Mercurial > libavcodec.hg
annotate dca.c @ 8064:111abd3d30de libavcodec
Merge OBJS-$(FOO) lists with same FOO
author | mru |
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date | Sun, 26 Oct 2008 14:26:28 +0000 |
parents | 6bc70b15451d |
children | 574a0dcc4488 |
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 | |
7670
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71 #define DCA_MAX_FRAME_SIZE 16384 |
4599 | 72 |
73 /** Bit allocation */ | |
74 typedef struct { | |
75 int offset; ///< code values offset | |
76 int maxbits[8]; ///< max bits in VLC | |
77 int wrap; ///< wrap for get_vlc2() | |
78 VLC vlc[8]; ///< actual codes | |
79 } BitAlloc; | |
80 | |
81 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select | |
82 static BitAlloc dca_tmode; ///< transition mode VLCs | |
83 static BitAlloc dca_scalefactor; ///< scalefactor VLCs | |
84 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs | |
85 | |
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Fix multiple "¡Æinline/static¡Ç is not at beginning of declaration" warnings.
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86 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx) |
4599 | 87 { |
88 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset; | |
89 } | |
90 | |
91 typedef struct { | |
92 AVCodecContext *avctx; | |
93 /* Frame header */ | |
94 int frame_type; ///< type of the current frame | |
95 int samples_deficit; ///< deficit sample count | |
96 int crc_present; ///< crc is present in the bitstream | |
97 int sample_blocks; ///< number of PCM sample blocks | |
98 int frame_size; ///< primary frame byte size | |
99 int amode; ///< audio channels arrangement | |
100 int sample_rate; ///< audio sampling rate | |
101 int bit_rate; ///< transmission bit rate | |
102 | |
103 int downmix; ///< embedded downmix enabled | |
104 int dynrange; ///< embedded dynamic range flag | |
105 int timestamp; ///< embedded time stamp flag | |
106 int aux_data; ///< auxiliary data flag | |
107 int hdcd; ///< source material is mastered in HDCD | |
108 int ext_descr; ///< extension audio descriptor flag | |
109 int ext_coding; ///< extended coding flag | |
110 int aspf; ///< audio sync word insertion flag | |
111 int lfe; ///< low frequency effects flag | |
112 int predictor_history; ///< predictor history flag | |
113 int header_crc; ///< header crc check bytes | |
114 int multirate_inter; ///< multirate interpolator switch | |
115 int version; ///< encoder software revision | |
116 int copy_history; ///< copy history | |
117 int source_pcm_res; ///< source pcm resolution | |
118 int front_sum; ///< front sum/difference flag | |
119 int surround_sum; ///< surround sum/difference flag | |
120 int dialog_norm; ///< dialog normalisation parameter | |
121 | |
122 /* Primary audio coding header */ | |
123 int subframes; ///< number of subframes | |
6463 | 124 int total_channels; ///< number of channels including extensions |
4599 | 125 int prim_channels; ///< number of primary audio channels |
126 int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count | |
127 int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband | |
128 int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index | |
129 int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book | |
130 int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book | |
131 int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select | |
132 int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select | |
133 float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment | |
134 | |
135 /* Primary audio coding side information */ | |
136 int subsubframes; ///< number of subsubframes | |
137 int partial_samples; ///< partial subsubframe samples count | |
138 int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not) | |
139 int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs | |
140 int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index | |
141 int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients) | |
142 int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient) | |
143 int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook | |
144 int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors | |
145 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients | |
146 int dynrange_coef; ///< dynamic range coefficient | |
147 | |
148 int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands | |
149 | |
150 float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX * | |
151 2 /*history */ ]; ///< Low frequency effect data | |
152 int lfe_scale_factor; | |
153 | |
154 /* Subband samples history (for ADPCM) */ | |
155 float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; | |
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156 DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]); |
7730
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Half the size of subband_fir_noidea and get rid of memmove & memset of it.
