GCC Code Coverage Report | |||||||||||||||||||||
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Line | Branch | Exec | Source |
1 |
/* |
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2 |
* QDM2 compatible decoder |
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3 |
* Copyright (c) 2003 Ewald Snel |
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4 |
* Copyright (c) 2005 Benjamin Larsson |
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5 |
* Copyright (c) 2005 Alex Beregszaszi |
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6 |
* Copyright (c) 2005 Roberto Togni |
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7 |
* |
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8 |
* This file is part of FFmpeg. |
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9 |
* |
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10 |
* FFmpeg is free software; you can redistribute it and/or |
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11 |
* modify it under the terms of the GNU Lesser General Public |
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12 |
* License as published by the Free Software Foundation; either |
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13 |
* version 2.1 of the License, or (at your option) any later version. |
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14 |
* |
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15 |
* FFmpeg is distributed in the hope that it will be useful, |
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16 |
* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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17 |
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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18 |
* Lesser General Public License for more details. |
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19 |
* |
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20 |
* You should have received a copy of the GNU Lesser General Public |
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21 |
* License along with FFmpeg; if not, write to the Free Software |
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22 |
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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23 |
*/ |
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24 |
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25 |
/** |
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26 |
* @file |
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27 |
* QDM2 decoder |
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28 |
* @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni |
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29 |
* |
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30 |
* The decoder is not perfect yet, there are still some distortions |
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31 |
* especially on files encoded with 16 or 8 subbands. |
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32 |
*/ |
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33 |
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34 |
#include <math.h> |
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35 |
#include <stddef.h> |
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36 |
#include <stdio.h> |
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37 |
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38 |
#include "libavutil/channel_layout.h" |
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39 |
#include "libavutil/mem_internal.h" |
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40 |
#include "libavutil/thread.h" |
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41 |
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42 |
#define BITSTREAM_READER_LE |
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43 |
#include "avcodec.h" |
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44 |
#include "get_bits.h" |
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45 |
#include "bytestream.h" |
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46 |
#include "internal.h" |
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47 |
#include "mpegaudio.h" |
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48 |
