Mercurial > libavcodec.hg
annotate dca.c @ 10720:bcc62edd9b41 libavcodec
16-bit Interplay video does not need palette control
author | kostya |
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date | Sun, 27 Dec 2009 14:07:21 +0000 |
parents | fedf07e14bee |
children | 34a65026fa06 |
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 /** | |
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26 * @file libavcodec/dca.c |
4599 | 27 */ |
28 | |
29 #include <math.h> | |
30 #include <stddef.h> | |
31 #include <stdio.h> | |
32 | |
33 #include "avcodec.h" | |
34 #include "dsputil.h" | |
9428 | 35 #include "get_bits.h" |
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36 #include "put_bits.h" |
4599 | 37 #include "dcadata.h" |
38 #include "dcahuff.h" | |
4899 | 39 #include "dca.h" |
10467 | 40 #include "synth_filter.h" |
4599 | 41 |
42 //#define TRACE | |
43 | |
44 #define DCA_PRIM_CHANNELS_MAX (5) | |
45 #define DCA_SUBBANDS (32) | |
46 #define DCA_ABITS_MAX (32) /* Should be 28 */ | |
47 #define DCA_SUBSUBFAMES_MAX (4) | |
48 #define DCA_LFE_MAX (3) | |
49 | |
50 enum DCAMode { | |
51 DCA_MONO = 0, | |
52 DCA_CHANNEL, | |
53 DCA_STEREO, | |
54 DCA_STEREO_SUMDIFF, | |
55 DCA_STEREO_TOTAL, | |
56 DCA_3F, | |
57 DCA_2F1R, | |
58 DCA_3F1R, | |
59 DCA_2F2R, | |
60 DCA_3F2R, | |
61 DCA_4F2R | |
62 }; | |
63 | |
8100 | 64 /* Tables for mapping dts channel configurations to libavcodec multichannel api. |
65 * Some compromises have been made for special configurations. Most configurations | |
66 * are never used so complete accuracy is not needed. | |
67 * | |
68 * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead. | |
8126 | 69 * S -> side, when both rear and back are configured move one of them to the side channel |
8100 | 70 * OV -> center back |
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71 * All 2 channel configurations -> CH_LAYOUT_STEREO |
8100 | 72 */ |
73 | |
74 static const int64_t dca_core_channel_layout[] = { | |
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75 CH_FRONT_CENTER, ///< 1, A |
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76 CH_LAYOUT_STEREO, ///< 2, A + B (dual mono) |
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77 CH_LAYOUT_STEREO, ///< 2, L + R (stereo) |
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78 CH_LAYOUT_STEREO, ///< 2, (L+R) + (L-R) (sum-difference) |
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79 CH_LAYOUT_STEREO, ///< 2, LT +RT (left and right total) |
8103 | 80 CH_LAYOUT_STEREO|CH_FRONT_CENTER, ///< 3, C+L+R |
81 CH_LAYOUT_STEREO|CH_BACK_CENTER, ///< 3, L+R+S | |
82 CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 4, C + L + R+ S | |
83 CH_LAYOUT_STEREO|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 4, L + R +SL+ SR | |
84 CH_LAYOUT_STEREO|CH_FRONT_CENTER|CH_SIDE_LEFT|CH_SIDE_RIGHT, ///< 5, C + L + R+ SL+SR | |
85 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 | |
86 CH_LAYOUT_STEREO|CH_BACK_LEFT|CH_BACK_RIGHT|CH_FRONT_CENTER|CH_BACK_CENTER, ///< 6, C + L + R+ LR + RR + OV | |
87 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 | |
88 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 | |
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89 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 | 90 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 | 91 }; |
92 | |
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93 static const int8_t dca_lfe_index[] = { |
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94 1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3 |
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95 }; |
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96 |
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97 static const int8_t dca_channel_reorder_lfe[][8] = { |
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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 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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102 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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103 { 2, 0, 1, -1, -1, -1, -1, -1}, |
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104 { 0, 1, 3, -1, -1, -1, -1, -1}, |
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105 { 2, 0, 1, 4, -1, -1, -1, -1}, |
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106 { 0, 1, 3, 4, -1, -1, -1, -1}, |
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107 { 2, 0, 1, 4, 5, -1, -1, -1}, |
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108 { 3, 4, 0, 1, 5, 6, -1, -1}, |
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109 { 2, 0, 1, 4, 5, 6, -1, -1}, |
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110 { 0, 6, 4, 5, 2, 3, -1, -1}, |
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111 { 4, 2, 5, 0, 1, 6, 7, -1}, |
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112 { 5, 6, 0, 1, 7, 3, 8, 4}, |
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113 { 4, 2, 5, 0, 1, 6, 8, 7}, |
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114 }; |
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115 |
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116 static const int8_t dca_channel_reorder_nolfe[][8] = { |
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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 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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121 { 0, 1, -1, -1, -1, -1, -1, -1}, |
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122 { 2, 0, 1, -1, -1, -1, -1, -1}, |
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123 { 0, 1, 2, -1, -1, -1, -1, -1}, |
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124 { 2, 0, 1, 3, -1, -1, -1, -1}, |
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125 { 0, 1, 2, 3, -1, -1, -1, -1}, |
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126 { 2, 0, 1, 3, 4, -1, -1, -1}, |
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127 { 2, 3, 0, 1, 4, 5, -1, -1}, |
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128 { 2, 0, 1, 3, 4, 5, -1, -1}, |
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129 { 0, 5, 3, 4, 1, 2, -1, -1}, |
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130 { 3, 2, 4, 0, 1, 5, 6, -1}, |
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131 { 4, 5, 0, 1, 6, 2, 7, 3}, |
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132 { 3, 2, 4, 0, 1, 5, 7, 6}, |
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133 }; |
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134 |
8100 | 135 |
4599 | 136 #define DCA_DOLBY 101 /* FIXME */ |
137 | |
138 #define DCA_CHANNEL_BITS 6 | |
139 #define DCA_CHANNEL_MASK 0x3F | |
140 | |
141 #define DCA_LFE 0x80 | |
142 | |
143 #define HEADER_SIZE 14 | |
144 | |
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145 #define DCA_MAX_FRAME_SIZE 16384 |
4599 | 146 |
147 /** Bit allocation */ | |
148 typedef struct { | |
149 int offset; ///< code values offset | |
150 int maxbits[8]; ///< max bits in VLC | |
151 int wrap; ///< wrap for get_vlc2() | |
152 VLC vlc[8]; ///< actual codes | |
153 } BitAlloc; | |
154 | |
155 static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select | |