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157 float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32]; |
7737 | 158 int hist_index[DCA_PRIM_CHANNELS_MAX]; |
4599 | 159 |
160 int output; ///< type of output | |
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161 float add_bias; ///< output bias |
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Fix dca decoder with non simd float2int16 conversion
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162 float scale_bias; ///< output scale |
4599 | 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 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
234 | |
235 /* Sync code */ | |
236 get_bits(&s->gb, 32); | |
237 | |
238 /* Frame header */ | |
239 s->frame_type = get_bits(&s->gb, 1); | |
240 s->samples_deficit = get_bits(&s->gb, 5) + 1; | |
241 s->crc_present = get_bits(&s->gb, 1); | |
242 s->sample_blocks = get_bits(&s->gb, 7) + 1; | |
243 s->frame_size = get_bits(&s->gb, 14) + 1; | |
244 if (s->frame_size < 95) | |
245 return -1; | |
246 s->amode = get_bits(&s->gb, 6); | |
247 s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)]; | |
248 if (!s->sample_rate) | |
249 return -1; | |
250 s->bit_rate = dca_bit_rates[get_bits(&s->gb, 5)]; | |
251 if (!s->bit_rate) | |
252 return -1; | |
253 | |
254 s->downmix = get_bits(&s->gb, 1); | |
255 s->dynrange = get_bits(&s->gb, 1); | |
256 s->timestamp = get_bits(&s->gb, 1); | |
257 s->aux_data = get_bits(&s->gb, 1); | |
258 s->hdcd = get_bits(&s->gb, 1); | |
259 s->ext_descr = get_bits(&s->gb, 3); | |
260 s->ext_coding = get_bits(&s->gb, 1); | |
261 s->aspf = get_bits(&s->gb, 1); | |
262 s->lfe = get_bits(&s->gb, 2); | |
263 s->predictor_history = get_bits(&s->gb, 1); | |
264 | |
265 /* TODO: check CRC */ | |
266 if (s->crc_present) | |
267 s->header_crc = get_bits(&s->gb, 16); | |
268 | |
269 s->multirate_inter = get_bits(&s->gb, 1); | |
270 s->version = get_bits(&s->gb, 4); | |
271 s->copy_history = get_bits(&s->gb, 2); | |
272 s->source_pcm_res = get_bits(&s->gb, 3); | |
273 s->front_sum = get_bits(&s->gb, 1); | |
274 s->surround_sum = get_bits(&s->gb, 1); | |
275 s->dialog_norm = get_bits(&s->gb, 4); | |
276 | |
277 /* FIXME: channels mixing levels */ | |
4893 | 278 s->output = s->amode; |
279 if(s->lfe) s->output |= DCA_LFE; | |
4599 | 280 |
281 #ifdef TRACE | |
282 av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); | |
283 av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit); | |
284 av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present); | |
285 av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n", | |
286 s->sample_blocks, s->sample_blocks * 32); | |
287 av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size); | |
288 av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n", | |
289 s->amode, dca_channels[s->amode]); | |
8061 | 290 av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n", |
291 s->sample_rate); | |
292 av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n", | |
293 s->bit_rate); | |
4599 | 294 av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix); |
295 av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange); | |
296 av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp); | |
297 av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data); | |
298 av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd); | |
299 av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr); | |
300 av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding); | |
301 av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf); | |
302 av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe); | |
303 av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n", | |
304 s->predictor_history); | |
305 av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc); | |
306 av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n", | |
307 s->multirate_inter); | |
308 av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version); | |
309 av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history); | |
310 av_log(s->avctx, AV_LOG_DEBUG, | |
311 "source pcm resolution: %i (%i bits/sample)\n", | |
312 s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]); | |
313 av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum); | |
314 av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum); | |
315 av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm); | |
316 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
317 #endif | |
318 | |
319 /* Primary audio coding header */ | |
320 s->subframes = get_bits(&s->gb, 4) + 1; | |
6463 | 321 s->total_channels = get_bits(&s->gb, 3) + 1; |
322 s->prim_channels = s->total_channels; | |
323 if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) | |
324 s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */ | |
4599 | 325 |
326 | |
327 for (i = 0; i < s->prim_channels; i++) { | |
328 s->subband_activity[i] = get_bits(&s->gb, 5) + 2; | |
329 if (s->subband_activity[i] > DCA_SUBBANDS) | |
330 s->subband_activity[i] = DCA_SUBBANDS; | |
331 } | |
332 for (i = 0; i < s->prim_channels; i++) { | |
333 s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; | |
334 if (s->vq_start_subband[i] > DCA_SUBBANDS) | |
335 s->vq_start_subband[i] = DCA_SUBBANDS; | |
336 } | |
337 get_array(&s->gb, s->joint_intensity, s->prim_channels, 3); | |
338 get_array(&s->gb, s->transient_huffman, s->prim_channels, 2); | |
339 get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3); | |
340 get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3); | |
341 | |
342 /* Get codebooks quantization indexes */ | |
343 memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); | |
344 for (j = 1; j < 11; j++) | |
345 for (i = 0; i < s->prim_channels; i++) | |
346 s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); | |
347 | |
348 /* Get scale factor adjustment */ | |
349 for (j = 0; j < 11; j++) | |
350 for (i = 0; i < s->prim_channels; i++) | |
351 s->scalefactor_adj[i][j] = 1; | |
352 | |
