Mercurial > libavcodec.hg
annotate dca.c @ 8519:cc64e1343397 libavcodec
Use H264 MMX chroma functions to accelerate RV40 decoding.
Patch by Mathieu Velten (matmaul A gmail)
author | cehoyos |
---|---|
date | Sun, 04 Jan 2009 01:36:11 +0000 |
parents | ee1b8c54a603 |
children | e9d9d946f213 |
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 | |
8100 | 62 /* Tables for mapping dts channel configurations to libavcodec multichannel api. |
63 * Some compromises have been made for special configurations. Most configurations | |
64 * are never used so complete accuracy is not needed. | |
65 * | |
66 * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead. | |
8126 | 67 * S -> side, when both rear and back are configured move one of them to the side channel |
8100 | 68 * OV -> center back |
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69 * All 2 channel configurations -> CH_LAYOUT_STEREO |
8100 | 70 */ |
71 | |
72 static const int64_t dca_core_channel_layout[] = { | |
8102
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73 CH_FRONT_CENTER, ///< 1, A |
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74 CH_LAYOUT_STEREO, ///< 2, A + B (dual mono) |
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75 CH_LAYOUT_STEREO, ///< 2, L + R (stereo) |
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76 CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference) |
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77 CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total) |
8103 | 78 CH_LAYOUT_STEREO|CH_FRONT_CENTER, ///< 3, C+L+R |
79 CH_LAYOUT_STEREO|CH_BACK_CENTER, ///< 3, L+R+S | |
80 CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S | |
81 CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR | |
82 CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR | |
83 CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR | |
84 CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV | |
85 CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_FRONT_LEFT_OF_CENTER|CH_BACK_CENTER|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 6, CF+ CR+LF+ RF+LR + RR | |
86 CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR | |
8102
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87 CH_FRONT_LEFT_OF_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT|CH_BACK_LEFT|CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2+ SR1 + SR2 |
8103 | 88 CH_FRONT_LEFT_OF_CENTER|CH_FRONT_CENTER|CH_FRONT_RIGHT_OF_CENTER|CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_BACK_CENTER|CH_SIDE_RIGHT, ///< 8, CL + C+ CR + L + R + SL + S+ SR |
8100 | 89 }; |
90 | |
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91 static const int8_t dca_lfe_index[] = { |
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92 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3 |
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93 }; |
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94 |
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95 static const int8_t dca_channel_reorder_lfe[][8] = { |
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96 { 0, -1, -1, -1, -1, -1, -1, -1}, |
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97 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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98 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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99 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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100 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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101 { 2, 0, 1, -1, -1, -1, -1, -1}, |
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102 { 0, 1, 3, -1, -1, -1, -1, -1}, |
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103 { 2, 0, 1, 4, -1, -1, -1, -1}, |
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104 { 0, 1, 3, 4, -1, -1, -1, -1}, |
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105 { 2, 0, 1, 4, 5, -1, -1, -1}, |
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106 { 3, 4, 0, 1, 5, 6, -1, -1}, |
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107 { 2, 0, 1, 4, 5, 6, -1, -1}, |
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108 { 0, 6, 4, 5, 2, 3, -1, -1}, |
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109 { 4, 2, 5, 0, 1, 6, 7, -1}, |
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110 { 5, 6, 0, 1, 7, 3, 8, 4}, |
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111 { 4, 2, 5, 0, 1, 6, 8, 7}, |
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112 }; |
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113 |
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114 static const int8_t dca_channel_reorder_nolfe[][8] = { |
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115 { 0, -1, -1, -1, -1, -1, -1, -1}, |
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116 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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117 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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118 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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119 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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120 { 2, 0, 1, -1, -1, -1, -1, -1}, |
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121 { 0, 1, 2, -1, -1, -1, -1, -1}, |
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122 { 2, 0, 1, 3, -1, -1, -1, -1}, |
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123 { 0, 1, 2, 3, -1, -1, -1, -1}, |
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124 { 2, 0, 1, 3, 4, -1, -1, -1}, |
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125 { 2, 3, 0, 1, 4, 5, -1, -1}, |
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126 { 2, 0, 1, 3, 4, 5, -1, -1}, |
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127 { 0, 5, 3, 4, 1, 2, -1, -1}, |
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128 { 3, 2, 4, 0, 1, 5, 6, -1}, |
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129 { 4, 5, 0, 1, 6, 2, 7, 3}, |
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130 { 3, 2, 4, 0, 1, 5, 7, 6}, |
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131 }; |
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132 |
8100 | 133 |
4599 | 134 #define DCA_DOLBY 101 /* FIXME */ |
135 | |
136 #define DCA_CHANNEL_BITS 6 | |
137 #define DCA_CHANNEL_MASK 0x3F | |
138 | |
139 #define DCA_LFE 0x80 | |
140 | |
141 #define HEADER_SIZE 14 | |
142 | |
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143 #define DCA_MAX_FRAME_SIZE 16384 |
4599 | 144 |
145 /** Bit allocation */ | |
146 typedef struct { | |
147 int offset; ///< code values offset | |
148 int maxbits[8]; ///< max bits in VLC | |
149 int wrap; ///< wrap for get_vlc2() | |
150 VLC vlc[8]; ///< actual codes | |
151 } BitAlloc; | |
152 | |
153 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select | |
154 static BitAlloc dca_tmode; ///< transition mode VLCs | |