156 static BitAlloc dca_tmode; ///< transition mode VLCs | |
157 static BitAlloc dca_scalefactor; ///< scalefactor VLCs | |
158 static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs | |
159 | |
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160 static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, int idx) |
4599 | 161 { |
162 return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ba->offset; | |
163 } | |
164 | |
165 typedef struct { | |
166 AVCodecContext *avctx; | |
167 /* Frame header */ | |
168 int frame_type; ///< type of the current frame | |
169 int samples_deficit; ///< deficit sample count | |
170 int crc_present; ///< crc is present in the bitstream | |
171 int sample_blocks; ///< number of PCM sample blocks | |
172 int frame_size; ///< primary frame byte size | |
173 int amode; ///< audio channels arrangement | |
174 int sample_rate; ///< audio sampling rate | |
175 int bit_rate; ///< transmission bit rate | |
8077 | 176 int bit_rate_index; ///< transmission bit rate index |
4599 | 177 |
178 int downmix; ///< embedded downmix enabled | |
179 int dynrange; ///< embedded dynamic range flag | |
180 int timestamp; ///< embedded time stamp flag | |
181 int aux_data; ///< auxiliary data flag | |
182 int hdcd; ///< source material is mastered in HDCD | |
183 int ext_descr; ///< extension audio descriptor flag | |
184 int ext_coding; ///< extended coding flag | |
185 int aspf; ///< audio sync word insertion flag | |
186 int lfe; ///< low frequency effects flag | |
187 int predictor_history; ///< predictor history flag | |
188 int header_crc; ///< header crc check bytes | |
189 int multirate_inter; ///< multirate interpolator switch | |
190 int version; ///< encoder software revision | |
191 int copy_history; ///< copy history | |
192 int source_pcm_res; ///< source pcm resolution | |
193 int front_sum; ///< front sum/difference flag | |
194 int surround_sum; ///< surround sum/difference flag | |
195 int dialog_norm; ///< dialog normalisation parameter | |
196 | |
197 /* Primary audio coding header */ | |
198 int subframes; ///< number of subframes | |
6463 | 199 int total_channels; ///< number of channels including extensions |
4599 | 200 int prim_channels; ///< number of primary audio channels |
201 int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count | |
202 int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband | |
203 int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index | |
204 int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book | |
205 int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book | |
206 int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select | |
207 int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select | |
208 float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment | |
209 | |
210 /* Primary audio coding side information */ | |
211 int subsubframes; ///< number of subsubframes | |
212 int partial_samples; ///< partial subsubframe samples count | |
213 int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not) | |
214 int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs | |
215 int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index | |
216 int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients) | |
217 int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient) | |
218 int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook | |
219 int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors | |
220 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients | |
221 int dynrange_coef; ///< dynamic range coefficient | |
222 | |
223 int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands | |
224 | |
225 float lfe_data[2 * DCA_SUBSUBFAMES_MAX * DCA_LFE_MAX * | |
226 2 /*history */ ]; ///< Low frequency effect data | |
227 int lfe_scale_factor; | |
228 | |
229 /* Subband samples history (for ADPCM) */ | |
230 float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; | |
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231 DECLARE_ALIGNED_16(float, subband_fir_hist[DCA_PRIM_CHANNELS_MAX][512]); |
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232 float subband_fir_noidea[DCA_PRIM_CHANNELS_MAX][32]; |
7737 | 233 int hist_index[DCA_PRIM_CHANNELS_MAX]; |
10152 | 234 DECLARE_ALIGNED_16(float, raXin[32]); |
4599 | 235 |
236 int output; ///< type of output | |
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237 float add_bias; ///< output bias |
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238 float scale_bias; ///< output scale |
4599 | 239 |
240 DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */ | |
7726 | 241 const float *samples_chanptr[6]; |
4599 | 242 |
243 uint8_t dca_buffer[DCA_MAX_FRAME_SIZE]; | |
244 int dca_buffer_size; ///< how much data is in the dca_buffer | |
245 | |
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246 const int8_t* channel_order_tab; ///< channel reordering table, lfe and non lfe |
4599 | 247 GetBitContext gb; |
248 /* Current position in DCA frame */ | |
249 int current_subframe; | |
250 int current_subsubframe; | |
251 | |
252 int debug_flag; ///< used for suppressing repeated error messages output | |
253 DSPContext dsp; | |
10199 | 254 FFTContext imdct; |
4599 | 255 } DCAContext; |
256 | |
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257 static const uint16_t dca_vlc_offs[] = { |
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258 0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364, |
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259 5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508, |
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260 5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564, |
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261 7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240, |
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262 12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264, |
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263 18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622, |
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264 }; |
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265 |
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266 static av_cold void dca_init_vlcs(void) |
4599 | 267 { |
6350 | 268 static int vlcs_initialized = 0; |
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269 int i, j, c = 14; |
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270 static VLC_TYPE dca_table[23622][2]; |
4599 | 271 |
6350 | 272 if (vlcs_initialized) |
4599 | 273 return; |
274 | |
275 dca_bitalloc_index.offset = 1; | |
5070 | 276 dca_bitalloc_index.wrap = 2; |
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277 for (i = 0; i < 5; i++) { |
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278 dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]]; |
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279 dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i]; |
4599 | 280 init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12, |
281 bitalloc_12_bits[i], 1, 1, | |
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282 bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); |