353 for (j = 1; j < 11; j++) | |
354 for (i = 0; i < s->prim_channels; i++) | |
355 if (s->quant_index_huffman[i][j] < thr[j]) | |
356 s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; | |
357 | |
358 if (s->crc_present) { | |
359 /* Audio header CRC check */ | |
360 get_bits(&s->gb, 16); | |
361 } | |
362 | |
363 s->current_subframe = 0; | |
364 s->current_subsubframe = 0; | |
365 | |
366 #ifdef TRACE | |
367 av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes); | |
368 av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels); | |
369 for(i = 0; i < s->prim_channels; i++){ | |
370 av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]); | |
371 av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]); | |
372 av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]); | |
373 av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]); | |
374 av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]); | |
375 av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]); | |
376 av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:"); | |
377 for (j = 0; j < 11; j++) | |
378 av_log(s->avctx, AV_LOG_DEBUG, " %i", | |
379 s->quant_index_huffman[i][j]); | |
380 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
381 av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:"); | |
382 for (j = 0; j < 11; j++) | |
383 av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]); | |
384 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
385 } | |
386 #endif | |
387 | |
388 return 0; | |
389 } | |
390 | |
391 | |
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392 static inline int get_scale(GetBitContext *gb, int level, int value) |
4599 | 393 { |
394 if (level < 5) { | |
395 /* huffman encoded */ | |
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396 value += get_bitalloc(gb, &dca_scalefactor, level); |
4599 | 397 } else if(level < 8) |
398 value = get_bits(gb, level + 1); | |
399 return value; | |
400 } | |
401 | |
402 static int dca_subframe_header(DCAContext * s) | |
403 { | |
404 /* Primary audio coding side information */ | |
405 int j, k; | |
406 | |
407 s->subsubframes = get_bits(&s->gb, 2) + 1; | |
408 s->partial_samples = get_bits(&s->gb, 3); | |
409 for (j = 0; j < s->prim_channels; j++) { | |
410 for (k = 0; k < s->subband_activity[j]; k++) | |
411 s->prediction_mode[j][k] = get_bits(&s->gb, 1); | |
412 } | |
413 | |
414 /* Get prediction codebook */ | |
415 for (j = 0; j < s->prim_channels; j++) { | |
416 for (k = 0; k < s->subband_activity[j]; k++) { | |
417 if (s->prediction_mode[j][k] > 0) { | |
418 /* (Prediction coefficient VQ address) */ | |
419 s->prediction_vq[j][k] = get_bits(&s->gb, 12); | |
420 } | |
421 } | |
422 } | |
423 | |
424 /* Bit allocation index */ | |
425 for (j = 0; j < s->prim_channels; j++) { | |
426 for (k = 0; k < s->vq_start_subband[j]; k++) { | |
427 if (s->bitalloc_huffman[j] == 6) | |
428 s->bitalloc[j][k] = get_bits(&s->gb, 5); | |
429 else if (s->bitalloc_huffman[j] == 5) | |
430 s->bitalloc[j][k] = get_bits(&s->gb, 4); | |
6463 | 431 else if (s->bitalloc_huffman[j] == 7) { |
432 av_log(s->avctx, AV_LOG_ERROR, | |
433 "Invalid bit allocation index\n"); | |
434 return -1; | |
435 } else { | |
4599 | 436 s->bitalloc[j][k] = |
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437 get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); |
4599 | 438 } |
439 | |
440 if (s->bitalloc[j][k] > 26) { | |
441 // av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n", | |
442 // j, k, s->bitalloc[j][k]); | |
443 return -1; | |
444 } | |
445 } | |
446 } | |
447 | |
448 /* Transition mode */ | |
449 for (j = 0; j < s->prim_channels; j++) { | |
450 for (k = 0; k < s->subband_activity[j]; k++) { | |
451 s->transition_mode[j][k] = 0; | |
452 if (s->subsubframes > 1 && | |
453 k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { | |
454 s->transition_mode[j][k] = | |
455 get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); | |
456 } | |
457 } | |
458 } | |
459 | |
460 for (j = 0; j < s->prim_channels; j++) { | |
6214 | 461 const uint32_t *scale_table; |
4599 | 462 int scale_sum; |
463 | |
464 memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); | |
465 | |
466 if (s->scalefactor_huffman[j] == 6) | |
6214 | 467 scale_table = scale_factor_quant7; |
4599 | 468 else |
6214 | 469 scale_table = scale_factor_quant6; |
4599 | 470 |
471 /* When huffman coded, only the difference is encoded */ | |
472 scale_sum = 0; | |
473 | |
474 for (k = 0; k < s->subband_activity[j]; k++) { | |
475 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { | |
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476 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 477 s->scale_factor[j][k][0] = scale_table[scale_sum]; |
478 } | |
479 | |
480 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { | |
481 /* Get second scale factor */ | |
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482 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 483 s->scale_factor[j][k][1] = scale_table[scale_sum]; |
484 } | |
485 } | |
486 } | |
487 | |
488 /* Joint subband scale factor codebook select */ | |
489 for (j = 0; j < s->prim_channels; j++) { | |
490 /* Transmitted only if joint subband coding enabled */ | |
491 if (s->joint_intensity[j] > 0) | |
492 s->joint_huff[j] = get_bits(&s->gb, 3); | |
493 } | |
494 | |
495 /* Scale factors for joint subband coding */ | |
496 for (j = 0; j < s->prim_channels; j++) { | |
497 int source_channel; | |
498 | |
499 /* Transmitted only if joint subband coding enabled */ | |
500 if (s->joint_intensity[j] > 0) { | |
501 int scale = 0; | |
502 source_channel = s->joint_intensity[j] - 1; | |
503 | |
504 /* When huffman coded, only the difference is encoded | |
505 * (is this valid as well for joint scales ???) */ | |
506 | |
507 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { | |
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508 scale = get_scale(&s->gb, s->joint_huff[j], 0); |
4599 | 509 scale += 64; /* bias */ |