155 static BitAlloc dca_scalefactor; ///< scalefactor VLCs | |
156 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs | |
157 | |
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158 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx) |
4599 | 159 { |
160 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset; | |
161 } | |
162 | |
163 typedef struct { | |
164 AVCodecContext *avctx; | |
165 /* Frame header */ | |
166 int frame_type; ///< type of the current frame | |
167 int samples_deficit; ///< deficit sample count | |
168 int crc_present; ///< crc is present in the bitstream | |
169 int sample_blocks; ///< number of PCM sample blocks | |
170 int frame_size; ///< primary frame byte size | |
171 int amode; ///< audio channels arrangement | |
172 int sample_rate; ///< audio sampling rate | |
173 int bit_rate; ///< transmission bit rate | |
8077 | 174 int bit_rate_index; ///< transmission bit rate index |
4599 | 175 |
176 int downmix; ///< embedded downmix enabled | |
177 int dynrange; ///< embedded dynamic range flag | |
178 int timestamp; ///< embedded time stamp flag | |
179 int aux_data; ///< auxiliary data flag | |
180 int hdcd; ///< source material is mastered in HDCD | |
181 int ext_descr; ///< extension audio descriptor flag | |
182 int ext_coding; ///< extended coding flag | |
183 int aspf; ///< audio sync word insertion flag | |
184 int lfe; ///< low frequency effects flag | |
185 int predictor_history; ///< predictor history flag | |
186 int header_crc; ///< header crc check bytes | |
187 int multirate_inter; ///< multirate interpolator switch | |
188 int version; ///< encoder software revision | |
189 int copy_history; ///< copy history | |
190 int source_pcm_res; ///< source pcm resolution | |
191 int front_sum; ///< front sum/difference flag | |
192 int surround_sum; ///< surround sum/difference flag | |
193 int dialog_norm; ///< dialog normalisation parameter | |
194 | |
195 /* Primary audio coding header */ | |
196 int subframes; ///< number of subframes | |
6463 | 197 int total_channels; ///< number of channels including extensions |
4599 | 198 int prim_channels; ///< number of primary audio channels |
199 int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count | |
200 int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband | |
201 int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index | |
202 int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book | |
203 int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book | |
204 int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select | |
205 int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select | |
206 float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment | |
207 | |
208 /* Primary audio coding side information */ | |
209 int subsubframes; ///< number of subsubframes | |
210 int partial_samples; ///< partial subsubframe samples count | |
211 int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not) | |
212 int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs | |
213 int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index | |
214 int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients) | |
215 int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient) | |
216 int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook | |
217 int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors | |
218 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients | |
219 int dynrange_coef; ///< dynamic range coefficient | |
220 | |
221 int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands | |
222 | |
223 float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX * | |
224 2 /*history */ ]; ///< Low frequency effect data | |
225 int lfe_scale_factor; | |
226 | |
227 /* Subband samples history (for ADPCM) */ | |
228 float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; | |
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229 DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]); |
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230 float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32]; |
7737 | 231 int hist_index[DCA_PRIM_CHANNELS_MAX]; |
4599 | 232 |
233 int output; ///< type of output | |
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234 float add_bias; ///< output bias |
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235 float scale_bias; ///< output scale |
4599 | 236 |
237 DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */ | |
7726 | 238 const float *samples_chanptr[6]; |
4599 | 239 |
240 uint8_t dca_buffer[DCA_MAX_FRAME_SIZE]; | |
241 int dca_buffer_size; ///< how much data is in the dca_buffer | |
242 | |
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243 const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe |
4599 | 244 GetBitContext gb; |
245 /* Current position in DCA frame */ | |
246 int current_subframe; | |
247 int current_subsubframe; | |
248 | |
249 int debug_flag; ///< used for suppressing repeated error messages output | |
250 DSPContext dsp; | |
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251 MDCTContext imdct; |
4599 | 252 } DCAContext; |
253 | |
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254 static av_cold void dca_init_vlcs(void) |
4599 | 255 { |
6350 | 256 static int vlcs_initialized = 0; |
4599 | 257 int i, j; |
258 | |
6350 | 259 if (vlcs_initialized) |
4599 | 260 return; |
261 | |
262 dca_bitalloc_index.offset = 1; | |
5070 | 263 dca_bitalloc_index.wrap = 2; |
4599 | 264 for (i = 0; i < 5; i++) |
265 init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12, | |
266 bitalloc_12_bits[i], 1, 1, | |
267 bitalloc_12_codes[i], 2, 2, 1); | |
268 dca_scalefactor.offset = -64; | |
269 dca_scalefactor.wrap = 2; | |
270 for (i = 0; i < 5; i++) | |
271 init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129, | |
272 scales_bits[i], 1, 1, | |
273 scales_codes[i], 2, 2, 1); | |
274 dca_tmode.offset = 0; | |
275 dca_tmode.wrap = 1; | |
276 for (i = 0; i < 4; i++) | |
277 init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4, | |
278 tmode_bits[i], 1, 1, | |
279 tmode_codes[i], 2, 2, 1); | |
280 | |
281 for(i = 0; i < 10; i++) | |
282 for(j = 0; j < 7; j++){ | |
283 if(!bitalloc_codes[i][j]) break; | |
284 dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i]; | |
285 dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4); | |
286 init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j], | |
287 bitalloc_sizes[i], | |
288 bitalloc_bits[i][j], 1, 1, | |
289 bitalloc_codes[i][j], 2, 2, 1); | |
290 } | |
6350 | 291 vlcs_initialized = 1; |
4599 | 292 } |
293 | |
294 static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) | |
295 { | |
296 while(len--) | |
297 *dst++ = get_bits(gb, bits); | |
298 } | |
299 | |
300 static int dca_parse_frame_header(DCAContext * s) | |
301 { | |
302 int i, j; | |
303 static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; | |
304 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; | |
305 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; | |
306 | |