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283 } |
4599 | 284 dca_scalefactor.offset = -64; |
285 dca_scalefactor.wrap = 2; | |
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286 for (i = 0; i < 5; i++) { |
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287 dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]]; |
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288 dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5]; |
4599 | 289 init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129, |
290 scales_bits[i], 1, 1, | |
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291 scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); |
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292 } |
4599 | 293 dca_tmode.offset = 0; |
294 dca_tmode.wrap = 1; | |
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295 for (i = 0; i < 4; i++) { |
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296 dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]]; |
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297 dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10]; |
4599 | 298 init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4, |
299 tmode_bits[i], 1, 1, | |
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300 tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); |
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301 } |
4599 | 302 |
303 for(i = 0; i < 10; i++) | |
304 for(j = 0; j < 7; j++){ | |
305 if(!bitalloc_codes[i][j]) break; | |
306 dca_smpl_bitalloc[i+1].offset = bitalloc_offsets[i]; | |
307 dca_smpl_bitalloc[i+1].wrap = 1 + (j > 4); | |
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308 dca_smpl_bitalloc[i+1].vlc[j].table = &dca_table[dca_vlc_offs[c]]; |
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309 dca_smpl_bitalloc[i+1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c]; |
4599 | 310 init_vlc(&dca_smpl_bitalloc[i+1].vlc[j], bitalloc_maxbits[i][j], |
311 bitalloc_sizes[i], | |
312 bitalloc_bits[i][j], 1, 1, | |
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313 bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC); |
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314 c++; |
4599 | 315 } |
6350 | 316 vlcs_initialized = 1; |
4599 | 317 } |
318 | |
319 static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) | |
320 { | |
321 while(len--) | |
322 *dst++ = get_bits(gb, bits); | |
323 } | |
324 | |
325 static int dca_parse_frame_header(DCAContext * s) | |
326 { | |
327 int i, j; | |
328 static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; | |
329 static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; | |
330 static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; | |
331 | |
332 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
333 | |
334 /* Sync code */ | |
335 get_bits(&s->gb, 32); | |
336 | |
337 /* Frame header */ | |
338 s->frame_type = get_bits(&s->gb, 1); | |
339 s->samples_deficit = get_bits(&s->gb, 5) + 1; | |
340 s->crc_present = get_bits(&s->gb, 1); | |
341 s->sample_blocks = get_bits(&s->gb, 7) + 1; | |
342 s->frame_size = get_bits(&s->gb, 14) + 1; | |
343 if (s->frame_size < 95) | |
344 return -1; | |
345 s->amode = get_bits(&s->gb, 6); | |
346 s->sample_rate = dca_sample_rates[get_bits(&s->gb, 4)]; | |
347 if (!s->sample_rate) | |
348 return -1; | |
8078 | 349 s->bit_rate_index = get_bits(&s->gb, 5); |
8077 | 350 s->bit_rate = dca_bit_rates[s->bit_rate_index]; |
4599 | 351 if (!s->bit_rate) |
352 return -1; | |
353 | |
354 s->downmix = get_bits(&s->gb, 1); | |
355 s->dynrange = get_bits(&s->gb, 1); | |
356 s->timestamp = get_bits(&s->gb, 1); | |
357 s->aux_data = get_bits(&s->gb, 1); | |
358 s->hdcd = get_bits(&s->gb, 1); | |
359 s->ext_descr = get_bits(&s->gb, 3); | |
360 s->ext_coding = get_bits(&s->gb, 1); | |
361 s->aspf = get_bits(&s->gb, 1); | |
362 s->lfe = get_bits(&s->gb, 2); | |
363 s->predictor_history = get_bits(&s->gb, 1); | |
364 | |
365 /* TODO: check CRC */ | |
366 if (s->crc_present) | |
367 s->header_crc = get_bits(&s->gb, 16); | |
368 | |
369 s->multirate_inter = get_bits(&s->gb, 1); | |
370 s->version = get_bits(&s->gb, 4); | |
371 s->copy_history = get_bits(&s->gb, 2); | |
372 s->source_pcm_res = get_bits(&s->gb, 3); | |
373 s->front_sum = get_bits(&s->gb, 1); | |
374 s->surround_sum = get_bits(&s->gb, 1); | |
375 s->dialog_norm = get_bits(&s->gb, 4); | |
376 | |
377 /* FIXME: channels mixing levels */ | |
4893 | 378 s->output = s->amode; |
379 if(s->lfe) s->output |= DCA_LFE; | |
4599 | 380 |
381 #ifdef TRACE | |
382 av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); | |
383 av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit); | |
384 av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present); | |
385 av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n", | |
386 s->sample_blocks, s->sample_blocks * 32); | |
387 av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size); | |
388 av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n", | |
389 s->amode, dca_channels[s->amode]); | |
8061 | 390 av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n", |
391 s->sample_rate); | |
392 av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n", | |
393 s->bit_rate); | |
4599 | 394 av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix); |
395 av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange); | |
396 av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp); | |
397 av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data); | |
398 av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd); | |
399 av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr); | |
400 av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding); | |
401 av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf); | |
402 av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe); | |
403 av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n", | |
404 s->predictor_history); | |
405 av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc); | |
406 av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n", | |
407 s->multirate_inter); | |
408 av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version); | |
409 av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history); | |
410 av_log(s->avctx, AV_LOG_DEBUG, | |
411 "source pcm resolution: %i (%i bits/sample)\n", | |
412 s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]); | |
413 av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum); | |
414 av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum); | |
415 av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm); | |
416 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
417 #endif | |
418 | |
419 /* Primary audio coding header */ | |
420 s->subframes = get_bits(&s->gb, 4) + 1; | |
6463 | 421 s->total_channels = get_bits(&s->gb, 3) + 1; |
422 s->prim_channels = s->total_channels; | |
423 if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) | |
424 s->prim_channels = DCA_PRIM_CHANNELS_MAX; /* We only support DTS core */ | |
4599 | 425 |
426 | |
427 for (i = 0; i < s->prim_channels; i++) { | |
428 s->subband_activity[i] = get_bits(&s->gb, 5) + 2; | |
429 if (s->subband_activity[i] > DCA_SUBBANDS) | |