510 s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ | |
511 } | |
512 | |
513 if (!s->debug_flag & 0x02) { | |
514 av_log(s->avctx, AV_LOG_DEBUG, | |
515 "Joint stereo coding not supported\n"); | |
516 s->debug_flag |= 0x02; | |
517 } | |
518 } | |
519 } | |
520 | |
521 /* Stereo downmix coefficients */ | |
4894 | 522 if (s->prim_channels > 2) { |
523 if(s->downmix) { | |
4895 | 524 for (j = 0; j < s->prim_channels; j++) { |
525 s->downmix_coef[j][0] = get_bits(&s->gb, 7); | |
526 s->downmix_coef[j][1] = get_bits(&s->gb, 7); | |
527 } | |
4894 | 528 } else { |
529 int am = s->amode & DCA_CHANNEL_MASK; | |
530 for (j = 0; j < s->prim_channels; j++) { | |
531 s->downmix_coef[j][0] = dca_default_coeffs[am][j][0]; | |
532 s->downmix_coef[j][1] = dca_default_coeffs[am][j][1]; | |
533 } | |
534 } | |
4599 | 535 } |
536 | |
537 /* Dynamic range coefficient */ | |
538 if (s->dynrange) | |
539 s->dynrange_coef = get_bits(&s->gb, 8); | |
540 | |
541 /* Side information CRC check word */ | |
542 if (s->crc_present) { | |
543 get_bits(&s->gb, 16); | |
544 } | |
545 | |
546 /* | |
547 * Primary audio data arrays | |
548 */ | |
549 | |
550 /* VQ encoded high frequency subbands */ | |
551 for (j = 0; j < s->prim_channels; j++) | |
552 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
553 /* 1 vector -> 32 samples */ | |
554 s->high_freq_vq[j][k] = get_bits(&s->gb, 10); | |
555 | |
556 /* Low frequency effect data */ | |
557 if (s->lfe) { | |
558 /* LFE samples */ | |
559 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
560 float lfe_scale; | |
561 | |
562 for (j = lfe_samples; j < lfe_samples * 2; j++) { | |
563 /* Signed 8 bits int */ | |
564 s->lfe_data[j] = get_sbits(&s->gb, 8); | |
565 } | |
566 | |
567 /* Scale factor index */ | |
568 s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)]; | |
569 | |
570 /* Quantization step size * scale factor */ | |
571 lfe_scale = 0.035 * s->lfe_scale_factor; | |
572 | |
573 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
574 s->lfe_data[j] *= lfe_scale; | |
575 } | |
576 | |
577 #ifdef TRACE | |
578 av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes); | |
579 av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", | |
580 s->partial_samples); | |
581 for (j = 0; j < s->prim_channels; j++) { | |
582 av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:"); | |
583 for (k = 0; k < s->subband_activity[j]; k++) | |
584 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]); | |
585 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
586 } | |
587 for (j = 0; j < s->prim_channels; j++) { | |
588 for (k = 0; k < s->subband_activity[j]; k++) | |
589 av_log(s->avctx, AV_LOG_DEBUG, | |
590 "prediction coefs: %f, %f, %f, %f\n", | |
591 (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192, | |
592 (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192, | |
593 (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192, | |
594 (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192); | |
595 } | |
596 for (j = 0; j < s->prim_channels; j++) { | |
597 av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: "); | |
598 for (k = 0; k < s->vq_start_subband[j]; k++) | |
599 av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]); | |
600 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
601 } | |
602 for (j = 0; j < s->prim_channels; j++) { | |
603 av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:"); | |
604 for (k = 0; k < s->subband_activity[j]; k++) | |
605 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]); | |
606 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
607 } | |
608 for (j = 0; j < s->prim_channels; j++) { | |
609 av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:"); | |
610 for (k = 0; k < s->subband_activity[j]; k++) { | |
611 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) | |
612 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]); | |
613 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) | |
614 av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]); | |
615 } | |
616 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
617 } | |
618 for (j = 0; j < s->prim_channels; j++) { | |
619 if (s->joint_intensity[j] > 0) { | |
5069 | 620 int source_channel = s->joint_intensity[j] - 1; |
4599 | 621 av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n"); |
622 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) | |
623 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]); | |
624 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
625 } | |
626 } | |
627 if (s->prim_channels > 2 && s->downmix) { | |
628 av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); | |
629 for (j = 0; j < s->prim_channels; j++) { | |
630 av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]); | |
631 av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]); | |
632 } | |
633 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
634 } | |
635 for (j = 0; j < s->prim_channels; j++) | |
636 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
637 av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]); | |
638 if(s->lfe){ | |
5069 | 639 int lfe_samples = 2 * s->lfe * s->subsubframes; |
4599 | 640 av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n"); |
641 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
642 av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]); | |
643 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
644 } | |
645 #endif | |
646 | |
647 return 0; | |
648 } | |
649 | |
650 static void qmf_32_subbands(DCAContext * s, int chans, | |
651 float samples_in[32][8], float *samples_out, | |
652 float scale, float bias) | |
653 { | |
5974
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654 const float *prCoeff; |
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655 int i, j; |
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656 DECLARE_ALIGNED_16(float, raXin[32]); |
4599 | 657 |
7737 | 658 int hist_index= s->hist_index[chans]; |