#include "mpegaudiodsp.h" |
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49 |
#include "rdft.h" |
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50 |
|||
51 |
#include "qdm2_tablegen.h" |
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52 |
|||
53 |
#define QDM2_LIST_ADD(list, size, packet) \ |
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54 |
do { \ |
||
55 |
if (size > 0) { \ |
||
56 |
list[size - 1].next = &list[size]; \ |
||
57 |
} \ |
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58 |
list[size].packet = packet; \ |
||
59 |
list[size].next = NULL; \ |
||
60 |
size++; \ |
||
61 |
} while(0) |
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62 |
|||
63 |
// Result is 8, 16 or 30 |
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64 |
#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling)) |
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65 |
|||
66 |
#define FIX_NOISE_IDX(noise_idx) \ |
||
67 |
if ((noise_idx) >= 3840) \ |
||
68 |
(noise_idx) -= 3840; \ |
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69 |
|||
70 |
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)]) |
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71 |
|||
72 |
#define SAMPLES_NEEDED \ |
||
73 |
av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n"); |
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74 |
|||
75 |
#define SAMPLES_NEEDED_2(why) \ |
||
76 |
av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why); |
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77 |
|||
78 |
#define QDM2_MAX_FRAME_SIZE 512 |
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79 |
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80 |
typedef int8_t sb_int8_array[2][30][64]; |
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81 |
|||
82 |
/** |
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83 |
* Subpacket |
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84 |
*/ |
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85 |
typedef struct QDM2SubPacket { |
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86 |
int type; ///< subpacket type |
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87 |
unsigned int size; ///< subpacket size |
||
88 |
const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy) |
||
89 |
} QDM2SubPacket; |
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90 |
|||
91 |
/** |
||
92 |
* A node in the subpacket list |
||
93 |
*/ |
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94 |
typedef struct QDM2SubPNode { |
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95 |
QDM2SubPacket *packet; ///< packet |
||
96 |
struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node |
||
97 |
} QDM2SubPNode; |
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98 |
|||
99 |
typedef struct QDM2Complex { |
||
100 |
float re; |
||
101 |
float im; |
||
102 |
} QDM2Complex; |
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103 |
|||
104 |
typedef struct FFTTone { |
||
105 |
float level; |
||
106 |
QDM2Complex *complex; |
||
107 |
const float *table; |
||
108 |
int phase; |
||
109 |
int phase_shift; |
||
110 |
int duration; |
||
111 |
short time_index; |
||
112 |
short cutoff; |
||
113 |
} FFTTone; |
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114 |
|||
115 |
typedef struct FFTCoefficient { |
||
116 |
int16_t sub_packet; |
||
117 |
uint8_t channel; |
||
118 |
int16_t offset; |
||
119 |
int16_t exp; |
||
120 |
uint8_t phase; |
||
121 |
} FFTCoefficient; |
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122 |
|||
123 |
typedef struct QDM2FFT { |
||
124 |
DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256]; |
||
125 |
} QDM2FFT; |
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126 |
|||
127 |
/** |
||
128 |
* QDM2 decoder context |
||
129 |
*/ |
||
130 |
typedef struct QDM2Context { |
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131 |
/// Parameters from codec header, do not change during playback |
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132 |
int nb_channels; ///< number of channels |
||
133 |
int channels; ///< number of channels |
||
134 |
int group_size; ///< size of frame group (16 frames per group) |
||
135 |
int fft_size; ///< size of FFT, in complex numbers |
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136 |
