \ | |
759 samples[i+256] += samples[si1] * coef[rs][1]; | |
4599 | 760 |
4894 | 761 #define MIX_REAR2(samples, si1, si2, rs, coef) \ |
762 samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ | |
763 samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; | |
4599 | 764 |
4894 | 765 #define MIX_FRONT3(samples, coef) \ |
4599 | 766 t = samples[i]; \ |
4894 | 767 samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \ |
768 samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1]; | |
4599 | 769 |
770 #define DOWNMIX_TO_STEREO(op1, op2) \ | |
771 for(i = 0; i < 256; i++){ \ | |
772 op1 \ | |
773 op2 \ | |
774 } | |
775 | |
4894 | 776 static void dca_downmix(float *samples, int srcfmt, |
777 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]) | |
4599 | 778 { |
779 int i; | |
780 float t; | |
4894 | 781 float coef[DCA_PRIM_CHANNELS_MAX][2]; |
782 | |
783 for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) { | |
784 coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]]; | |
785 coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]]; | |
786 } | |
4599 | 787 |
788 switch (srcfmt) { | |
789 case DCA_MONO: | |
790 case DCA_CHANNEL: | |
791 case DCA_STEREO_TOTAL: | |
792 case DCA_STEREO_SUMDIFF: | |
793 case DCA_4F2R: | |
794 av_log(NULL, 0, "Not implemented!\n"); | |
795 break; | |
796 case DCA_STEREO: | |
797 break; | |
798 case DCA_3F: | |
4894 | 799 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),); |
4599 | 800 break; |
801 case DCA_2F1R: | |
4894 | 802 DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),); |
4599 | 803 break; |
804 case DCA_3F1R: | |
4894 | 805 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
806 MIX_REAR1(samples, i + 768, 3, coef)); | |
4599 | 807 break; |
808 case DCA_2F2R: | |
4894 | 809 DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),); |
4599 | 810 break; |
811 case DCA_3F2R: | |
4894 | 812 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
813 MIX_REAR2(samples, i + 768, i + 1024, 3, coef)); | |
4599 | 814 break; |
815 } | |
816 } | |
817 | |
818 | |
819 /* Very compact version of the block code decoder that does not use table | |
820 * look-up but is slightly slower */ | |
821 static int decode_blockcode(int code, int levels, int *values) | |
822 { | |
823 int i; | |
824 int offset = (levels - 1) >> 1; | |
825 | |
826 for (i = 0; i < 4; i++) { | |
827 values[i] = (code % levels) - offset; | |
828 code /= levels; | |
829 } | |
830 | |
831 if (code == 0) | |
832 return 0; | |
833 else { | |
834 av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n"); | |
835 return -1; | |
836 } | |
837 } | |
838 | |
839 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; | |
840 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; | |
841 | |
842 static int dca_subsubframe(DCAContext * s) | |
843 { | |
844 int k, l; | |
845 int subsubframe = s->current_subsubframe; | |
846 | |
6214 | 847 const float *quant_step_table; |
4599 | 848 |
849 /* FIXME */ | |
850 float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; | |
851 | |
852 /* | |
853 * Audio data | |
854 */ | |
855 | |
856 /* Select quantization step size table */ | |
857 if (s->bit_rate == 0x1f) | |
6214 | 858 quant_step_table = lossless_quant_d; |
4599 | 859 else |
6214 | 860 quant_step_table = lossy_quant_d; |
4599 | 861 |
862 for (k = 0; k < s->prim_channels; k++) { | |
863 for (l = 0; l < s->vq_start_subband[k]; l++) { | |
864 int m; | |
865 | |
866 /* Select the mid-tread linear quantizer */ | |
867 int abits = s->bitalloc[k][l]; | |
868 | |
869 float quant_step_size = quant_step_table[abits]; | |
870 float rscale; | |
871 | |
872 /* | |
873 * Determine quantization index code book and its type | |
874 */ | |
875 | |
876 /* Select quantization index code book */ | |
877 int sel = s->quant_index_huffman[k][abits]; | |
878 | |
879 /* | |
880 * Extract bits from the bit stream | |
881 */ | |
882 if(!abits){ | |
883 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0])); | |
884 }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){ | |
885 if(abits <= 7){ | |
886 /* Block code */ | |
887 int block_code1, block_code2, size, levels; | |
888 int block[8]; | |
889 | |
890 size = abits_sizes[abits-1]; | |