4599 | 659 float *subband_fir_hist2 = s->subband_fir_noidea[chans]; |
660 | |
7738
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661 int subindex; |
4599 | 662 |
7738
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663 scale *= sqrt(1/8.0); |
4599 | 664 |
665 /* Select filter */ | |
666 if (!s->multirate_inter) /* Non-perfect reconstruction */ | |
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667 prCoeff = fir_32bands_nonperfect; |
4599 | 668 else /* Perfect reconstruction */ |
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669 prCoeff = fir_32bands_perfect; |
4599 | 670 |
671 /* Reconstructed channel sample index */ | |
672 for (subindex = 0; subindex < 8; subindex++) { | |
7737 | 673 float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index; |
4599 | 674 /* Load in one sample from each subband and clear inactive subbands */ |
7738
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675 for (i = 0; i < s->subband_activity[chans]; i++){ |
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676 if((i-1)&2) raXin[i] = -samples_in[i][subindex]; |
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677 else raXin[i] = samples_in[i][subindex]; |
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678 } |
4599 | 679 for (; i < 32; i++) |
680 raXin[i] = 0.0; | |
681 | |
7738
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682 ff_imdct_half(&s->imdct, subband_fir_hist, raXin); |
4599 | 683 |
684 /* Multiply by filter coefficients */ | |
7738
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685 for (i = 0; i < 16; i++){ |
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686 float a= subband_fir_hist2[i ]; |
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687 float b= subband_fir_hist2[i+16]; |
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688 float c= 0; |
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689 float d= 0; |
7737 | 690 for (j = 0; j < 512-hist_index; j += 64){ |
7738
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691 a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]); |
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692 b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]); |
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693 c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]); |
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694 d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]); |
4599 | 695 } |
7737 | 696 for ( ; j < 512; j += 64){ |
7738
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697 a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]); |
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698 b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]); |
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699 c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]); |
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700 d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]); |
7737 | 701 } |
7738
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702 samples_out[i ] = a * scale + bias; |
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703 samples_out[i+16] = b * scale + bias; |
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704 subband_fir_hist2[i ] = c; |
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705 subband_fir_hist2[i+16] = d; |
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706 } |
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707 samples_out+= 32; |
4599 | 708 |
7737 | 709 hist_index = (hist_index-32)&511; |
4599 | 710 } |
7737 | 711 s->hist_index[chans]= hist_index; |
4599 | 712 } |
713 | |
714 static void lfe_interpolation_fir(int decimation_select, | |
715 int num_deci_sample, float *samples_in, | |
716 float *samples_out, float scale, | |
717 float bias) | |
718 { | |
719 /* samples_in: An array holding decimated samples. | |
720 * Samples in current subframe starts from samples_in[0], | |
721 * while samples_in[-1], samples_in[-2], ..., stores samples | |
722 * from last subframe as history. | |
723 * | |
724 * samples_out: An array holding interpolated samples | |
725 */ | |
726 | |
727 int decifactor, k, j; | |
728 const float *prCoeff; | |
729 | |
730 int interp_index = 0; /* Index to the interpolated samples */ | |
731 int deciindex; | |
732 | |
733 /* Select decimation filter */ | |
734 if (decimation_select == 1) { | |
735 decifactor = 128; | |
736 prCoeff = lfe_fir_128; | |
737 } else { | |
738 decifactor = 64; | |
739 prCoeff = lfe_fir_64; | |
740 } | |
741 /* Interpolation */ | |
742 for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { | |
743 /* One decimated sample generates decifactor interpolated ones */ | |
744 for (k = 0; k < decifactor; k++) { | |
745 float rTmp = 0.0; | |
746 //FIXME the coeffs are symetric, fix that | |
747 for (j = 0; j < 512 / decifactor; j++) | |
748 rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor]; | |
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changeset
|
749 samples_out[interp_index++] = (rTmp * scale) + bias; |
4599 | 750 } |
751 } | |
752 } | |
753 | |
754 /* downmixing routines */ | |
4894 | 755 #define MIX_REAR1(samples, si1, rs, coef) \ |
756 samples[i] += samples[si1] * coef[rs][0]; \ | |
757 samples[i+256] += samples[si1] * coef[rs][1]; | |
4599 | 758 |
4894 | 759 #define MIX_REAR2(samples, si1, si2, rs, coef) \ |
760 samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ | |
761 samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; | |
4599 | 762 |
4894 | 763 #define MIX_FRONT3(samples, coef) \ |
4599 | 764 t = samples[i]; \ |
4894 | 765 samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \ |
766 samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1]; | |
4599 | 767 |
768 #define DOWNMIX_TO_STEREO(op1, op2) \ | |
769 for(i = 0; i < 256; i++){ \ | |
770 op1 \ | |
771 op2 \ | |
772 } | |
773 | |
4894 | 774 static void dca_downmix(float *samples, int srcfmt, |
775 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]) | |
4599 | 776 { |
777 int i; | |