int checksum_size; ///< size of data block, used also for checksum |
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137 |
|||
138 |
/// Parameters built from header parameters, do not change during playback |
||
139 |
int group_order; ///< order of frame group |
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140 |
int fft_order; ///< order of FFT (actually fftorder+1) |
||
141 |
int frame_size; ///< size of data frame |
||
142 |
int frequency_range; |
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143 |
int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */ |
||
144 |
int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2 |
||
145 |
int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4) |
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146 |
|||
147 |
/// Packets and packet lists |
||
148 |
QDM2SubPacket sub_packets[16]; ///< the packets themselves |
||
149 |
QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets |
||
150 |
QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list |
||
151 |
int sub_packets_B; ///< number of packets on 'B' list |
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152 |
QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors? |
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153 |
QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets |
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154 |
|||
155 |
/// FFT and tones |
||
156 |
FFTTone fft_tones[1000]; |
||
157 |
int fft_tone_start; |
||
158 |
int fft_tone_end; |
||
159 |
FFTCoefficient fft_coefs[1000]; |
||
160 |
int fft_coefs_index; |
||
161 |
int fft_coefs_min_index[5]; |
||
162 |
int fft_coefs_max_index[5]; |
||
163 |
int fft_level_exp[6]; |
||
164 |
RDFTContext rdft_ctx; |
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165 |
QDM2FFT fft; |
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166 |
|||
167 |
/// I/O data |
||
168 |
const uint8_t *compressed_data; |
||
169 |
int compressed_size; |
||
170 |
float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2]; |
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171 |
|||
172 |
/// Synthesis filter |
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173 |
MPADSPContext mpadsp; |
||
174 |
DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2]; |
||
175 |
int synth_buf_offset[MPA_MAX_CHANNELS]; |
||
176 |
DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; |
||
177 |
DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE]; |
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178 |
|||
179 |
/// Mixed temporary data used in decoding |
||
180 |
float tone_level[MPA_MAX_CHANNELS][30][64]; |
||
181 |
int8_t coding_method[MPA_MAX_CHANNELS][30][64]; |
||
182 |
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]; |
||
183 |
int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]; |
||
184 |
int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]; |
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185 |
int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]; |
||
186 |
int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]; |
||
187 |
int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]; |
||
188 |
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]; |
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189 |
|||
190 |
// Flags |
||
191 |
int has_errors; ///< packet has errors |
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192 |
int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type |
||
193 |
int do_synth_filter; ///< used to perform or skip synthesis filter |
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194 |
|||
195 |
int sub_packet; |
||
196 |
int noise_idx; ///< index for dithering noise table |
||
197 |
} QDM2Context; |
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198 |
|||
199 |
static const int switchtable[23] = { |
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200 |
0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4 |
||
201 |
}; |
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202 |
|||
203 |
67245 |
static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth) |
|
204 |
{ |
||
205 |
int value; |
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206 |
|||
207 |
67245 |
value = get_vlc2(gb, vlc->table, vlc->bits, depth); |