430 s->subband_activity[i] = DCA_SUBBANDS; | |
431 } | |
432 for (i = 0; i < s->prim_channels; i++) { | |
433 s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; | |
434 if (s->vq_start_subband[i] > DCA_SUBBANDS) | |
435 s->vq_start_subband[i] = DCA_SUBBANDS; | |
436 } | |
437 get_array(&s->gb, s->joint_intensity, s->prim_channels, 3); | |
438 get_array(&s->gb, s->transient_huffman, s->prim_channels, 2); | |
439 get_array(&s->gb, s->scalefactor_huffman, s->prim_channels, 3); | |
440 get_array(&s->gb, s->bitalloc_huffman, s->prim_channels, 3); | |
441 | |
442 /* Get codebooks quantization indexes */ | |
443 memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); | |
444 for (j = 1; j < 11; j++) | |
445 for (i = 0; i < s->prim_channels; i++) | |
446 s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); | |
447 | |
448 /* Get scale factor adjustment */ | |
449 for (j = 0; j < 11; j++) | |
450 for (i = 0; i < s->prim_channels; i++) | |
451 s->scalefactor_adj[i][j] = 1; | |
452 | |
453 for (j = 1; j < 11; j++) | |
454 for (i = 0; i < s->prim_channels; i++) | |
455 if (s->quant_index_huffman[i][j] < thr[j]) | |
456 s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; | |
457 | |
458 if (s->crc_present) { | |
459 /* Audio header CRC check */ | |
460 get_bits(&s->gb, 16); | |
461 } | |
462 | |
463 s->current_subframe = 0; | |
464 s->current_subsubframe = 0; | |
465 | |
466 #ifdef TRACE | |
467 av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes); | |
468 av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels); | |
469 for(i = 0; i < s->prim_channels; i++){ | |
470 av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", s->subband_activity[i]); | |
471 av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", s->vq_start_subband[i]); | |
472 av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", s->joint_intensity[i]); | |
473 av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", s->transient_huffman[i]); | |
474 av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", s->scalefactor_huffman[i]); | |
475 av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", s->bitalloc_huffman[i]); | |
476 av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:"); | |
477 for (j = 0; j < 11; j++) | |
478 av_log(s->avctx, AV_LOG_DEBUG, " %i", | |
479 s->quant_index_huffman[i][j]); | |
480 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
481 av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:"); | |
482 for (j = 0; j < 11; j++) | |
483 av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]); | |
484 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
485 } | |
486 #endif | |
487 | |
488 return 0; | |
489 } | |
490 | |
491 | |
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492 static inline int get_scale(GetBitContext *gb, int level, int value) |
4599 | 493 { |
494 if (level < 5) { | |
495 /* huffman encoded */ | |
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496 value += get_bitalloc(gb, &dca_scalefactor, level); |
4599 | 497 } else if(level < 8) |
498 value = get_bits(gb, level + 1); | |
499 return value; | |
500 } | |
501 | |
502 static int dca_subframe_header(DCAContext * s) | |
503 { | |
504 /* Primary audio coding side information */ | |
505 int j, k; | |
506 | |
507 s->subsubframes = get_bits(&s->gb, 2) + 1; | |
508 s->partial_samples = get_bits(&s->gb, 3); | |
509 for (j = 0; j < s->prim_channels; j++) { | |
510 for (k = 0; k < s->subband_activity[j]; k++) | |
511 s->prediction_mode[j][k] = get_bits(&s->gb, 1); | |
512 } | |
513 | |
514 /* Get prediction codebook */ | |
515 for (j = 0; j < s->prim_channels; j++) { | |
516 for (k = 0; k < s->subband_activity[j]; k++) { | |
517 if (s->prediction_mode[j][k] > 0) { | |
518 /* (Prediction coefficient VQ address) */ | |
519 s->prediction_vq[j][k] = get_bits(&s->gb, 12); | |
520 } | |
521 } | |
522 } | |
523 | |
524 /* Bit allocation index */ | |
525 for (j = 0; j < s->prim_channels; j++) { | |
526 for (k = 0; k < s->vq_start_subband[j]; k++) { | |
527 if (s->bitalloc_huffman[j] == 6) | |
528 s->bitalloc[j][k] = get_bits(&s->gb, 5); | |
529 else if (s->bitalloc_huffman[j] == 5) | |
530 s->bitalloc[j][k] = get_bits(&s->gb, 4); | |
6463 | 531 else if (s->bitalloc_huffman[j] == 7) { |
532 av_log(s->avctx, AV_LOG_ERROR, | |
533 "Invalid bit allocation index\n"); | |
534 return -1; | |
535 } else { | |
4599 | 536 s->bitalloc[j][k] = |
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537 get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); |
4599 | 538 } |
539 | |
540 if (s->bitalloc[j][k] > 26) { | |
541 // av_log(s->avctx,AV_LOG_DEBUG,"bitalloc index [%i][%i] too big (%i)\n", | |
542 // j, k, s->bitalloc[j][k]); | |
543 return -1; | |
544 } | |
545 } | |
546 } | |
547 | |
548 /* Transition mode */ | |
549 for (j = 0; j < s->prim_channels; j++) { | |
550 for (k = 0; k < s->subband_activity[j]; k++) { | |
551 s->transition_mode[j][k] = 0; | |
552 if (s->subsubframes > 1 && | |
553 k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { | |
554 s->transition_mode[j][k] = | |
555 get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); | |
556 } | |
557 } | |
558 } | |
559 | |
560 for (j = 0; j < s->prim_channels; j++) { | |
6214 | 561 const uint32_t *scale_table; |
4599 | 562 int scale_sum; |
563 | |
564 memset(s->scale_factor[j], 0, s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); | |
565 | |
566 if (s->scalefactor_huffman[j] == 6) | |
6214 | 567 scale_table = scale_factor_quant7; |
4599 | 568 else |
6214 | 569 scale_table = scale_factor_quant6; |
4599 | 570 |
571 /* When huffman coded, only the difference is encoded */ | |
572 scale_sum = 0; | |
573 | |
574 for (k = 0; k < s->subband_activity[j]; k++) { | |
575 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { | |
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576 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 577 s->scale_factor[j][k][0] = scale_table[scale_sum]; |
578 } | |
579 | |
580 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { | |
581 /* Get second scale factor */ | |
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582 scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum); |
4599 | 583 s->scale_factor[j][k][1] = scale_table[scale_sum]; |
584 } | |
585 } | |
586 } | |
587 | |
588 /* Joint subband scale factor codebook select */ | |
589 for (j = 0; j < s->prim_channels; j++) { | |
590 /* Transmitted only if joint subband coding enabled */ | |
591 if (s->joint_intensity[j] > 0) | |
592 s->joint_huff[j] = get_bits(&s->gb, 3); | |
593 } | |
594 | |
595 /* Scale factors for joint subband coding */ | |
596 for (j = 0; j < s->prim_channels; j++) { | |
597 int source_channel; | |
598 | |
599 /* Transmitted only if joint subband coding enabled */ | |
600 if (s->joint_intensity[j] > 0) { | |
601 int scale = 0; | |
602 source_channel = s->joint_intensity[j] - 1; | |
603 | |
604 /* When huffman coded, only the difference is encoded | |
605 * (is this valid as well for joint scales ???) */ | |
606 | |
607 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { | |
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608 scale = get_scale(&s->gb, s->joint_huff[j], 0); |
4599 | 609 scale += 64; /* bias */ |
610 s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ | |
611 } | |
612 | |
10380 | 613 if (!(s->debug_flag & 0x02)) { |
4599 | 614 av_log(s->avctx, AV_LOG_DEBUG, |
615 "Joint stereo coding not supported\n"); | |
616 s->debug_flag |= 0x02; | |
617 } | |
618 } | |
619 } | |
620 | |
621 /* Stereo downmix coefficients */ | |
4894 | 622 if (s->prim_channels > 2) { |
623 if(s->downmix) { | |
4895 | 624 for (j = 0; j < s->prim_channels; j++) { |
625 s->downmix_coef[j][0] = get_bits(&s->gb, 7); | |
626 s->downmix_coef[j][1] = get_bits(&s->gb, 7); | |
627 } | |
4894 | 628 } else { |
629 int am = s->amode & DCA_CHANNEL_MASK; | |
630 for (j = 0; j < s->prim_channels; j++) { | |
631 s->downmix_coef[j][0] = dca_default_coeffs[am][j][0]; | |
632 s->downmix_coef[j][1] = dca_default_coeffs[am][j][1]; | |
633 } | |
634 } | |
4599 | 635 } |
636 | |
637 /* Dynamic range coefficient */ | |
638 if (s->dynrange) | |
639 s->dynrange_coef = get_bits(&s->gb, 8); | |
640 | |
641 /* Side information CRC check word */ | |
642 if (s->crc_present) { | |
643 get_bits(&s->gb, 16); | |
644 } | |
645 | |
646 /* | |
647 * Primary audio data arrays | |
648 */ | |
649 | |
650 /* VQ encoded high frequency subbands */ | |
651 for (j = 0; j < s->prim_channels; j++) | |
652 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
653 /* 1 vector -> 32 samples */ | |
654 s->high_freq_vq[j][k] = get_bits(&s->gb, 10); | |
655 | |
656 /* Low frequency effect data */ | |
657 if (s->lfe) { | |
658 /* LFE samples */ | |
659 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
660 float lfe_scale; | |
661 | |
662 for (j = lfe_samples; j < lfe_samples * 2; j++) { | |
663 /* Signed 8 bits int */ | |
664 s->lfe_data[j] = get_sbits(&s->gb, 8); | |
665 } | |
666 | |
667 /* Scale factor index */ | |
668 s->lfe_scale_factor = scale_factor_quant7[get_bits(&s->gb, 8)]; | |
669 | |
670 /* Quantization step size * scale factor */ | |
671 lfe_scale = 0.035 * s->lfe_scale_factor; | |
672 | |
673 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
674 s->lfe_data[j] *= lfe_scale; | |
675 } | |
676 | |
677 #ifdef TRACE | |
678 av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", s->subsubframes); | |
679 av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", | |
680 s->partial_samples); | |
681 for (j = 0; j < s->prim_channels; j++) { | |
682 av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:"); | |
683 for (k = 0; k < s->subband_activity[j]; k++) | |
684 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]); | |
685 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
686 } | |
687 for (j = 0; j < s->prim_channels; j++) { | |
688 for (k = 0; k < s->subband_activity[j]; k++) | |
689 av_log(s->avctx, AV_LOG_DEBUG, | |
690 "prediction coefs: %f, %f, %f, %f\n", | |
691 (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192, | |
692 (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192, | |
693 (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192, | |
694 (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192); | |
695 } | |
696 for (j = 0; j < s->prim_channels; j++) { | |
697 av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: "); | |
698 for (k = 0; k < s->vq_start_subband[j]; k++) | |
699 av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]); | |
700 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
701 } | |
702 for (j = 0; j < s->prim_channels; j++) { | |
703 av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:"); | |
704 for (k = 0; k < s->subband_activity[j]; k++) | |
705 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]); | |
706 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
707 } | |
708 for (j = 0; j < s->prim_channels; j++) { | |
709 av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:"); | |
710 for (k = 0; k < s->subband_activity[j]; k++) { | |
711 if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) | |
712 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]); | |
713 if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) | |
714 av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]); | |
715 } | |
716 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
717 } | |
718 for (j = 0; j < s->prim_channels; j++) { | |
719 if (s->joint_intensity[j] > 0) { | |
5069 | 720 int source_channel = s->joint_intensity[j] - 1; |
4599 | 721 av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n"); |
722 for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) | |
723 av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]); | |
724 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
725 } | |
726 } | |
727 if (s->prim_channels > 2 && s->downmix) { | |
728 av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); | |
729 for (j = 0; j < s->prim_channels; j++) { | |
730 av_log(s->avctx, AV_LOG_DEBUG, "Channel 0,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][0]]); | |
731 av_log(s->avctx, AV_LOG_DEBUG, "Channel 1,%d = %f\n", j, dca_downmix_coeffs[s->downmix_coef[j][1]]); | |
732 } | |
733 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
734 } | |
735 for (j = 0; j < s->prim_channels; j++) | |
736 for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) | |
737 av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]); | |
738 if(s->lfe){ | |
5069 | 739 int lfe_samples = 2 * s->lfe * s->subsubframes; |
4599 | 740 av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n"); |
741 for (j = lfe_samples; j < lfe_samples * 2; j++) | |
742 av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]); | |
743 av_log(s->avctx, AV_LOG_DEBUG, "\n"); | |
744 } | |
745 #endif | |
746 | |
747 return 0; | |
748 } | |
749 | |
750 static void qmf_32_subbands(DCAContext * s, int chans, | |
751 float samples_in[32][8], float *samples_out, | |
752 float scale, float bias) | |
753 { | |
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754 const float *prCoeff; |
10468 | 755 int i; |
4599 | 756 |
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757 int subindex; |
4599 | 758 |
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759 scale *= sqrt(1/8.0); |
4599 | 760 |
761 /* Select filter */ | |
762 if (!s->multirate_inter) /* Non-perfect reconstruction */ | |
5974
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763 prCoeff = fir_32bands_nonperfect; |
4599 | 764 else /* Perfect reconstruction */ |
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765 prCoeff = fir_32bands_perfect; |
4599 | 766 |
767 /* Reconstructed channel sample index */ | |
768 for (subindex = 0; subindex < 8; subindex++) { | |
769 /* Load in one sample from each subband and clear inactive subbands */ | |
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770 for (i = 0; i < s->subband_activity[chans]; i++){ |
10152 | 771 if((i-1)&2) s->raXin[i] = -samples_in[i][subindex]; |
772 else s->raXin[i] = samples_in[i][subindex]; | |
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773 } |
4599 | 774 for (; i < 32; i++) |
10152 | 775 s->raXin[i] = 0.0; |
4599 | 776 |
10467 | 777 ff_synth_filter_float(&s->imdct, |
778 s->subband_fir_hist[chans], &s->hist_index[chans], | |
779 s->subband_fir_noidea[chans], prCoeff, | |
780 samples_out, s->raXin, scale, bias); | |
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781 samples_out+= 32; |
4599 | 782 |
783 } | |
784 } | |
785 | |