778 float t; | |
4894 | 779 float coef[DCA_PRIM_CHANNELS_MAX][2]; |
780 | |
781 for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) { | |
782 coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]]; | |
783 coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]]; | |
784 } | |
4599 | 785 |
786 switch (srcfmt) { | |
787 case DCA_MONO: | |
788 case DCA_CHANNEL: | |
789 case DCA_STEREO_TOTAL: | |
790 case DCA_STEREO_SUMDIFF: | |
791 case DCA_4F2R: | |
792 av_log(NULL, 0, "Not implemented!\n"); | |
793 break; | |
794 case DCA_STEREO: | |
795 break; | |
796 case DCA_3F: | |
4894 | 797 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),); |
4599 | 798 break; |
799 case DCA_2F1R: | |
4894 | 800 DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),); |
4599 | 801 break; |
802 case DCA_3F1R: | |
4894 | 803 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
804 MIX_REAR1(samples, i + 768, 3, coef)); | |
4599 | 805 break; |
806 case DCA_2F2R: | |
4894 | 807 DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),); |
4599 | 808 break; |
809 case DCA_3F2R: | |
4894 | 810 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
811 MIX_REAR2(samples, i + 768, i + 1024, 3, coef)); | |
4599 | 812 break; |
813 } | |
814 } | |
815 | |
816 | |
817 /* Very compact version of the block code decoder that does not use table | |
818 * look-up but is slightly slower */ | |
819 static int decode_blockcode(int code, int levels, int *values) | |
820 { | |
821 int i; | |
822 int offset = (levels - 1) >> 1; | |
823 | |
824 for (i = 0; i < 4; i++) { | |
825 values[i] = (code % levels) - offset; | |
826 code /= levels; | |
827 } | |
828 | |
829 if (code == 0) | |
830 return 0; | |
831 else { | |
832 av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n"); | |
833 return -1; | |
834 } | |
835 } | |
836 | |
837 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; | |
838 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; | |
839 | |
840 static int dca_subsubframe(DCAContext * s) | |
841 { | |
842 int k, l; | |
843 int subsubframe = s->current_subsubframe; | |
844 | |
6214 | 845 const float *quant_step_table; |
4599 | 846 |
847 /* FIXME */ | |
848 float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; | |
849 | |
850 /* | |
851 * Audio data | |
852 */ | |
853 | |
854 /* Select quantization step size table */ | |
855 if (s->bit_rate == 0x1f) | |
6214 | 856 quant_step_table = lossless_quant_d; |
4599 | 857 else |
6214 | 858 quant_step_table = lossy_quant_d; |
4599 | 859 |
860 for (k = 0; k < s->prim_channels; k++) { | |
861 for (l = 0; l < s->vq_start_subband[k]; l++) { | |
862 int m; | |
863 | |
864 /* Select the mid-tread linear quantizer */ | |
865 int abits = s->bitalloc[k][l]; | |
866 | |
867 float quant_step_size = quant_step_table[abits]; | |
868 float rscale; | |
869 | |
870 /* | |
871 * Determine quantization index code book and its type | |
872 */ | |
873 | |
874 /* Select quantization index code book */ | |
875 int sel = s->quant_index_huffman[k][abits]; | |
876 | |
877 /* | |
878 * Extract bits from the bit stream | |
879 */ | |
880 if(!abits){ | |
881 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0])); | |
882 }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){ | |
883 if(abits <= 7){ | |
884 /* Block code */ | |
885 int block_code1, block_code2, size, levels; | |
886 int block[8]; | |
887 | |
888 size = abits_sizes[abits-1]; | |
889 levels = abits_levels[abits-1]; | |
890 | |
891 block_code1 = get_bits(&s->gb, size); | |
892 /* FIXME Should test return value */ | |
893 decode_blockcode(block_code1, levels, block); | |
894 block_code2 = get_bits(&s->gb, size); | |
895 decode_blockcode(block_code2, levels, &block[4]); | |
896 for (m = 0; m < 8; m++) | |
897 subband_samples[k][l][m] = block[m]; | |
898 }else{ | |
899 /* no coding */ | |
900 for (m = 0; m < 8; m++) | |
901 subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3); | |
902 } | |
903 }else{ | |
904 /* Huffman coded */ | |
905 for (m = 0; m < 8; m++) | |
906 subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); | |
907 } | |
908 | |
909 /* Deal with transients */ | |
910 if (s->transition_mode[k][l] && | |
911 subsubframe >= s->transition_mode[k][l]) | |
912 rscale = quant_step_size * s->scale_factor[k][l][1]; | |
913 else | |
914 rscale = quant_step_size * s->scale_factor[k][l][0]; | |
915 | |
916 rscale *= s->scalefactor_adj[k][sel]; | |
917 | |
918 for (m = 0; m < 8; m++) | |
919 subband_samples[k][l][m] *= rscale; | |
920 | |
921 /* | |
922 * Inverse ADPCM if in prediction mode | |
923 */ | |
924 if (s->prediction_mode[k][l]) { | |
925 int n; | |
926 for (m = 0; m < 8; m++) { | |
927 for (n = 1; n <= 4; n++) | |
928 if (m >= n) | |
929 subband_samples[k][l][m] += | |
930 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
931 subband_samples[k][l][m - n] / 8192); | |
932 else if (s->predictor_history) | |
933 subband_samples[k][l][m] += | |
934 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
935 s->subband_samples_hist[k][l][m - n + | |
936 4] / 8192); | |
937 } | |
938 } | |
939 } | |
940 | |
941 /* | |
942 * Decode VQ encoded high frequencies | |
943 */ | |
944 for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { | |
945 /* 1 vector -> 32 samples but we only need the 8 samples | |
946 * for this subsubframe. */ | |
947 int m; | |
948 | |
949 if (!s->debug_flag & 0x01) { | |
950 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); | |
951 s->debug_flag |= 0x01; | |
952 } | |
953 | |
954 for (m = 0; m < 8; m++) { | |
955 subband_samples[k][l][m] = | |
956 high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 + | |
957 m] | |
958 * (float) s->scale_factor[k][l][0] / 16.0; | |
959 } | |
960 } | |
961 } | |
962 | |
963 /* Check for DSYNC after subsubframe */ | |
964 if (s->aspf || subsubframe == s->subsubframes - 1) { | |
965 if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */ | |
966 #ifdef TRACE | |
967 av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n"); | |
968 #endif | |