|
208 |
|||
209 |
/* stage-2, 3 bits exponent escape sequence */ |
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210 |
✓✓ | 67245 |
if (value < 0) |
211 |
556 |
value = get_bits(gb, get_bits(gb, 3) + 1); |
|
212 |
|||
213 |
/* stage-3, optional */ |
||
214 |
✓✓ | 67245 |
if (flag) { |
215 |
int tmp; |
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216 |
|||
217 |
✗✓ | 18212 |
if (value >= 60) { |
218 |
av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value); |
||
219 |
return 0; |
||
220 |
} |
||
221 |
|||
222 |
18212 |
tmp= vlc_stage3_values[value]; |
|
223 |
|||
224 |
✓✓ | 18212 |
if ((value & ~3) > 0) |
225 |
14145 |
tmp += get_bits(gb, (value >> 2)); |
|
226 |
18212 |
value = tmp; |
|
227 |
} |
||
228 |
|||
229 |
67245 |
return value; |
|
230 |
} |
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231 |
|||
232 |
10996 |
static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth) |
|
233 |
{ |
||
234 |
10996 |
int value = qdm2_get_vlc(gb, vlc, 0, depth); |
|
235 |
|||
236 |
✓✓ | 10996 |
return (value & 1) ? ((value + 1) >> 1) : -(value >> 1); |
237 |
} |
||
238 |
|||
239 |
/** |
||
240 |
* QDM2 checksum |
||
241 |
* |
||
242 |
* @param data pointer to data to be checksummed |
||
243 |
* @param length data length |
||
244 |
* @param value checksum value |
||
245 |
* |
||
246 |
* @return 0 if checksum is OK |
||
247 |
*/ |
||
248 |
139 |
static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value) |
|
249 |
{ |
||
250 |
int i; |
||
251 |
|||
252 |
✓✓ | 51569 |
for (i = 0; i < length; i++) |
253 |
51430 |
value -= data[i]; |
|
254 |
|||
255 |
139 |
return (uint16_t)(value & 0xffff); |
|
256 |
} |
||
257 |
|||
258 |
/** |
||
259 |
* Fill a QDM2SubPacket structure with packet type, size, and data pointer. |
||
260 |
* |
||
261 |
* @param gb bitreader context |
||
262 |
* @param sub_packet packet under analysis |
||
263 |
*/ |
||
264 |
1112 |
static void qdm2_decode_sub_packet_header(GetBitContext *gb, |
|
265 |
QDM2SubPacket *sub_packet) |
||
266 |
{ |
||
267 |
1112 |
sub_packet->type = get_bits(gb, 8); |
|
268 |
|||
269 |
✓✓ | 1112 |
if (sub_packet->type == 0) { |
270 |
139 |
sub_packet->size = 0; |
|
271 |
139 |
sub_packet->data = NULL; |
|
272 |
} else { |
||
273 |
973 |
sub_packet->size = get_bits(gb, 8); |
|
274 |
|||
275 |
✓✓ | 973 |
if (sub_packet->type & 0x80) { |
276 |
139 |
sub_packet->size <<= 8; |
|
277 |
139 |
sub_packet->size |= get_bits(gb, 8); |
|
278 |
139 |
sub_packet->type &= 0x7f; |
|
279 |
} |
||
280 |
|||
281 |
✗✓ | 973 |
if (sub_packet->type == 0x7f) |
282 |
sub_packet->type |= (get_bits(gb, 8) << 8); |
||
283 |
|||
284 |
// FIXME: this depends on bitreader-internal data |
||
285 |
973 |
sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; |
|
286 |
} |
||
287 |
|||
288 |
1112 |
av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n", |
|
289 |
1112 |
sub_packet->type, sub_packet->size, get_bits_count(gb) / 8); |
|
290 |
1112 |
} |
|
291 |
|||
292 |
/** |
||
293 |
* Return node pointer to first packet of requested type in list. |
||
294 |
* |
||
295 |
* @param list list of subpackets to be scanned |
||
296 |
* @param type type of searched subpacket |
||
297 |
* @return node pointer for subpacket if found, else NULL |
||
298 |
*/ |
||
299 |
556 |
static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, |
|
300 |
int type) |
||
301 |
{ |
||
302 |
✓✓✓✗ |
973 |
while (list && list->packet) { |
303 |
✓✓ | 556 |
if (list->packet->type == type) |
304 |
139 |
return list; |
|
305 |
417 |
list = list->next; |
|
306 |
} |
||
307 |
417 |
return NULL; |
|
308 |
} |
||
309 |
|||
310 |
/** |
||
311 |
* Replace 8 elements with their average value. |
||
312 |
* Called by qdm2_decode_superblock before starting subblock decoding. |
||
313 |
* |
||
314 |
* @param q context |
||
315 |
*/ |
||
316 |
139 |
static void average_quantized_coeffs(QDM2Context *q) |
|
317 |
{ |
||
318 |
int i, j, n, ch, sum; |
||
319 |
|||
320 |
✗✓ | 139 |
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
321 |
|||
322 |
✓✓ | 417 |
for (ch = 0; ch < q->nb_channels; ch++) |
323 |
✓✓ | 3058 |
for (i = 0; i < n; i++) { |
324 |
2780 |
sum = 0; |
|
325 |
|||
326 |
✓✓ | 25020 |
for (j = 0; j < 8; j++) |
327 |
22240 |
sum += q->quantized_coeffs[ch][i][j]; |
|
328 |
|||
329 |
2780 |
sum /= 8; |
|
330 |
✓✓ | 2780 |
if (sum > 0) |