307 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
308 | |
309 /* Sync code */ | |
310 get_bits(&s->gb, 32); | |
311 | |
312 /* Frame header */ | |
313 s->frame_type = get_bits(&s->gb, 1); | |
314 s->samples_deficit = get_bits(&s->gb, 5) + 1; | |
315 s->crc_present = get_bits(&s->gb, 1); | |
316 s->sample_blocks = get_bits(&s->gb, 7) + 1; | |
317 s->frame_size = get_bits(&s->gb, 14) + 1; | |
318 if (s->frame_size < 95) | |
319 return -1; | |
320 s->amode = get_bits(&s->gb, 6); | |
321 s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)]; | |
322 if (!s->sample_rate) | |
323 return -1; | |
8078 | 324 s->bit_rate_index = get_bits(&s->gb, 5); |
8077 | 325 s->bit_rate = dca_bit_rates[s->bit_rate_index]; |
4599 | 326 if (!s->bit_rate) |
327 return -1; | |
328 | |
329 s->downmix = get_bits(&s->gb, 1); | |
330 s->dynrange = get_bits(&s->gb, 1); | |
331 s->timestamp = get_bits(&s->gb, 1); | |
332 s->aux_data = get_bits(&s->gb, 1); | |
333 s->hdcd = get_bits(&s->gb, 1); | |
334 s->ext_descr = get_bits(&s->gb, 3); | |
335 s->ext_coding = get_bits(&s->gb, 1); | |
336 s->aspf = get_bits(&s->gb, 1); | |
337 s->lfe = get_bits(&s->gb, 2); | |
338 s->predictor_history = get_bits(&s->gb, 1); | |
339 | |
340 /* TODO: check CRC */ | |
341 if (s->crc_present) | |
342 s->header_crc = get_bits(&s->gb, 16); | |
343 | |
344 s->multirate_inter = get_bits(&s->gb, 1); | |
345 s->version = get_bits(&s->gb, 4); | |
346 s->copy_history = get_bits(&s->gb, 2); | |
347 s->source_pcm_res = get_bits(&s->gb, 3); | |
348 s->front_sum = get_bits(&s->gb, 1); | |
349 s->surround_sum = get_bits(&s->gb, 1); | |
350 s->dialog_norm = get_bits(&s->gb, 4); | |
351 | |
352 /* FIXME: channels mixing levels */ | |
4893 | 353 s->output = s->amode; |
354 if(s->lfe) s->output |= DCA_LFE; | |
4599 | 355 |
356 #ifdef TRACE | |
357 av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); | |
358 av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit); | |
359 av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present); | |
360 av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n", | |
361 s->sample_blocks, s->sample_blocks * 32); | |
362 av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size); | |
363 av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n", | |
364 s->amode, dca_channels[s->amode]); | |
8061 | 365 av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n", |
366 s->sample_rate); | |
367 av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n", | |
368 s->bit_rate); | |
4599 | 369 av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix); |
370 av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange); | |
371 av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp); | |
372 av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data); | |
373 av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd); | |
374 av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr); | |
375 av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding); | |
376 av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf); | |
377 av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe); | |
378 av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n", | |
379 s->predictor_history); | |
380 av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc); | |
381 av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n", | |
382 s->multirate_inter); | |
383 av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version); | |
384 av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history); | |
385 av_log(s->avctx, AV_LOG_DEBUG, | |
386 "source pcm resolution: %i (%i bits/sample)\n", | |
387 s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]); | |
388 av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum); | |
389 av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum); | |
390 av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm); | |
391 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
392 #endif | |
393 | |
394 /* Primary audio coding header */ | |
395 s->subframes = get_bits(&s->gb, 4) + 1; | |
6463 | 396 s->total_channels = get_bits(&s->gb, 3) + 1; |
397 s->prim_channels = s->total_channels; | |
398 if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) | |
399 s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */ | |
4599 | 400 |
401 | |
402 for (i = 0; i < s->prim_channels; i++) { | |
403 s->subband_activity[i] = get_bits(&s->gb, 5) + 2; | |
404 if (s->subband_activity[i] > DCA_SUBBANDS) | |
405 s->subband_activity[i] = DCA_SUBBANDS; | |
406 } | |
407 for (i = 0; i < s->prim_channels; i++) { | |
408 s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; | |
409 if (s->vq_start_subband[i] > DCA_SUBBANDS) | |
410 s->vq_start_subband[i] = DCA_SUBBANDS; | |
411 } | |
412 get_array(&s->gb, s->joint_intensity, s->prim_channels, 3); | |
413 get_array(&s->gb, s->transient_huffman, s->prim_channels, 2); | |
414 get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3); | |
415 get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3); | |
416 | |
417 /* Get codebooks quantization indexes */ | |
418 memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); | |
419 for (j = 1; j < 11; j++) | |
420 for (i = 0; i < s->prim_channels; i++) | |
421 s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); | |
422 | |
423 /* Get scale factor adjustment */ | |
424 for (j = 0; j < 11; j++) | |
425 for (i = 0; i < s->prim_channels; i++) | |
426 s->scalefactor_adj[i][j] = 1; | |
427 | |
428 for (j = 1; j < 11; j++) | |
429 for (i = 0; i < s->prim_channels; i++) | |
430 if (s->quant_index_huffman[i][j] < thr[j]) | |
431 s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; | |
432 | |
433 if (s->crc_present) { | |
434 /* Audio header CRC check */ | |
435 get_bits(&s->gb, 16); | |
436 } | |
437 | |
438 s->current_subframe = 0; | |
439 s->current_subsubframe = 0; | |
440 | |
441 #ifdef TRACE | |
442 av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes); | |
443 av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels); | |
444 for(i = 0; i < s->prim_channels; i++){ | |
445 av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]); | |
446 av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]); | |
447 av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]); | |
448 av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]); | |
449 av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]); | |
450 av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]); | |
451 av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:"); | |
452 for (j = 0; j < 11; j++) | |
453 av_log(s->avctx, AV_LOG_DEBUG, " %i", | |
454 s->quant_index_huffman[i][j]); | |
455 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
456 av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:"); | |
457 for (j = 0; j < 11; j++) | |
458 av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]); | |
459 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
460 } | |
461 #endif | |
462 | |
463 return 0; | |
464 } | |
465 | |
466 | |
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467 static inline int get_scale(GetBitContext *gb, int level, int value) |