969 } else { | |
970 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); | |
971 } | |
972 } | |
973 | |
974 /* Backup predictor history for adpcm */ | |
975 for (k = 0; k < s->prim_channels; k++) | |
976 for (l = 0; l < s->vq_start_subband[k]; l++) | |
977 memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4], | |
978 4 * sizeof(subband_samples[0][0][0])); | |
979 | |
980 /* 32 subbands QMF */ | |
981 for (k = 0; k < s->prim_channels; k++) { | |
982 /* static float pcm_to_double[8] = | |
983 {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/ | |
984 qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k], | |
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985 M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ , |
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986 s->add_bias ); |
4599 | 987 } |
988 | |
989 /* Down mixing */ | |
990 | |
991 if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) { | |
4894 | 992 dca_downmix(s->samples, s->amode, s->downmix_coef); |
4599 | 993 } |
994 | |
995 /* Generate LFE samples for this subsubframe FIXME!!! */ | |
996 if (s->output & DCA_LFE) { | |
997 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
998 int i_channels = dca_channels[s->output & DCA_CHANNEL_MASK]; | |
999 | |
1000 lfe_interpolation_fir(s->lfe, 2 * s->lfe, | |
1001 s->lfe_data + lfe_samples + | |
1002 2 * s->lfe * subsubframe, | |
1003 &s->samples[256 * i_channels], | |
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1004 (1.0/256.0)*s->scale_bias, s->add_bias); |
4599 | 1005 /* Outputs 20bits pcm samples */ |
1006 } | |
1007 | |
1008 return 0; | |
1009 } | |
1010 | |
1011 | |
1012 static int dca_subframe_footer(DCAContext * s) | |
1013 { | |
1014 int aux_data_count = 0, i; | |
1015 int lfe_samples; | |
1016 | |
1017 /* | |
1018 * Unpack optional information | |
1019 */ | |
1020 | |
1021 if (s->timestamp) | |
1022 get_bits(&s->gb, 32); | |
1023 | |
1024 if (s->aux_data) | |
1025 aux_data_count = get_bits(&s->gb, 6); | |
1026 | |
1027 for (i = 0; i < aux_data_count; i++) | |
1028 get_bits(&s->gb, 8); | |
1029 | |
1030 if (s->crc_present && (s->downmix || s->dynrange)) | |
1031 get_bits(&s->gb, 16); | |
1032 | |
1033 lfe_samples = 2 * s->lfe * s->subsubframes; | |
1034 for (i = 0; i < lfe_samples; i++) { | |
1035 s->lfe_data[i] = s->lfe_data[i + lfe_samples]; | |
1036 } | |
1037 | |
1038 return 0; | |
1039 } | |
1040 | |
1041 /** | |
1042 * Decode a dca frame block | |
1043 * | |
1044 * @param s pointer to the DCAContext | |
1045 */ | |
1046 | |
1047 static int dca_decode_block(DCAContext * s) | |
1048 { | |
1049 | |
1050 /* Sanity check */ | |
1051 if (s->current_subframe >= s->subframes) { | |
1052 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", | |
1053 s->current_subframe, s->subframes); | |
1054 return -1; | |
1055 } | |
1056 | |
1057 if (!s->current_subsubframe) { | |
1058 #ifdef TRACE | |
1059 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); | |
1060 #endif | |
1061 /* Read subframe header */ | |
1062 if (dca_subframe_header(s)) | |
1063 return -1; | |
1064 } | |
1065 | |
1066 /* Read subsubframe */ | |
1067 #ifdef TRACE | |
1068 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); | |
1069 #endif | |
1070 if (dca_subsubframe(s)) | |
1071 return -1; | |
1072 | |
1073 /* Update state */ | |
1074 s->current_subsubframe++; | |
1075 if (s->current_subsubframe >= s->subsubframes) { | |
1076 s->current_subsubframe = 0; | |
1077 s->current_subframe++; | |
1078 } | |
1079 if (s->current_subframe >= s->subframes) { | |
1080 #ifdef TRACE | |
1081 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); | |
1082 #endif | |
1083 /* Read subframe footer */ | |
1084 if (dca_subframe_footer(s)) | |
1085 return -1; | |
1086 } | |
1087 | |
1088 return 0; | |
1089 } | |
1090 | |
1091 /** | |
1092 * Convert bitstream to one representation based on sync marker | |
1093 */ | |
6214 | 1094 static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst, |
4599 | 1095 int max_size) |
1096 { | |
1097 uint32_t mrk; | |
1098 int i, tmp; | |
6214 | 1099 const uint16_t *ssrc = (const uint16_t *) src; |
1100 uint16_t *sdst = (uint16_t *) dst; | |
4599 | 1101 PutBitContext pb; |
1102 | |
5027 | 1103 if((unsigned)src_size > (unsigned)max_size) { |
1104 av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n"); | |
4883 | 1105 return -1; |
5027 | 1106 } |
4883 | 1107 |
4599 | 1108 mrk = AV_RB32(src); |
1109 switch (mrk) { | |
1110 case DCA_MARKER_RAW_BE: | |
7671 | 1111 memcpy(dst, src, src_size); |
1112 return src_size; | |
4599 | 1113 case DCA_MARKER_RAW_LE: |
7671 | 1114 for (i = 0; i < (src_size + 1) >> 1; i++) |
4599 | 1115 *sdst++ = bswap_16(*ssrc++); |
7671 | 1116 return src_size; |
4599 | 1117 case DCA_MARKER_14B_BE: |
1118 case DCA_MARKER_14B_LE: | |
1119 init_put_bits(&pb, dst, max_size); | |
1120 for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) { | |
1121 tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF; | |
1122 put_bits(&pb, 14, tmp); | |
1123 } | |
1124 flush_put_bits(&pb); | |
1125 return (put_bits_count(&pb) + 7) >> 3; | |
1126 default: | |
1127 return -1; | |
1128 } | |
1129 } | |
1130 | |
1131 /** | |
1132 * Main frame decoding function | |
1133 * FIXME add arguments | |
1134 */ | |
1135 static int dca_decode_frame(AVCodecContext * avctx, | |
1136 void *data, int *data_size, | |
6214 | 1137 const uint8_t * buf, int buf_size) |
4599 | 1138 { |
1139 | |
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1140 int i; |
4599 | 1141 int16_t *samples = data; |
1142 DCAContext *s = avctx->priv_data; | |
1143 int channels; | |
1144 | |
1145 | |
1146 s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE); | |
1147 if (s->dca_buffer_size == -1) { | |
5027 | 1148 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); |
4599 | 1149 return -1; |
1150 } | |
1151 | |
1152 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
1153 if (dca_parse_frame_header(s) < 0) { | |
1154 //seems like the frame is corrupt, try with the next one | |