331 |
2466 |
sum--; |
|
332 |
|||
333 |
✓✓ | 25020 |
for (j = 0; j < 8; j++) |
334 |
22240 |
q->quantized_coeffs[ch][i][j] = sum; |
|
335 |
} |
||
336 |
139 |
} |
|
337 |
|||
338 |
/** |
||
339 |
* Build subband samples with noise weighted by q->tone_level. |
||
340 |
* Called by synthfilt_build_sb_samples. |
||
341 |
* |
||
342 |
* @param q context |
||
343 |
* @param sb subband index |
||
344 |
*/ |
||
345 |
4170 |
static void build_sb_samples_from_noise(QDM2Context *q, int sb) |
|
346 |
{ |
||
347 |
int ch, j; |
||
348 |
|||
349 |
✓✓ | 4170 |
FIX_NOISE_IDX(q->noise_idx); |
350 |
|||
351 |
✗✓ | 4170 |
if (!q->nb_channels) |
352 |
return; |
||
353 |
|||
354 |
✓✓ | 12510 |
for (ch = 0; ch < q->nb_channels; ch++) { |
355 |
✓✓ | 542100 |
for (j = 0; j < 64; j++) { |
356 |
533760 |
q->sb_samples[ch][j * 2][sb] = |
|
357 |
533760 |
SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; |
|
358 |
533760 |
q->sb_samples[ch][j * 2 + 1][sb] = |
|
359 |
533760 |
SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j]; |
|
360 |
} |
||
361 |
} |
||
362 |
} |
||
363 |
|||
364 |
/** |
||
365 |
* Called while processing data from subpackets 11 and 12. |
||
366 |
* Used after making changes to coding_method array. |
||
367 |
* |
||
368 |
* @param sb subband index |
||
369 |
* @param channels number of channels |
||
370 |
* @param coding_method q->coding_method[0][0][0] |
||
371 |
*/ |
||
372 |
static int fix_coding_method_array(int sb, int channels, |
||
373 |
sb_int8_array coding_method) |
||
374 |
{ |
||
375 |
int j, k; |
||
376 |
int ch; |
||
377 |
int run, case_val; |
||
378 |
|||
379 |
for (ch = 0; ch < channels; ch++) { |
||
380 |
for (j = 0; j < 64; ) { |
||
381 |
if (coding_method[ch][sb][j] < 8) |
||
382 |
return -1; |
||
383 |
if ((coding_method[ch][sb][j] - 8) > 22) { |
||
384 |
run = 1; |
||
385 |
case_val = 8; |
||
386 |
} else { |
||
387 |
switch (switchtable[coding_method[ch][sb][j] - 8]) { |
||
388 |
case 0: run = 10; |
||
389 |
case_val = 10; |
||
390 |
break; |
||
391 |
case 1: run = 1; |
||
392 |
case_val = 16; |
||
393 |
break; |
||
394 |
case 2: run = 5; |
||
395 |
case_val = 24; |
||
396 |
break; |
||
397 |
case 3: run = 3; |
||
398 |
case_val = 30; |
||
399 |
break; |
||
400 |
case 4: run = 1; |
||
401 |
case_val = 30; |
||
402 |
break; |
||
403 |
case 5: run = 1; |
||
404 |
case_val = 8; |
||
405 |
break; |
||
406 |
default: run = 1; |
||
407 |
case_val = 8; |
||
408 |
break; |
||
409 |
} |
||
410 |
} |
||
411 |
for (k = 0; k < run; k++) { |
||
412 |
if (j + k < 128) { |
||
413 |
int sbjk = sb + (j + k) / 64; |
||
414 |
if (sbjk > 29) { |
||
415 |
SAMPLES_NEEDED |
||
416 |
continue; |
||
417 |
} |
||
418 |
if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) { |
||
419 |
if (k > 0) { |
||
420 |
SAMPLES_NEEDED |
||
421 |
//not debugged, almost never used |
||
422 |
memset(&coding_method[ch][sb][j + k], case_val, |
||
423 |
k *sizeof(int8_t)); |
||
424 |
memset(&coding_method[ch][sb][j + k], case_val, |
||
425 |
3 * sizeof(int8_t)); |
||
426 |
} |
||
427 |
} |
||
428 |
} |
||
429 |
} |
||
430 |
j += run; |
||
431 |
} |
||
432 |
} |
||
433 |
return 0; |
||
434 |
} |
||
435 |
|||
436 |
/** |
||
437 |
* Related to synthesis filter |
||
438 |
* Called by process_subpacket_10 |
||
439 |
* |
||
440 |
* @param q context |
||
441 |
* @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10 |
||
442 |
*/ |
||
443 |
139 |
static void fill_tone_level_array(QDM2Context *q, int flag) |
|
444 |
{ |
||
445 |
int i, sb, ch, sb_used; |
||
446 |
int tmp, tab; |
||
447 |
|||
448 |
✓✓ | 417 |
for (ch = 0; ch < q->nb_channels; ch++) |
449 |
✓✓ | 8618 |
for (sb = 0; sb < 30; sb++) |
450 |
✓✓ | 75060 |
for (i = 0; i < 8; i++) { |
451 |
✓✓ | 66720 |
if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1)) |
452 |
53376 |
tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+ |
|
453 |
53376 |
q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; |
|
454 |
else |
||
455 |
13344 |
tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb]; |
|
456 |
✗✓ | 66720 |
if(tmp < 0) |
457 |
tmp += 0xff; |
||
458 |
66720 |
q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff; |
|
459 |
} |
||
460 |
|||
461 |
✗✓ | 139 |
sb_used = QDM2_SB_USED(q->sub_sampling); |
462 |
|||
463 |
✓✗✓✗ |
139 |
if ((q->superblocktype_2_3 != 0) && !flag) { |
464 |