4599 | 468 { |
469 if (level < 5) { | |
470 /* huffman encoded */ | |
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471 value += get_bitalloc(gb, &dca_scalefactor, level); |
4599 | 472 } else if(level < 8) |
473 value = get_bits(gb, level + 1); | |
474 return value; | |
475 } | |
476 | |
477 static int dca_subframe_header(DCAContext * s) | |
478 { | |
479 /* Primary audio coding side information */ | |
480 int j, k; | |
481 | |
482 s->subsubframes = get_bits(&s->gb, 2) + 1; | |
483 s->partial_samples = get_bits(&s->gb, 3); | |
484 for (j = 0; j < s->prim_channels; j++) { | |
485 for (k = 0; k < s->subband_activity[j]; k++) | |
486 s->prediction_mode[j][k] = get_bits(&s->gb, 1); | |
487 } | |
488 | |
489 /* Get prediction codebook */ | |
490 for (j = 0; j < s->prim_channels; j++) { | |
491 for (k = 0; k < s->subband_activity[j]; k++) { | |
492 if (s->prediction_mode[j][k] > 0) { | |
493 /* (Prediction coefficient VQ address) */ | |
494 s->prediction_vq[j][k] = get_bits(&s->gb, 12); | |
495 } | |
496 } | |
497 } | |
498 | |
499 /* Bit allocation index */ | |
500 for (j = 0; j < s->prim_channels; j++) { | |
501 for (k = 0; k < s->vq_start_subband[j]; k++) { | |
502 if (s->bitalloc_huffman[j] == 6) | |
503 s->bitalloc[j][k] = get_bits(&s->gb, 5); | |
504 else if (s->bitalloc_huffman[j] == 5) | |
505 s->bitalloc[j][k] = get_bits(&s->gb, 4); | |
6463 | 506 else if (s->bitalloc_huffman[j] == 7) { |
507 av_log(s->avctx, AV_LOG_ERROR, | |
508 "Invalid bit allocation index\n"); | |
509 return -1; | |
510 } else { | |
4599 | 511 s->bitalloc[j][k] = |
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512 get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); |
4599 | 513 } |
514 | |
515 if (s->bitalloc[j][k] > 26) { | |
516 // av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n", | |
517 // j, k, s->bitalloc[j][k]); | |
518 return -1; | |
519 } | |
520 } | |
521 } | |
522 | |
523 /* Transition mode */ | |
524 for (j = 0; j < s->prim_channels; j++) { | |
525 for (k = 0; k < s->subband_activity[j]; k++) { | |
526 s->transition_mode[j][k] = 0; | |
527 if (s->subsubframes > 1 && | |
528 k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { | |
529 s->transition_mode[j][k] = | |
530 get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); | |
531 } | |
532 } | |
533 } | |
534 | |
535 for (j = 0; j < s->prim_channels; j++) { | |
6214 | 536 const uint32_t *scale_table; |
4599 | 537 int scale_sum; |
538 | |
539 memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); | |
540 | |
541 if (s->scalefactor_huffman[j] == 6) | |
6214 | 542 scale_table = scale_factor_quant7; |
4599 | 543 else |
6214 | 544 scale_table = scale_factor_quant6; |
4599 | 545 |
546 /* When huffman coded, only the difference is encoded */ | |
547 scale_sum = 0; | |
548 | |
549 for (k = 0; k < s->subband_activity[j]; k++) { | |
550 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { | |
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551 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 552 s->scale_factor[j][k][0] = scale_table[scale_sum]; |
553 } | |
554 | |
555 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { | |
556 /* Get second scale factor */ | |
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557 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 558 s->scale_factor[j][k][1] = scale_table[scale_sum]; |
559 } | |
560 } | |
561 } | |
562 | |
563 /* Joint subband scale factor codebook select */ | |
564 for (j = 0; j < s->prim_channels; j++) { | |
565 /* Transmitted only if joint subband coding enabled */ | |
566 if (s->joint_intensity[j] > 0) | |
567 s->joint_huff[j] = get_bits(&s->gb, 3); | |
568 } | |
569 | |
570 /* Scale factors for joint subband coding */ | |
571 for (j = 0; j < s->prim_channels; j++) { | |
572 int source_channel; | |
573 | |
574 /* Transmitted only if joint subband coding enabled */ | |
575 if (s->joint_intensity[j] > 0) { | |
576 int scale = 0; | |
577 source_channel = s->joint_intensity[j] - 1; | |
578 | |
579 /* When huffman coded, only the difference is encoded | |
580 * (is this valid as well for joint scales ???) */ | |
581 | |
582 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { | |
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583 scale = get_scale(&s->gb, s->joint_huff[j], 0); |
4599 | 584 scale += 64; /* bias */ |
585 s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ | |
586 } | |
587 | |
588 if (!s->debug_flag & 0x02) { | |
589 av_log(s->avctx, AV_LOG_DEBUG, | |
590 "Joint stereo coding not supported\n"); | |
591 s->debug_flag |= 0x02; | |
592 } | |
593 } | |
594 } | |
595 | |
596 /* Stereo downmix coefficients */ | |
4894 | 597 if (s->prim_channels > 2) { |
598 if(s->downmix) { | |
4895 | 599 for (j = 0; j < s->prim_channels; j++) { |
600 s->downmix_coef[j][0] = get_bits(&s->gb, 7); | |
601 s->downmix_coef[j][1] = get_bits(&s->gb, 7); | |
602 } | |
4894 | 603 } else { |
604 int am = s->amode & DCA_CHANNEL_MASK; | |
605 for (j = 0; j < s->prim_channels; j++) { | |
606 s->downmix_coef[j][0] = dca_default_coeffs[am][j][0]; | |
607 s->downmix_coef[j][1] = dca_default_coeffs[am][j][1]; | |
608 } | |
609 } | |
4599 | 610 } |
611 | |
612 /* Dynamic range coefficient */ | |
613 if (s->dynrange) | |
614 s->dynrange_coef = get_bits(&s->gb, 8); | |
615 | |
616 /* Side information CRC check word */ | |
617 if (s->crc_present) { | |
618 get_bits(&s->gb, 16); | |
619 } | |
620 | |
621 /* | |
622 * Primary audio data arrays | |
623 */ | |
624 | |
625 /* VQ encoded high frequency subbands */ | |
626 for (j = 0; j < s->prim_channels; j++) | |
627 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
628 /* 1 vector -> 32 samples */ | |
629 s->high_freq_vq[j][k] = get_bits(&s->gb, 10); | |
630 | |
631 /* Low frequency effect data */ | |
632 if (s->lfe) { | |
633 /* LFE samples */ | |
634 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
635 float lfe_scale; | |
636 | |
637 for (j = lfe_samples; j < lfe_samples * 2; j++) { | |
638 /* Signed 8 bits int */ | |
639 s->lfe_data[j] = get_sbits(&s->gb, 8); | |
640 } | |
641 | |
642 /* Scale factor index */ | |
643 s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)]; | |
644 | |
645 /* Quantization step size * scale factor */ | |
646 lfe_scale = 0.035 * s->lfe_scale_factor; | |
647 | |
648 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
649 s->lfe_data[j] *= lfe_scale; | |
650 } | |
651 | |
652 #ifdef TRACE | |
653 av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes); | |
654 av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", | |
655 s->partial_samples); | |
656 for (j = 0; j < s->prim_channels; j++) { | |
657 av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:"); | |
658 for (k = 0; k < s->subband_activity[j]; k++) | |
659 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]); | |
660 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
661 } | |
662 for (j = 0; j < s->prim_channels; j++) { | |
663 for (k = 0; k < s->subband_activity[j]; k++) | |
664 av_log(s->avctx, AV_LOG_DEBUG, | |
665 "prediction coefs: %f, %f, %f, %f\n", | |