786 static void lfe_interpolation_fir(int decimation_select, | |
787 int num_deci_sample, float *samples_in, | |
788 float *samples_out, float scale, | |
789 float bias) | |
790 { | |
791 /* samples_in: An array holding decimated samples. | |
792 * Samples in current subframe starts from samples_in[0], | |
793 * while samples_in[-1], samples_in[-2], ..., stores samples | |
794 * from last subframe as history. | |
795 * | |
796 * samples_out: An array holding interpolated samples | |
797 */ | |
798 | |
799 int decifactor, k, j; | |
800 const float *prCoeff; | |
801 | |
802 int interp_index = 0; /* Index to the interpolated samples */ | |
803 int deciindex; | |
804 | |
805 /* Select decimation filter */ | |
806 if (decimation_select == 1) { | |
807 decifactor = 128; | |
808 prCoeff = lfe_fir_128; | |
809 } else { | |
810 decifactor = 64; | |
811 prCoeff = lfe_fir_64; | |
812 } | |
813 /* Interpolation */ | |
814 for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { | |
815 /* One decimated sample generates decifactor interpolated ones */ | |
816 for (k = 0; k < decifactor; k++) { | |
817 float rTmp = 0.0; | |
818 //FIXME the coeffs are symetric, fix that | |
819 for (j = 0; j < 512 / decifactor; j++) | |
820 rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor]; | |
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821 samples_out[interp_index++] = (rTmp * scale) + bias; |
4599 | 822 } |
823 } | |
824 } | |
825 | |
826 /* downmixing routines */ | |
4894 | 827 #define MIX_REAR1(samples, si1, rs, coef) \ |
828 samples[i] += samples[si1] * coef[rs][0]; \ | |
829 samples[i+256] += samples[si1] * coef[rs][1]; | |
4599 | 830 |
4894 | 831 #define MIX_REAR2(samples, si1, si2, rs, coef) \ |
832 samples[i] += samples[si1] * coef[rs][0] + samples[si2] * coef[rs+1][0]; \ | |
833 samples[i+256] += samples[si1] * coef[rs][1] + samples[si2] * coef[rs+1][1]; | |
4599 | 834 |
4894 | 835 #define MIX_FRONT3(samples, coef) \ |
4599 | 836 t = samples[i]; \ |
4894 | 837 samples[i] = t * coef[0][0] + samples[i+256] * coef[1][0] + samples[i+512] * coef[2][0]; \ |
838 samples[i+256] = t * coef[0][1] + samples[i+256] * coef[1][1] + samples[i+512] * coef[2][1]; | |
4599 | 839 |
840 #define DOWNMIX_TO_STEREO(op1, op2) \ | |
841 for(i = 0; i < 256; i++){ \ | |
842 op1 \ | |
843 op2 \ | |
844 } | |
845 | |
4894 | 846 static void dca_downmix(float *samples, int srcfmt, |
847 int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]) | |
4599 | 848 { |
849 int i; | |
850 float t; | |
4894 | 851 float coef[DCA_PRIM_CHANNELS_MAX][2]; |
852 | |
853 for(i=0; i<DCA_PRIM_CHANNELS_MAX; i++) { | |
854 coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]]; | |
855 coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]]; | |
856 } | |
4599 | 857 |
858 switch (srcfmt) { | |
859 case DCA_MONO: | |
860 case DCA_CHANNEL: | |
861 case DCA_STEREO_TOTAL: | |
862 case DCA_STEREO_SUMDIFF: | |
863 case DCA_4F2R: | |
864 av_log(NULL, 0, "Not implemented!\n"); | |
865 break; | |
866 case DCA_STEREO: | |
867 break; | |
868 case DCA_3F: | |
4894 | 869 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef),); |
4599 | 870 break; |
871 case DCA_2F1R: | |
4894 | 872 DOWNMIX_TO_STEREO(MIX_REAR1(samples, i + 512, 2, coef),); |
4599 | 873 break; |
874 case DCA_3F1R: | |
4894 | 875 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
876 MIX_REAR1(samples, i + 768, 3, coef)); | |
4599 | 877 break; |
878 case DCA_2F2R: | |
4894 | 879 DOWNMIX_TO_STEREO(MIX_REAR2(samples, i + 512, i + 768, 2, coef),); |
4599 | 880 break; |
881 case DCA_3F2R: | |
4894 | 882 DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), |
883 MIX_REAR2(samples, i + 768, i + 1024, 3, coef)); | |
4599 | 884 break; |
885 } | |
886 } | |
887 | |
888 | |
889 /* Very compact version of the block code decoder that does not use table | |
890 * look-up but is slightly slower */ | |
891 static int decode_blockcode(int code, int levels, int *values) | |
892 { | |
893 int i; | |
894 int offset = (levels - 1) >> 1; | |
895 | |
896 for (i = 0; i < 4; i++) { | |
897 values[i] = (code % levels) - offset; | |
898 code /= levels; | |
899 } | |
900 | |
901 if (code == 0) | |
902 return 0; | |
903 else { | |
904 av_log(NULL, AV_LOG_ERROR, "ERROR: block code look-up failed\n"); | |
905 return -1; | |
906 } | |
907 } | |
908 | |
909 static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; | |
910 static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; | |
911 | |
912 static int dca_subsubframe(DCAContext * s) | |
913 { | |
914 int k, l; | |
915 int subsubframe = s->current_subsubframe; | |
916 | |
6214 | 917 const float *quant_step_table; |
4599 | 918 |
919 /* FIXME */ | |
920 float subband_samples[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; | |
921 | |
922 /* | |
923 * Audio data | |
924 */ | |
925 | |
926 /* Select quantization step size table */ | |
8077 | 927 if (s->bit_rate_index == 0x1f) |
6214 | 928 quant_step_table = lossless_quant_d; |
4599 | 929 else |
6214 | 930 quant_step_table = lossy_quant_d; |
4599 | 931 |
932 for (k = 0; k < s->prim_channels; k++) { | |
933 for (l = 0; l < s->vq_start_subband[k]; l++) { | |
934 int m; | |
935 | |
936 /* Select the mid-tread linear quantizer */ | |
937 int abits = s->bitalloc[k][l]; | |
938 | |
939 float quant_step_size = quant_step_table[abits]; | |
940 float rscale; | |
941 | |
942 /* | |
943 * Determine quantization index code book and its type | |
944 */ | |
945 | |
946 /* Select quantization index code book */ | |
947 int sel = s->quant_index_huffman[k][abits]; | |
948 | |
949 /* | |
950 * Extract bits from the bit stream | |
951 */ | |
952 if(!abits){ | |
953 memset(subband_samples[k][l], 0, 8 * sizeof(subband_samples[0][0][0])); | |
954 }else if(abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table){ | |
955 if(abits <= 7){ | |
956 /* Block code */ | |
957 int block_code1, block_code2, size, levels; | |
958 int block[8]; | |
959 | |
960 size = abits_sizes[abits-1]; | |
961 levels = abits_levels[abits-1]; | |
962 | |
963 block_code1 = get_bits(&s->gb, size); | |
964 /* FIXME Should test return value */ | |
965 decode_blockcode(block_code1, levels, block); | |
966 block_code2 = get_bits(&s->gb, size); | |
967 decode_blockcode(block_code2, levels, &block[4]); | |
968 for (m = 0; m < 8; m++) | |
969 subband_samples[k][l][m] = block[m]; | |
970 }else{ | |
971 /* no coding */ | |
972 for (m = 0; m < 8; m++) | |
973 subband_samples[k][l][m] = get_sbits(&s->gb, abits - 3); | |
974 } | |
975 }else{ | |
976 /* Huffman coded */ | |
977 for (m = 0; m < 8; m++) | |
978 subband_samples[k][l][m] = get_bitalloc(&s->gb, &dca_smpl_bitalloc[abits], sel); | |
979 } | |
980 | |
981 /* Deal with transients */ | |
982 if (s->transition_mode[k][l] && | |
983 subsubframe >= s->transition_mode[k][l]) | |
984 rscale = quant_step_size * s->scale_factor[k][l][1]; | |
985 else | |
986 rscale = quant_step_size * s->scale_factor[k][l][0]; | |
987 | |
988 rscale *= s->scalefactor_adj[k][sel]; | |
989 | |
990 for (m = 0; m < 8; m++) | |
991 subband_samples[k][l][m] *= rscale; | |
992 | |
993 /* | |
994 * Inverse ADPCM if in prediction mode | |
995 */ | |
996 if (s->prediction_mode[k][l]) { | |
997 int n; | |
998 for (m = 0; m < 8; m++) { | |
999 for (n = 1; n <= 4; n++) | |