5645 | 1155 *data_size=0; |
4599 | 1156 return buf_size; |
1157 } | |
1158 //set AVCodec values with parsed data | |
1159 avctx->sample_rate = s->sample_rate; | |
1160 avctx->bit_rate = s->bit_rate; | |
1161 | |
4893 | 1162 channels = s->prim_channels + !!s->lfe; |
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1163 if(avctx->request_channels == 2 && s->prim_channels > 2) { |
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1164 channels = 2; |
4893 | 1165 s->output = DCA_STEREO; |
1166 } | |
1167 | |
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1168 /* There is nothing that prevents a dts frame to change channel configuration |
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1169 but FFmpeg doesn't support that so only set the channels if it is previously |
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1170 unset. Ideally during the first probe for channels the crc should be checked |
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1171 and only set avctx->channels when the crc is ok. Right now the decoder could |
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1172 set the channels based on a broken first frame.*/ |
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1173 if (!avctx->channels) |
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1174 avctx->channels = channels; |
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1175 |
4599 | 1176 if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) |
1177 return -1; | |
7725 | 1178 *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels; |
4599 | 1179 for (i = 0; i < (s->sample_blocks / 8); i++) { |
1180 dca_decode_block(s); | |
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1181 s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); |
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1182 samples += 256 * channels; |
4599 | 1183 } |
1184 | |
1185 return buf_size; | |
1186 } | |
1187 | |
1188 | |
1189 | |
1190 /** | |
1191 * DCA initialization | |
1192 * | |
1193 * @param avctx pointer to the AVCodecContext | |
1194 */ | |
1195 | |
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1196 static av_cold int dca_decode_init(AVCodecContext * avctx) |
4599 | 1197 { |
1198 DCAContext *s = avctx->priv_data; | |
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1199 int i; |
4599 | 1200 |
1201 s->avctx = avctx; | |
1202 dca_init_vlcs(); | |
1203 | |
1204 dsputil_init(&s->dsp, avctx); | |
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1205 ff_mdct_init(&s->imdct, 6, 1); |
6120 | 1206 |
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1207 for(i = 0; i < 6; i++) |
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1208 s->samples_chanptr[i] = s->samples + i * 256; |
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1209 avctx->sample_fmt = SAMPLE_FMT_S16; |
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1210 |
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1211 if(s->dsp.float_to_int16 == ff_float_to_int16_c) { |
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1212 s->add_bias = 385.0f; |
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1213 s->scale_bias = 1.0 / 32768.0; |
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1214 } else { |
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1215 s->add_bias = 0.0f; |
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1216 s->scale_bias = 1.0; |
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Disable codec downmix when not using simd instead of silently produce silence
banan
parents:
8062
diff
changeset
|
1217 |
6bc70b15451d
Disable codec downmix when not using simd instead of silently produce silence
banan
parents:
8062
diff
changeset
|
1218 /* allow downmixing to stereo */ |
6bc70b15451d
Disable codec downmix when not using simd instead of silently produce silence
banan
parents:
8062
diff
changeset
|
1219 if (avctx->channels > 0 && avctx->request_channels < avctx->channels && |
6bc70b15451d
Disable codec downmix when not using simd instead of silently produce silence
banan
parents:
8062
diff
changeset
|
1220 avctx->request_channels == 2) { |
6bc70b15451d
Disable codec downmix when not using simd instead of silently produce silence
banan
parents:
8062
diff
changeset
|
1221 avctx->channels = avctx->request_channels; |
6bc70b15451d
Disable codec downmix when not using simd instead of silently produce silence
banan
parents:
8062
diff
changeset
|
1222 } |
8062
17aeecee2a97
Fix dca decoder with non simd float2int16 conversion
banan
parents:
8061
diff
changeset
|
1223 } |
17aeecee2a97
Fix dca decoder with non simd float2int16 conversion
banan
parents:
8061
diff
changeset
|
1224 |
17aeecee2a97
Fix dca decoder with non simd float2int16 conversion
banan
parents:
8061
diff
changeset
|
1225 |
4599 | 1226 return 0; |
1227 } | |
1228 | |
7738
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
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1229 static av_cold int dca_decode_end(AVCodecContext * avctx) |
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1230 { |
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1231 DCAContext *s = avctx->priv_data; |
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1232 ff_mdct_end(&s->imdct); |
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1233 return 0; |
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1234 } |
4599 | 1235 |
1236 AVCodec dca_decoder = { | |
1237 .name = "dca", | |
1238 .type = CODEC_TYPE_AUDIO, | |
1239 .id = CODEC_ID_DTS, | |
1240 .priv_data_size = sizeof(DCAContext), | |
1241 .init = dca_decode_init, | |
1242 .decode = dca_decode_frame, | |
7738
93ba37a9098c
Replace obfuscated mdct in qmf_32_subbands() by ff_imdct_half().
michael
parents:
7737
diff
changeset
|
1243 .close = dca_decode_end, |
7040
e943e1409077
Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents:
6710
diff
changeset
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1244 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), |
4599 | 1245 }; |