666 (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192, | |
667 (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192, | |
668 (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192, | |
669 (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192); | |
670 } | |
671 for (j = 0; j < s->prim_channels; j++) { | |
672 av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: "); | |
673 for (k = 0; k < s->vq_start_subband[j]; k++) | |
674 av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]); | |
675 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
676 } | |
677 for (j = 0; j < s->prim_channels; j++) { | |
678 av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:"); | |
679 for (k = 0; k < s->subband_activity[j]; k++) | |
680 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]); | |
681 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
682 } | |
683 for (j = 0; j < s->prim_channels; j++) { | |
684 av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:"); | |
685 for (k = 0; k < s->subband_activity[j]; k++) { | |
686 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) | |
687 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]); | |
688 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) | |
689 av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]); | |
690 } | |
691 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
692 } | |
693 for (j = 0; j < s->prim_channels; j++) { | |
694 if (s->joint_intensity[j] > 0) { | |
5069 | 695 int source_channel = s->joint_intensity[j] - 1; |
4599 | 696 av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n"); |
697 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) | |
698 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]); | |
699 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
700 } | |
701 } | |
702 if (s->prim_channels > 2 && s->downmix) { | |
703 av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); | |
704 for (j = 0; j < s->prim_channels; j++) { | |
705 av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]); | |
706 av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]); | |
707 } | |
708 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
709 } | |
710 for (j = 0; j < s->prim_channels; j++) | |
711 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
712 av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]); | |
713 if(s->lfe){ | |
5069 | 714 int lfe_samples = 2 * s->lfe * s->subsubframes; |
4599 | 715 av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n"); |
716 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
717 av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]); | |
718 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
719 } | |
720 #endif | |
721 | |
722 return 0; | |
723 } | |
724 | |
725 static void qmf_32_subbands(DCAContext * s, int chans, | |
726 float samples_in[32][8], float *samples_out, | |
727 float scale, float bias) | |
728 { | |
5974
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729 const float *prCoeff; |
7738
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730 int i, j; |
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731 DECLARE_ALIGNED_16(float, raXin[32]); |
4599 | 732 |
7737 | 733 int hist_index= s->hist_index[chans]; |
4599 | 734 float *subband_fir_hist2 = s->subband_fir_noidea[chans]; |
735 | |
7738
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736 int subindex; |
4599 | 737 |
7738
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738 scale *= sqrt(1/8.0); |
4599 | 739 |
740 /* Select filter */ | |
741 if (!s->multirate_inter) /* Non-perfect reconstruction */ | |
5974
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742 prCoeff = fir_32bands_nonperfect; |
4599 | 743 else /* Perfect reconstruction */ |
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744 prCoeff = fir_32bands_perfect; |
4599 | 745 |
746 /* Reconstructed channel sample index */ | |
747 for (subindex = 0; subindex < 8; subindex++) { | |
7737 | 748 float *subband_fir_hist = s->subband_fir_hist[chans] + hist_index; |
4599 | 749 /* Load in one sample from each subband and clear inactive subbands */ |
7738
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750 for (i = 0; i < s->subband_activity[chans]; i++){ |
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751 if((i-1)&2) raXin[i] = -samples_in[i][subindex]; |
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752 else raXin[i] = samples_in[i][subindex]; |
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753 } |
4599 | 754 for (; i < 32; i++) |
755 raXin[i] = 0.0; | |
756 | |
7738
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757 ff_imdct_half(&s->imdct, subband_fir_hist, raXin); |
4599 | 758 |
759 /* Multiply by filter coefficients */ | |
7738
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760 for (i = 0; i < 16; i++){ |
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761 float a= subband_fir_hist2[i ]; |
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762 float b= subband_fir_hist2[i+16]; |
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763 float c= 0; |
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764 float d= 0; |
7737 | 765 for (j = 0; j < 512-hist_index; j += 64){ |
7738
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|
766 a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j]); |
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767 b += prCoeff[i+j+16]*( subband_fir_hist[ i+j]); |
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768 c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j]); |
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769 d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j]); |
4599 | 770 } |
7737 | 771 for ( ; j < 512; j += 64){ |
7738
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772 a += prCoeff[i+j ]*(-subband_fir_hist[15-i+j-512]); |
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773 b += prCoeff[i+j+16]*( subband_fir_hist[ i+j-512]); |
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774 c += prCoeff[i+j+32]*( subband_fir_hist[16+i+j-512]); |
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775 d += prCoeff[i+j+48]*( subband_fir_hist[31-i+j-512]); |
7737 | 776 } |
7738
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777 samples_out[i ] = a * scale + bias; |
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778 samples_out[i+16] = b * scale + bias; |
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779 subband_fir_hist2[i ] = c; |
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780 subband_fir_hist2[i+16] = d; |
7730
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781 } |
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782 samples_out+= 32; |
4599 | 783 |
7737 | 784 hist_index = (hist_index-32)&511; |
4599 | 785 } |
7737 | 786 s->hist_index[chans]= hist_index; |
4599 | 787 } |
788 | |
789 static void lfe_interpolation_fir(int decimation_select, | |
790 int num_deci_sample, float *samples_in, | |
791 float *samples_out, float scale, | |
792 float bias) | |
793 { | |
794 /* samples_in: An array holding decimated samples. | |