1000 if (m >= n) | |
1001 subband_samples[k][l][m] += | |
1002 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
1003 subband_samples[k][l][m - n] / 8192); | |
1004 else if (s->predictor_history) | |
1005 subband_samples[k][l][m] += | |
1006 (adpcm_vb[s->prediction_vq[k][l]][n - 1] * | |
1007 s->subband_samples_hist[k][l][m - n + | |
1008 4] / 8192); | |
1009 } | |
1010 } | |
1011 } | |
1012 | |
1013 /* | |
1014 * Decode VQ encoded high frequencies | |
1015 */ | |
1016 for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { | |
1017 /* 1 vector -> 32 samples but we only need the 8 samples | |
1018 * for this subsubframe. */ | |
1019 int m; | |
1020 | |
1021 if (!s->debug_flag & 0x01) { | |
1022 av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n"); | |
1023 s->debug_flag |= 0x01; | |
1024 } | |
1025 | |
1026 for (m = 0; m < 8; m++) { | |
1027 subband_samples[k][l][m] = | |
1028 high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 + | |
1029 m] | |
1030 * (float) s->scale_factor[k][l][0] / 16.0; | |
1031 } | |
1032 } | |
1033 } | |
1034 | |
1035 /* Check for DSYNC after subsubframe */ | |
1036 if (s->aspf || subsubframe == s->subsubframes - 1) { | |
1037 if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */ | |
1038 #ifdef TRACE | |
1039 av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n"); | |
1040 #endif | |
1041 } else { | |
1042 av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); | |
1043 } | |
1044 } | |
1045 | |
1046 /* Backup predictor history for adpcm */ | |
1047 for (k = 0; k < s->prim_channels; k++) | |
1048 for (l = 0; l < s->vq_start_subband[k]; l++) | |
1049 memcpy(s->subband_samples_hist[k][l], &subband_samples[k][l][4], | |
1050 4 * sizeof(subband_samples[0][0][0])); | |
1051 | |
1052 /* 32 subbands QMF */ | |
1053 for (k = 0; k < s->prim_channels; k++) { | |
1054 /* static float pcm_to_double[8] = | |
1055 {32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/ | |
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1056 qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * s->channel_order_tab[k]], |
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1057 M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ , |
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1058 s->add_bias ); |
4599 | 1059 } |
1060 | |
1061 /* Down mixing */ | |
1062 | |
1063 if (s->prim_channels > dca_channels[s->output & DCA_CHANNEL_MASK]) { | |
4894 | 1064 dca_downmix(s->samples, s->amode, s->downmix_coef); |
4599 | 1065 } |
1066 | |
1067 /* Generate LFE samples for this subsubframe FIXME!!! */ | |
1068 if (s->output & DCA_LFE) { | |
1069 int lfe_samples = 2 * s->lfe * s->subsubframes; | |
1070 | |
1071 lfe_interpolation_fir(s->lfe, 2 * s->lfe, | |
1072 s->lfe_data + lfe_samples + | |
1073 2 * s->lfe * subsubframe, | |
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1074 &s->samples[256 * dca_lfe_index[s->amode]], |
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1075 (1.0/256.0)*s->scale_bias, s->add_bias); |
4599 | 1076 /* Outputs 20bits pcm samples */ |
1077 } | |
1078 | |
1079 return 0; | |
1080 } | |
1081 | |
1082 | |
1083 static int dca_subframe_footer(DCAContext * s) | |
1084 { | |
1085 int aux_data_count = 0, i; | |
1086 int lfe_samples; | |
1087 | |
1088 /* | |
1089 * Unpack optional information | |
1090 */ | |
1091 | |
1092 if (s->timestamp) | |
1093 get_bits(&s->gb, 32); | |
1094 | |
1095 if (s->aux_data) | |
1096 aux_data_count = get_bits(&s->gb, 6); | |
1097 | |
1098 for (i = 0; i < aux_data_count; i++) | |
1099 get_bits(&s->gb, 8); | |
1100 | |
1101 if (s->crc_present && (s->downmix || s->dynrange)) | |
1102 get_bits(&s->gb, 16); | |
1103 | |
1104 lfe_samples = 2 * s->lfe * s->subsubframes; | |
1105 for (i = 0; i < lfe_samples; i++) { | |
1106 s->lfe_data[i] = s->lfe_data[i + lfe_samples]; | |
1107 } | |
1108 | |
1109 return 0; | |
1110 } | |
1111 | |
1112 /** | |
1113 * Decode a dca frame block | |
1114 * | |
1115 * @param s pointer to the DCAContext | |
1116 */ | |
1117 | |
1118 static int dca_decode_block(DCAContext * s) | |
1119 { | |
1120 | |
1121 /* Sanity check */ | |
1122 if (s->current_subframe >= s->subframes) { | |
1123 av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", | |
1124 s->current_subframe, s->subframes); | |
1125 return -1; | |
1126 } | |
1127 | |
1128 if (!s->current_subsubframe) { | |
1129 #ifdef TRACE | |
1130 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); | |
1131 #endif | |
1132 /* Read subframe header */ | |
1133 if (dca_subframe_header(s)) | |
1134 return -1; | |
1135 } | |
1136 | |
1137 /* Read subsubframe */ | |
1138 #ifdef TRACE | |
1139 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); | |
1140 #endif | |
1141 if (dca_subsubframe(s)) | |
1142 return -1; | |
1143 | |
1144 /* Update state */ | |
1145 s->current_subsubframe++; | |
1146 if (s->current_subsubframe >= s->subsubframes) { | |
1147 s->current_subsubframe = 0; | |
1148 s->current_subframe++; | |
1149 } | |
1150 if (s->current_subframe >= s->subframes) { | |
1151 #ifdef TRACE | |
1152 av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); | |
1153 #endif | |
1154 /* Read subframe footer */ | |
1155 if (dca_subframe_footer(s)) | |
1156 return -1; | |
1157 } | |
1158 | |
1159 return 0; | |
1160 } | |
1161 | |
1162 /** | |
1163 * Convert bitstream to one representation based on sync marker | |
1164 */ | |
6214 | 1165 static int dca_convert_bitstream(const uint8_t * src, int src_size, uint8_t * dst, |
4599 | 1166 int max_size) |
1167 { | |
1168 uint32_t mrk; | |
1169 int i, tmp; | |
6214 | 1170 const uint16_t *ssrc = (const uint16_t *) src; |
1171 uint16_t *sdst = (uint16_t *) dst; | |
4599 | 1172 PutBitContext pb; |
1173 | |
5027 | 1174 if((unsigned)src_size > (unsigned)max_size) { |
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1175 // av_log(NULL, AV_LOG_ERROR, "Input frame size larger then DCA_MAX_FRAME_SIZE!\n"); |
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1176 // return -1; |
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1177 src_size = max_size; |
5027 | 1178 } |
4883 | 1179 |
4599 | 1180 mrk = AV_RB32(src); |
1181 switch (mrk) { | |
1182 case DCA_MARKER_RAW_BE: | |
7671 | 1183 memcpy(dst, src, src_size); |
1184 return src_size; | |
4599 | 1185 case DCA_MARKER_RAW_LE: |
7671 | 1186 for (i = 0; i < (src_size + 1) >> 1; i++) |
4599 | 1187 *sdst++ = bswap_16(*ssrc++); |
7671 | 1188 return src_size; |
4599 | 1189 case DCA_MARKER_14B_BE: |
1190 case DCA_MARKER_14B_LE: | |
1191 init_put_bits(&pb, dst, max_size); | |
1192 for (i = 0; i < (src_size + 1) >> 1; i++, src += 2) { | |
1193 tmp = ((mrk == DCA_MARKER_14B_BE) ? AV_RB16(src) : AV_RL16(src)) & 0x3FFF; | |
1194 put_bits(&pb, 14, tmp); | |
1195 } | |
1196 flush_put_bits(&pb); | |
1197 return (put_bits_count(&pb) + 7) >> 3; | |
1198 default: | |
1199 return -1; | |
1200 } | |
1201 } | |
1202 | |
1203 /** | |
1204 * Main frame decoding function | |
1205 * FIXME add arguments | |
1206 */ | |
1207 static int dca_decode_frame(AVCodecContext * avctx, | |
1208 void *data, int *data_size, | |
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1209 AVPacket *avpkt) |
4599 | 1210 { |
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1211 const uint8_t *buf = avpkt->data; |