891 levels = abits_levels[abits-1]; | |
892 | |
893 block_code1 = get_bits(&s->gb, size); | |
894 /* FIXME Should test return value */ | |
895 decode_blockcode(block_code1, levels, block); | |
896 block_code2 = get_bits(&s->gb, size); | |
897 decode_blockcode(block_code2, levels, &block[4]); | |
898 for (m = 0; m < 8; m++) | |
899 subband_samples[k][l][m] = block[m]; | |
900 }else{ | |
901 /* no coding */ | |
902 for (m = 0; m < 8; m++) | |
903 subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3); | |
904 } | |
905 }else{ | |
906 /* Huffman coded */ | |
907 for (m = 0; m < 8; m++) | |
908 subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); | |
909 } | |
910 | |
911 /* Deal with transients */ | |
912 if (s->transition_mode[k][l] && | |
913 subsubframe >= s->transition_mode[k][l]) | |
914 rscale = quant_step_size * s->scale_factor[k][l][1]; | |
915 else | |
916 rscale = quant_step_size * s->scale_factor[k][l][0]; | |
917 | |
918 rscale *= s->scalefactor_adj[k][sel]; | |
919 | |
920 for (m = 0; m < 8; m++) | |
921 subband_samples[k][l][m] *= rscale; | |
922 | |
923 /* | |
924 * Inverse ADPCM if in prediction mode | |
925 */ | |
926 if (s->prediction_mode[k][l]) { | |
927 int n; | |
928 for (m = 0; m < 8; m++) { | |
929 for (n = 1; n <= 4; n++) | |
930 if (m >= n) | |
931 subband_samples[k][l][m] += | |
932 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
933 subband_samples[k][l][m - n] / 8192); | |
934 else if (s->predictor_history) | |
935 subband_samples[k][l][m] += | |
936 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
937 s->subband_samples_hist[k][l][m - n + | |
938 4] / 8192); | |
939 } | |
940 } | |
941 } | |
942 | |
943 /* | |
944 * Decode VQ encoded high frequencies | |
945 */ | |
946 for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { | |
947 /* 1 vector -> 32 samples but we only need the 8 samples | |
948 * for this subsubframe. */ | |
949 int m; | |
950 | |
951 if (!s->debug_flag & 0x01) { | |
952 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); | |
953 s->debug_flag |= 0x01; | |
954 } | |
955 | |
956 for (m = 0; m < 8; m++) { | |
957 subband_samples[k][l][m] = | |
958 high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 + | |
959 m] | |
960 * (float) s->scale_factor[k][l][0] / 16.0; | |
961 } | |
962 } | |
963 } | |
964 | |
965 /* Check for DSYNC after subsubframe */ | |
966 if (s->aspf || subsubframe == s->subsubframes - 1) { | |
967 if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */ | |
968 #ifdef TRACE | |
969 av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n"); | |
970 #endif | |
971 } else { | |
972 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); | |
973 } | |
974 } | |
975 | |
976 /* Backup predictor history for adpcm */ | |
977 for (k = 0; k < s->prim_channels; k++) | |
978 for (l = 0; l < s->vq_start_subband[k]; l++) | |
979 memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4], | |
980 4 * sizeof(subband_samples[0][0][0])); | |
981 | |
982 /* 32 subbands QMF */ | |
983 for (k = 0; k < s->prim_channels; k++) { | |
984 /* static float pcm_to_double[8] = | |
985 {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/ | |
986 qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k], | |
7680 | 987 M_SQRT1_2 /*pcm_to_double[s->source_pcm_res] */ , |
4599 | 988 0 /*s->bias */ ); |
989 } | |
990 | |
991 /* Down mixing */ | |
992 | |
993 if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) { | |
4894 | 994 dca_downmix(s->samples, s->amode, s->downmix_coef); |
4599 | 995 } |
996 | |
997 /* Generate LFE samples for this subsubframe FIXME!!! */ | |
998 if (s->output & DCA_LFE) { | |
999 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
1000 int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK]; | |
1001 | |
1002 lfe_interpolation_fir(s->lfe, 2 * s->lfe, | |
1003 s->lfe_data + lfe_samples + | |
1004 2 * s->lfe * subsubframe, | |
1005 &s->samples[256 * i_channels], | |
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1006 256.0, 0 /* s->bias */); |
4599 | 1007 /* Outputs 20bits pcm samples */ |
1008 } | |
1009 | |
1010 return 0; | |
1011 } | |
1012 | |
1013 | |
1014 static int dca_subframe_footer(DCAContext * s) | |