795 * Samples in current subframe starts from samples_in[0], | |
796 * while samples_in[-1], samples_in[-2], ..., stores samples | |
797 * from last subframe as history. | |
798 * | |
799 * samples_out: An array holding interpolated samples | |
800 */ | |
801 | |
802 int decifactor, k, j; | |
803 const float *prCoeff; | |
804 | |
805 int interp_index = 0; /* Index to the interpolated samples */ | |
806 int deciindex; | |
807 | |
808 /* Select decimation filter */ | |
809 if (decimation_select == 1) { | |
810 decifactor = 128; | |
811 prCoeff = lfe_fir_128; | |
812 } else { | |
813 decifactor = 64; | |
814 prCoeff = lfe_fir_64; | |
815 } | |
816 /* Interpolation */ | |
817 for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { | |
818 /* One decimated sample generates decifactor interpolated ones */ | |
819 for (k = 0; k < decifactor; k++) { | |
820 float rTmp = 0.0; | |
821 //FIXME the coeffs are symetric, fix that | |
822 for (j = 0; j < 512 / decifactor; j++) | |
823 rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor]; | |
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824 samples_out[interp_index++] = (rTmp * scale) + bias; |
4599 | 825 } |
826 } | |
827 } | |
828 | |
829 /* downmixing routines */ | |
4894 | 830 #define MIX_REAR1(samples, si1, rs, coef) \ |
831 samples[i] += samples[si1] * coef[rs][0]; \ | |
832 samples[i+256] += samples[si1] * coef[rs][1]; | |
4599 | 833 |
4894 | 834 #define MIX_REAR2(samples, si1, si2, rs, coef) \ |
835 samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ | |
836 samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; | |
4599 | 837 |
4894 | 838 #define MIX_FRONT3(samples, coef) \ |
4599 | 839 t = samples[i]; \ |
4894 | 840 samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \ |
841 samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1]; | |
4599 | 842 |
843 #define DOWNMIX_TO_STEREO(op1, op2) \ | |
844 for(i = 0; i < 256; i++){ \ | |
845 op1 \ | |
846 op2 \ | |
847 } | |
848 | |
4894 | 849 static void dca_downmix(float *samples, int srcfmt, |
850 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]) | |
4599 | 851 { |
852 int i; | |
853 float t; | |
4894 | 854 float coef[DCA_PRIM_CHANNELS_MAX][2]; |
855 | |
856 for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) { | |
857 coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]]; | |
858 coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]]; | |
859 } | |
4599 | 860 |
861 switch (srcfmt) { | |
862 case DCA_MONO: | |
863 case DCA_CHANNEL: | |
864 case DCA_STEREO_TOTAL: | |
865 case DCA_STEREO_SUMDIFF: | |
866 case DCA_4F2R: | |
867 av_log(NULL, 0, "Not implemented!\n"); | |
868 break; | |
869 case DCA_STEREO: | |
870 break; | |
871 case DCA_3F: | |
4894 | 872 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),); |
4599 | 873 break; |
874 case DCA_2F1R: | |
4894 | 875 DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),); |
4599 | 876 break; |
877 case DCA_3F1R: | |
4894 | 878 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
879 MIX_REAR1(samples, i + 768, 3, coef)); | |
4599 | 880 break; |
881 case DCA_2F2R: | |
4894 | 882 DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),); |
4599 | 883 break; |
884 case DCA_3F2R: | |
4894 | 885 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
886 MIX_REAR2(samples, i + 768, i + 1024, 3, coef)); | |
4599 | 887 break; |
888 } | |
889 } | |
890 | |
891 | |
892 /* Very compact version of the block code decoder that does not use table | |
893 * look-up but is slightly slower */ | |
894 static int decode_blockcode(int code, int levels, int *values) | |
895 { | |
896 int i; | |
897 int offset = (levels - 1) >> 1; | |
898 | |
899 for (i = 0; i < 4; i++) { | |
900 values[i] = (code % levels) - offset; | |
901 code /= levels; | |
902 } | |
903 | |
904 if (code == 0) | |
905 return 0; | |
906 else { | |
907 av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n"); | |
908 return -1; | |
909 } | |
910 } | |
911 | |
912 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; | |
913 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; | |
914 | |
915 static int dca_subsubframe(DCAContext * s) | |
916 { | |
917 int k, l; | |
918 int subsubframe = s->current_subsubframe; | |
919 | |
6214 | 920 const float *quant_step_table; |
4599 | 921 |
922 /* FIXME */ | |
923 float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; | |
924 | |
925 /* | |
926 * Audio data | |
927 */ | |
928 | |
929 /* Select quantization step size table */ | |
8077 | 930 if (s->bit_rate_index == 0x1f) |
6214 | 931 quant_step_table = lossless_quant_d; |
4599 | 932 else |
6214 | 933 quant_step_table = lossy_quant_d; |
4599 | 934 |
935 for (k = 0; k < s->prim_channels; k++) { | |
936 for (l = 0; l < s->vq_start_subband[k]; l++) { | |
937 int m; | |
938 | |
939 /* Select the mid-tread linear quantizer */ | |
940 int abits = s->bitalloc[k][l]; | |
941 | |
942 float quant_step_size = quant_step_table[abits]; | |
943 float rscale; | |
944 | |
945 /* | |
946 * Determine quantization index code book and its type | |
947 */ | |
948 | |
949 /* Select quantization index code book */ | |
950 int sel = s->quant_index_huffman[k][abits]; | |
951 | |
952 /* | |
953 * Extract bits from the bit stream | |
954 */ | |
955 if(!abits){ | |
956 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0])); | |
957 }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){ | |
958 if(abits <= 7){ | |
959 /* Block code */ | |
960 int block_code1, block_code2, size, levels; | |
961 int block[8]; | |
962 | |
963 size = abits_sizes[abits-1]; | |
964 levels = abits_levels[abits-1]; | |
965 | |
966 block_code1 = get_bits(&s->gb, size); | |
967 /* FIXME Should test return value */ | |
968 decode_blockcode(block_code1, levels, block); | |
969 block_code2 = get_bits(&s->gb, size); | |
970 decode_blockcode(block_code2, levels, &block[4]); | |
971 for (m = 0; m < 8; m++) | |
972 subband_samples[k][l][m] = block[m]; | |
973 }else{ | |
974 /* no coding */ | |
975 for (m = 0; m < 8; m++) | |
976 subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3); | |
977 } | |
978 }else{ | |
979 /* Huffman coded */ | |
980 for (m = 0; m < 8; m++) | |
981 subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); | |
982 } | |
983 | |
984 /* Deal with transients */ | |
985 if (s->transition_mode[k][l] && | |
986 subsubframe >= s->transition_mode[k][l]) | |
987 rscale = quant_step_size * s->scale_factor[k][l][1]; | |
988 else | |
989 rscale = quant_step_size * s->scale_factor[k][l][0]; | |
990 | |
991 rscale *= s->scalefactor_adj[k][sel]; | |
992 | |
993 for (m = 0; m < 8; m++) | |
994 subband_samples[k][l][m] *= rscale; | |
995 | |
996 /* | |
997 * Inverse ADPCM if in prediction mode | |
998 */ | |
999 if (s->prediction_mode[k][l]) { | |
1000 int n; | |
1001 for (m = 0; m < 8; m++) { | |
1002 for (n = 1; n <= 4; n++) | |
1003 if (m >= n) | |
1004 subband_samples[k][l][m] += | |
1005 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
1006 subband_samples[k][l][m - n] / 8192); | |
1007 else if (s->predictor_history) | |
1008 subband_samples[k][l][m] += | |
1009 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