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1212 int buf_size = avpkt->size; |
4599 | 1213 |
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1214 int i; |
4599 | 1215 int16_t *samples = data; |
1216 DCAContext *s = avctx->priv_data; | |
1217 int channels; | |
1218 | |
1219 | |
1220 s->dca_buffer_size = dca_convert_bitstream(buf, buf_size, s->dca_buffer, DCA_MAX_FRAME_SIZE); | |
1221 if (s->dca_buffer_size == -1) { | |
5027 | 1222 av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); |
4599 | 1223 return -1; |
1224 } | |
1225 | |
1226 init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); | |
1227 if (dca_parse_frame_header(s) < 0) { | |
1228 //seems like the frame is corrupt, try with the next one | |
5645 | 1229 *data_size=0; |
4599 | 1230 return buf_size; |
1231 } | |
1232 //set AVCodec values with parsed data | |
1233 avctx->sample_rate = s->sample_rate; | |
1234 avctx->bit_rate = s->bit_rate; | |
1235 | |
4893 | 1236 channels = s->prim_channels + !!s->lfe; |
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1237 |
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1238 if (s->amode<16) { |
8100 | 1239 avctx->channel_layout = dca_core_channel_layout[s->amode]; |
1240 | |
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1241 if (s->lfe) { |
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1242 avctx->channel_layout |= CH_LOW_FREQUENCY; |
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1243 s->channel_order_tab = dca_channel_reorder_lfe[s->amode]; |
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1244 } else |
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1245 s->channel_order_tab = dca_channel_reorder_nolfe[s->amode]; |
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1246 |
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1247 if(avctx->request_channels == 2 && s->prim_channels > 2) { |
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1248 channels = 2; |
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1249 s->output = DCA_STEREO; |
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1250 avctx->channel_layout = CH_LAYOUT_STEREO; |
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1251 } |
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1252 } else { |
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1253 av_log(avctx, AV_LOG_ERROR, "Non standard configuration %d !\n",s->amode); |
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1254 return -1; |
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1255 } |
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1256 |
4893 | 1257 |
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1258 /* There is nothing that prevents a dts frame to change channel configuration |
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1259 but FFmpeg doesn't support that so only set the channels if it is previously |
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1260 unset. Ideally during the first probe for channels the crc should be checked |
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1261 and only set avctx->channels when the crc is ok. Right now the decoder could |
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1262 set the channels based on a broken first frame.*/ |
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1263 if (!avctx->channels) |
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1264 avctx->channels = channels; |
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1265 |
4599 | 1266 if(*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels) |
1267 return -1; | |
7725 | 1268 *data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels; |
4599 | 1269 for (i = 0; i < (s->sample_blocks / 8); i++) { |
1270 dca_decode_block(s); | |
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1271 s->dsp.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels); |
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1272 samples += 256 * channels; |
4599 | 1273 } |
1274 | |
1275 return buf_size; | |
1276 } | |
1277 | |
1278 | |
1279 | |
1280 /** | |
1281 * DCA initialization | |
1282 * | |
1283 * @param avctx pointer to the AVCodecContext | |
1284 */ | |
1285 | |
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1286 static av_cold int dca_decode_init(AVCodecContext * avctx) |
4599 | 1287 { |
1288 DCAContext *s = avctx->priv_data; | |
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1289 int i; |
4599 | 1290 |
1291 s->avctx = avctx; | |
1292 dca_init_vlcs(); | |
1293 | |
1294 dsputil_init(&s->dsp, avctx); | |
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1295 ff_mdct_init(&s->imdct, 6, 1, 1.0); |
6120 | 1296 |
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1297 for(i = 0; i < 6; i++) |
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1298 s->samples_chanptr[i] = s->samples + i * 256; |
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1299 avctx->sample_fmt = SAMPLE_FMT_S16; |
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1300 |
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1301 if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { |
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1302 s->add_bias = 385.0f; |
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1303 s->scale_bias = 1.0 / 32768.0; |
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1304 } else { |
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1305 s->add_bias = 0.0f; |
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1306 s->scale_bias = 1.0; |
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1307 |
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1308 /* allow downmixing to stereo */ |
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1309 if (avctx->channels > 0 && avctx->request_channels < avctx->channels && |
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1310 avctx->request_channels == 2) { |
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1311 avctx->channels = avctx->request_channels; |
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1312 } |
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1313 } |
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1314 |
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1315 |
4599 | 1316 return 0; |
1317 } | |
1318 | |
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1319 static av_cold int dca_decode_end(AVCodecContext * avctx) |
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1320 { |
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1321 DCAContext *s = avctx->priv_data; |
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1322 ff_mdct_end(&s->imdct); |
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1323 return 0; |
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1324 } |
4599 | 1325 |
1326 AVCodec dca_decoder = { | |
1327 .name = "dca", | |
1328 .type = CODEC_TYPE_AUDIO, | |
1329 .id = CODEC_ID_DTS, | |
1330 .priv_data_size = sizeof(DCAContext), | |
1331 .init = dca_decode_init, | |
1332 .decode = dca_decode_frame, | |
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1333 .close = dca_decode_end, |
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1334 .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), |
4599 | 1335 }; |