1015 { | |
1016 int aux_data_count = 0, i; | |
1017 int lfe_samples; | |
1018 | |
1019 /* | |
1020 * Unpack optional information | |
1021 */ | |
1022 | |
1023 if (s->timestamp) | |
1024 get_bits(&s->gb, 32); | |
1025 | |
1026 if (s->aux_data) | |
1027 aux_data_count = get_bits(&s->gb, 6); | |
1028 | |
1029 for (i = 0; i < aux_data_count; i++) | |
1030 get_bits(&s->gb, 8); | |
1031 | |
1032 if (s->crc_present && (s->downmix || s->dynrange)) | |
1033 get_bits(&s->gb, 16); | |
1034 | |
1035 lfe_samples = 2 * s->lfe * s->subsubframes; | |
1036 for (i = 0; i < lfe_samples; i++) { | |
1037 s->lfe_data[i] = s->lfe_data[i + lfe_samples]; | |
1038 } | |
1039 | |
1040 return 0; | |
1041 } | |
1042 | |
1043 /** | |
1044 * Decode a dca frame block | |
1045 * | |
1046 * @param s pointer to the DCAContext | |
1047 */ | |
1048 | |
1049 static int dca_decode_block(DCAContext * s) | |
1050 { | |
1051 | |
1052 /* Sanity check */ | |
1053 if (s->current_subframe >= s->subframes) { | |
1054 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", | |
1055 s->current_subframe, s->subframes); | |
1056 return -1; | |
1057 } | |
1058 | |
1059 if (!s->current_subsubframe) { | |
1060 #ifdef TRACE | |
1061 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); | |
1062 #endif | |
1063 /* Read subframe header */ | |
1064 if (dca_subframe_header(s)) | |
1065 return -1; | |
1066 } | |
1067 | |
1068 /* Read subsubframe */ | |
1069 #ifdef TRACE | |
1070 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); | |
1071 #endif | |
1072 if (dca_subsubframe(s)) | |
1073 return -1; | |
1074 | |
1075 /* Update state */ | |
1076 s->current_subsubframe++; | |
1077 if (s->current_subsubframe >= s->subsubframes) { | |
1078 s->current_subsubframe = 0; | |
1079 s->current_subframe++; | |
1080 } | |
1081 if (s->current_subframe >= s->subframes) { | |
1082 #ifdef TRACE | |
1083 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); | |
1084 #endif | |
1085 /* Read subframe footer */ | |
1086 if (dca_subframe_footer(s)) | |
1087 return -1; | |
1088 } | |
1089 | |
1090 return 0; | |
1091 } | |
1092 | |
1093 /** | |
1094 * Convert bitstream to one representation based on sync marker | |
1095 */ | |
6214 | 1096 static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst, |
4599 | 1097 int max_size) |
1098 { | |
1099 uint32_t mrk; | |
1100 int i, tmp; | |
6214 | 1101 const uint16_t *ssrc = (const uint16_t *) src; |
1102 uint16_t *sdst = (uint16_t *) dst; | |
4599 | 1103 PutBitContext pb; |
1104 | |
5027 | 1105 if((unsigned)src_size > (unsigned)max_size) { |
1106 av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n"); | |
4883 | 1107 return -1; |
5027 | 1108 } |
4883 | 1109 |
4599 | 1110 mrk = AV_RB32(src); |
1111 switch (mrk) { | |
1112 case DCA_MARKER_RAW_BE: | |
7671 | 1113 memcpy(dst, src, src_size); |
1114 return src_size; | |
4599 | 1115 case DCA_MARKER_RAW_LE: |
7671 | 1116 for (i = 0; i < (src_size + 1) >> 1; i++) |
4599 | 1117 *sdst++ = bswap_16(*ssrc++); |
7671 | 1118 return src_size; |
4599 | 1119 case DCA_MARKER_14B_BE: |
1120 case DCA_MARKER_14B_LE: | |
1121 init_put_bits(&pb, dst, max_size); | |
1122 for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) { | |
1123 tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF; | |
1124 put_bits(&pb, 14, tmp); | |
1125 } | |
1126 flush_put_bits(&pb); | |
1127 return (put_bits_count(&pb) + 7) >> 3; | |
1128 default: | |
1129 return -1; | |
1130 } | |
1131 } | |
1132 | |
1133 /** | |
1134 * Main frame decoding function | |
1135 * FIXME add arguments | |
1136 */ | |
1137 static int dca_decode_frame(AVCodecContext * avctx, | |
1138 void *data, int *data_size, | |
6214 | 1139 const uint8_t * buf, int buf_size) |
4599 | 1140 { |
1141 | |
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1142 int i; |
4599 | 1143 int16_t *samples = data; |
1144 DCAContext *s = avctx->priv_data; | |
1145 int channels; | |
1146 | |
1147 | |
1148 s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE); | |
1149 if (s->dca_buffer_size == -1) { | |
5027 | 1150 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); |
4599 | 1151 return -1; |
1152 } | |
1153 | |
1154 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