1010 s->subband_samples_hist[k][l][m - n + | |
1011 4] / 8192); | |
1012 } | |
1013 } | |
1014 } | |
1015 | |
1016 /* | |
1017 * Decode VQ encoded high frequencies | |
1018 */ | |
1019 for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { | |
1020 /* 1 vector -> 32 samples but we only need the 8 samples | |
1021 * for this subsubframe. */ | |
1022 int m; | |
1023 | |
1024 if (!s->debug_flag & 0x01) { | |
1025 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); | |
1026 s->debug_flag |= 0x01; | |
1027 } | |
1028 | |
1029 for (m = 0; m < 8; m++) { | |
1030 subband_samples[k][l][m] = | |
1031 high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 + | |
1032 m] | |
1033 * (float) s->scale_factor[k][l][0] / 16.0; | |
1034 } | |
1035 } | |
1036 } | |
1037 | |
1038 /* Check for DSYNC after subsubframe */ | |
1039 if (s->aspf || subsubframe == s->subsubframes - 1) { | |
1040 if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */ | |
1041 #ifdef TRACE | |
1042 av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n"); | |
1043 #endif | |
1044 } else { | |
1045 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); | |
1046 } | |
1047 } | |
1048 | |
1049 /* Backup predictor history for adpcm */ | |
1050 for (k = 0; k < s->prim_channels; k++) | |
1051 for (l = 0; l < s->vq_start_subband[k]; l++) | |
1052 memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4], | |
1053 4 * sizeof(subband_samples[0][0][0])); | |
1054 | |
1055 /* 32 subbands QMF */ | |
1056 for (k = 0; k < s->prim_channels; k++) { | |
1057 /* static float pcm_to_double[8] = | |
1058 {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/ | |
8148
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diff
changeset
|
1059 qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]], |
8062
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banan
parents:
8061
diff
changeset
|
1060 M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ , |
17aeecee2a97
Fix dca decoder with non simd float2int16 conversion
banan
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diff
changeset
|
1061 s->add_bias ); |
4599 | 1062 } |
1063 | |
1064 /* Down mixing */ | |
1065 | |
1066 if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) { | |
4894 | 1067 dca_downmix(s->samples, s->amode, s->downmix_coef); |
4599 | 1068 } |
1069 | |
1070 /* Generate LFE samples for this subsubframe FIXME!!! */ | |
1071 if (s->output & DCA_LFE) { | |
1072 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
1073 | |
1074 lfe_interpolation_fir(s->lfe, 2 * s->lfe, | |
1075 s->lfe_data + lfe_samples + | |
1076 2 * s->lfe * subsubframe, | |
8148
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Proper channel output reordering for the dca decoder.
banan
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8126
diff
changeset
|
1077 &s->samples[256 * dca_lfe_index[s->amode]], |
8062
17aeecee2a97
Fix dca decoder with non simd float2int16 conversion
banan
parents:
8061
diff
changeset
|
1078 (1.0/256.0)*s->scale_bias, s->add_bias); |
4599 | 1079 /* Outputs 20bits pcm samples */ |
1080 } | |
1081 | |
1082 return 0; | |
1083 } | |
1084 | |
1085 | |
1086 static int dca_subframe_footer(DCAContext * s) | |
1087 { | |
1088 int aux_data_count = 0, i; | |
1089 int lfe_samples; | |
1090 | |
1091 /* | |
1092 * Unpack optional information | |
1093 */ | |
1094 | |
1095 if (s->timestamp) | |
1096 get_bits(&s->gb, 32); | |
1097 | |
1098 if (s->aux_data) | |
1099 aux_data_count = get_bits(&s->gb, 6); | |
1100 | |
1101 for (i = 0; i < aux_data_count; i++) | |
1102 get_bits(&s->gb, 8); | |
1103 | |
1104 if (s->crc_present && (s->downmix || s->dynrange)) | |
1105 get_bits(&s->gb, 16); | |
1106 | |
1107 lfe_samples = 2 * s->lfe * s->subsubframes; | |
1108 for (i = 0; i < lfe_samples; i++) { | |
1109 s->lfe_data[i] = s->lfe_data[i + lfe_samples]; | |
1110 } | |
1111 | |
1112 return 0; | |
1113 } | |
1114 | |
1115 /** | |
1116 * Decode a dca frame block | |
1117 * | |
1118 * @param s pointer to the DCAContext | |
1119 */ | |
1120 | |
1121 static int dca_decode_block(DCAContext * s) | |
1122 { | |
1123 | |
1124 /* Sanity check */ | |
1125 if (s->current_subframe >= s->subframes) { | |
1126 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", | |
1127 s->current_subframe, s->subframes); | |
1128 return -1; | |
1129 } | |
1130 | |
1131 if (!s->current_subsubframe) { | |
1132 #ifdef TRACE | |
1133 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); | |
1134 #endif | |
1135 /* Read subframe header */ | |
1136 if (dca_subframe_header(s)) | |
1137 return -1; | |
1138 } | |
1139 | |
1140 /* Read subsubframe */ | |
1141 #ifdef TRACE | |
1142 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); | |
1143 #endif | |
1144 if (dca_subsubframe(s)) | |
1145 return -1; | |
1146 | |
1147 /* Update state */ | |
1148 s->current_subsubframe++; | |
1149 if (s->current_subsubframe >= s->subsubframes) { | |
1150 s->current_subsubframe = 0; | |
1151 s->current_subframe++; | |
1152 } | |
1153 if (s->current_subframe >= s->subframes) { | |
1154 #ifdef TRACE | |
1155 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); | |
1156 #endif | |
1157 /* Read subframe footer */ | |
1158 if (dca_subframe_footer(s)) | |
1159 return -1; | |
1160 } | |
1161 | |
1162 return 0; | |
1163 } | |
1164 | |
1165 /** | |
1166 * Convert bitstream to one representation based on sync marker | |
1167 */ | |
6214 | 1168 static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst, |
4599 | 1169 int max_size) |
1170 { | |
1171 uint32_t mrk; | |
1172 int i, tmp; | |
6214 | 1173 const uint16_t *ssrc = (const uint16_t *) src; |
1174 uint16_t *sdst = (uint16_t *) dst; | |
4599 | 1175 PutBitContext pb; |
1176 | |
5027 | 1177 if((unsigned)src_size > (unsigned)max_size) { |
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1178 // av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n"); |
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1179 // return -1; |
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1180 src_size = max_size; |
5027 | 1181 } |
4883 | 1182 |
4599 | 1183 mrk = AV_RB32(src); |
1184 switch (mrk) { | |
1185 case DCA_MARKER_RAW_BE: | |
7671 | 1186 memcpy(dst, src, src_size); |
1187 return src_size; | |
4599 | 1188 case DCA_MARKER_RAW_LE: |
7671 | 1189 for (i = 0; i < (src_size + 1) >> 1; i++) |
4599 | 1190 *sdst++ = bswap_16(*ssrc++); |
7671 | 1191 return src_size; |
4599 | 1192 case DCA_MARKER_14B_BE: |
1193 case DCA_MARKER_14B_LE: | |
1194 init_put_bits(&pb, dst, max_size); | |
1195 for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) { | |
1196 tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF; | |
1197 put_bits(&pb, 14, tmp); | |
1198 } | |
1199 flush_put_bits(&pb); | |
1200 return (put_bits_count(&pb) + 7) >> 3; | |
1201 default: | |
1202 return -1; | |
1203 } | |
1204 } | |
1205 | |
1206 /** | |
1207 * Main frame decoding function | |
1208 * FIXME add arguments | |
1209 */ | |
1210 static int dca_decode_frame(AVCodecContext * avctx, | |
1211 void *data, int *data_size, | |
6214 | 1212 const uint8_t * buf, int buf_size) |
4599 | 1213 { |
1214 | |
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1215 int i; |
4599 | 1216 int16_t *samples = data; |
1217 DCAContext *s = avctx->priv_data; | |
1218 int channels; | |
1219 | |
1220 | |
1221 s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE); | |
1222 if (s->dca_buffer_size == -1) { | |
5027 | 1223 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); |
4599 | 1224 return -1; |
1225 } | |
1226 | |
1227 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
1228 if (dca_parse_frame_header(s) < 0) { | |
1229 //seems like the frame is corrupt, try with the next one | |
5645 | 1230 *data_size=0; |
4599 | 1231 return buf_size; |
1232 } | |
1233 //set AVCodec values with parsed data | |
1234 avctx->sample_rate = s->sample_rate; | |
1235 avctx->bit_rate = s->bit_rate; | |
1236 | |
4893 | 1237 channels = s->prim_channels + !!s->lfe; |
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1238 |
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1239 if (s->amode<16) { |
8100 | 1240 avctx->channel_layout = dca_core_channel_layout[s->amode]; |
1241 | |
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1242 if (s->lfe) { |
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1243 avctx->channel_layout |= CH_LOW_FREQUENCY; |
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1244 s->channel_order_tab = dca_channel_reorder_lfe[s->amode]; |
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1245 } else |
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1246 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode]; |
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1247 |
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1248 if(avctx->request_channels == 2 && s->prim_channels > 2) { |
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1249 channels = 2; |
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1250 s->output = DCA_STEREO; |
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1251 avctx->channel_layout = CH_LAYOUT_STEREO; |
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1252 } |
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1253 } else { |
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1254 av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode); |
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1255 return -1; |
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1256 } |
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1257 |
4893 | 1258 |
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1259 /* There is nothing that prevents a dts frame to change channel configuration |
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1260 but FFmpeg doesn't support that so only set the channels if it is previously |
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1261 unset. Ideally during the first probe for channels the crc should be checked |
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1262 and only set avctx->channels when the crc is ok. Right now the decoder could |
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1263 set the channels based on a broken first frame.*/ |
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1264 if (!avctx->channels) |
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1265 avctx->channels = channels; |
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1266 |
4599 | 1267 if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) |
1268 return -1; | |
7725 | 1269 *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels; |
4599 | 1270 for (i = 0; i < (s->sample_blocks / 8); i++) { |
1271 dca_decode_block(s); | |
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1272 s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); |
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1273 samples += 256 * channels; |
4599 | 1274 } |
1275 | |
1276 return buf_size; | |
1277 } | |
1278 | |
1279 | |
1280 | |
1281 /** | |
1282 * DCA initialization | |
1283 * | |
1284 * @param avctx pointer to the AVCodecContext | |
1285 */ | |
1286 | |
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1287 static av_cold int dca_decode_init(AVCodecContext * avctx) |
4599 | 1288 { |
1289 DCAContext *s = avctx->priv_data; | |
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1290 int i; |
4599 | 1291 |
1292 s->avctx = avctx; | |
1293 dca_init_vlcs(); | |
1294 | |
1295 dsputil_init(&s->dsp, avctx); | |
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1296 ff_mdct_init(&s->imdct, 6, 1); |
6120 | 1297 |
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1298 for(i = 0; i < 6; i++) |
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1299 s->samples_chanptr[i] = s->samples + i * 256; |
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1300 avctx->sample_fmt = SAMPLE_FMT_S16; |
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1301 |
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1302 if(s->dsp.float_to_int16 == ff_float_to_int16_c) { |
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1303 s->add_bias = 385.0f; |
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1304 s->scale_bias = 1.0 / 32768.0; |
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1305 } else { |
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1306 s->add_bias = 0.0f; |
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1307 s->scale_bias = 1.0; |
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1308 |
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1309 /* allow downmixing to stereo */ |
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1310 if (avctx->channels > 0 && avctx->request_channels < avctx->channels && |
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1311 avctx->request_channels == 2) { |
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1312 avctx->channels = avctx->request_channels; |
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1313 } |
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1314 } |
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1315 |
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1316 |
4599 | 1317 return 0; |
1318 } | |
1319 | |
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1320 static av_cold int dca_decode_end(AVCodecContext * avctx) |
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1321 { |
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1322 DCAContext *s = avctx->priv_data; |
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1323 ff_mdct_end(&s->imdct); |
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1324 return 0; |
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1325 } |
4599 | 1326 |
1327 AVCodec dca_decoder = { | |
1328 .name = "dca", | |
1329 .type = CODEC_TYPE_AUDIO, | |
1330 .id = CODEC_ID_DTS, | |
1331 .priv_data_size = sizeof(DCAContext), | |
1332 .init = dca_decode_init, | |
1333 .decode = dca_decode_frame, | |
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1334 .close = dca_decode_end, |
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e943e1409077
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1335 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), |
4599 | 1336 }; |