1155 if (dca_parse_frame_header(s) < 0) { | |
1156 //seems like the frame is corrupt, try with the next one | |
5645 | 1157 *data_size=0; |
4599 | 1158 return buf_size; |
1159 } | |
1160 //set AVCodec values with parsed data | |
1161 avctx->sample_rate = s->sample_rate; | |
1162 avctx->bit_rate = s->bit_rate; | |
1163 | |
4893 | 1164 channels = s->prim_channels + !!s->lfe; |
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1165 if(avctx->request_channels == 2 && s->prim_channels > 2) { |
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1166 channels = 2; |
4893 | 1167 s->output = DCA_STEREO; |
1168 } | |
1169 | |
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1170 /* There is nothing that prevents a dts frame to change channel configuration |
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1171 but FFmpeg doesn't support that so only set the channels if it is previously |
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1172 unset. Ideally during the first probe for channels the crc should be checked |
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1173 and only set avctx->channels when the crc is ok. Right now the decoder could |
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1174 set the channels based on a broken first frame.*/ |
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1175 if (!avctx->channels) |
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1176 avctx->channels = channels; |
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1177 |
4599 | 1178 if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) |
1179 return -1; | |
7725 | 1180 *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels; |
4599 | 1181 for (i = 0; i < (s->sample_blocks / 8); i++) { |
1182 dca_decode_block(s); | |
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1183 s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); |
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1184 samples += 256 * channels; |
4599 | 1185 } |
1186 | |
1187 return buf_size; | |
1188 } | |
1189 | |
1190 | |
1191 | |
1192 /** | |
1193 * DCA initialization | |
1194 * | |
1195 * @param avctx pointer to the AVCodecContext | |
1196 */ | |
1197 | |
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1198 static av_cold int dca_decode_init(AVCodecContext * avctx) |
4599 | 1199 { |
1200 DCAContext *s = avctx->priv_data; | |
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1201 int i; |
4599 | 1202 |
1203 s->avctx = avctx; | |
1204 dca_init_vlcs(); | |
1205 | |
1206 dsputil_init(&s->dsp, avctx); | |
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1207 ff_mdct_init(&s->imdct, 6, 1); |
6120 | 1208 |
1209 /* allow downmixing to stereo */ | |
1210 if (avctx->channels > 0 && avctx->request_channels < avctx->channels && | |
1211 avctx->request_channels == 2) { | |
1212 avctx->channels = avctx->request_channels; | |
1213 } | |
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1214 for(i = 0; i < 6; i++) |
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1215 s->samples_chanptr[i] = s->samples + i * 256; |
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1216 avctx->sample_fmt = SAMPLE_FMT_S16; |
4599 | 1217 return 0; |
1218 } | |
1219 | |
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1220 static av_cold int dca_decode_end(AVCodecContext * avctx) |
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1221 { |
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1222 DCAContext *s = avctx->priv_data; |
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1223 ff_mdct_end(&s->imdct); |
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1224 return 0; |
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1225 } |
4599 | 1226 |
1227 AVCodec dca_decoder = { | |
1228 .name = "dca", | |
1229 .type = CODEC_TYPE_AUDIO, | |
1230 .id = CODEC_ID_DTS, | |
1231 .priv_data_size = sizeof(DCAContext), | |
1232 .init = dca_decode_init, | |
1233 .decode = dca_decode_frame, | |
7738
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1234 .close = dca_decode_end, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6710
diff
changeset
|
1235 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), |
4599 | 1236 }; |