✓✓ | 4309 |
for (sb = 0; sb < sb_used; sb++) |
465 |
✓✓ | 12510 |
for (ch = 0; ch < q->nb_channels; ch++) |
466 |
✓✓ | 542100 |
for (i = 0; i < 64; i++) { |
467 |
533760 |
q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; |
|
468 |
✗✓ | 533760 |
if (q->tone_level_idx[ch][sb][i] < 0) |
469 |
q->tone_level[ch][sb][i] = 0; |
||
470 |
else |
||
471 |
533760 |
q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f]; |
|
472 |
} |
||
473 |
} else { |
||
474 |
tab = q->superblocktype_2_3 ? 0 : 1; |
||
475 |
for (sb = 0; sb < sb_used; sb++) { |
||
476 |
if ((sb >= 4) && (sb <= 23)) { |
||
477 |
for (ch = 0; ch < q->nb_channels; ch++) |
||
478 |
for (i = 0; i < 64; i++) { |
||
479 |
tmp = q->tone_level_idx_base[ch][sb][i / 8] - |
||
480 |
q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] - |
||
481 |
q->tone_level_idx_mid[ch][sb - 4][i / 8] - |
||
482 |
q->tone_level_idx_hi2[ch][sb - 4]; |
||
483 |
q->tone_level_idx[ch][sb][i] = tmp & 0xff; |
||
484 |
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
||
485 |
q->tone_level[ch][sb][i] = 0; |
||
486 |
else |
||
487 |
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; |
||
488 |
} |
||
489 |
} else { |
||
490 |
if (sb > 4) { |
||
491 |
for (ch = 0; ch < q->nb_channels; ch++) |
||
492 |
for (i = 0; i < 64; i++) { |
||
493 |
tmp = q->tone_level_idx_base[ch][sb][i / 8] - |
||
494 |
q->tone_level_idx_hi1[ch][2][i / 8][i % 8] - |
||
495 |
q->tone_level_idx_hi2[ch][sb - 4]; |
||
496 |
q->tone_level_idx[ch][sb][i] = tmp & 0xff; |
||
497 |
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
||
498 |
q->tone_level[ch][sb][i] = 0; |
||
499 |
else |
||
500 |
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; |
||
501 |
} |
||
502 |
} else { |
||
503 |
for (ch = 0; ch < q->nb_channels; ch++) |
||
504 |
for (i = 0; i < 64; i++) { |
||
505 |
tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8]; |
||
506 |
if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp)) |
||
507 |
q->tone_level[ch][sb][i] = 0; |
||
508 |
else |
||
509 |
q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f]; |
||
510 |
} |
||
511 |
} |
||
512 |
} |
||
513 |
} |
||
514 |
} |
||
515 |
139 |
} |
|
516 |
|||
517 |
/** |
||
518 |
* Related to synthesis filter |
||
519 |
* Called by process_subpacket_11 |
||
520 |
* c is built with data from subpacket 11 |
||
521 |
* Most of this function is used only if superblock_type_2_3 == 0, |
||
522 |
* never seen it in samples. |
||
523 |
* |
||
524 |
* @param tone_level_idx |
||
525 |
* @param tone_level_idx_temp |
||
526 |
* @param coding_method q->coding_method[0][0][0] |
||
527 |
* @param nb_channels number of channels |
||
528 |
* @param c coming from subpacket 11, passed as 8*c |
||
529 |
* @param superblocktype_2_3 flag based on superblock packet type |
||
530 |
* @param cm_table_select q->cm_table_select |
||
531 |
*/ |
||
532 |
static void fill_coding_method_array(sb_int8_array tone_level_idx, |
||
533 |
sb_int8_array tone_level_idx_temp, |
||
534 |
sb_int8_array coding_method, |
||
535 |
int nb_channels, |
||
536 |
int c, int superblocktype_2_3, |
||
537 |
int cm_table_select) |
||
538 |
{ |
||
539 |
int ch, sb, j; |
||
540 |
int tmp, acc, esp_40, comp; |
||
541 |
int add1, add2, add3, add4; |
||
542 |
int64_t multres; |
||
543 |
|||
544 |
if (!superblocktype_2_3) { |
||
545 |
/* This case is untested, no samples available */ |
||
546 |
avpriv_request_sample(NULL, "!superblocktype_2_3"); |
||
547 |
return; |
||
548 |
for (ch = 0; ch < nb_channels; ch++) { |
||
549 |
for (sb = 0; sb < 30; sb++) { |
||
550 |
for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer |
||
551 |
add1 = tone_level_idx[ch][sb][j] - 10; |
||
552 |
if (add1 < 0) |
||
553 |
add1 = 0; |
||
554 |
add2 = add3 = add4 = 0; |
||
555 |
if (sb > 1) { |
||
556 |
add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6; |
||
557 |
if (add2 < 0) |
||
558 |
add2 = 0; |
||
559 |
} |
||
560 |
if (sb > 0) { |
||
561 |
add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6; |
||
562 |
if (add3 < 0) |
||
563 |
add3 = 0; |
||
564 |
} |
||
565 |
if (sb < 29) { |
||
566 |
add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6; |
||
567 |
if (add4 < 0) |
||
568 |
add4 = 0; |
||
569 |
} |
||
570 |
tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1; |
||
571 |
if (tmp < 0) |
||
572 |
tmp = 0; |
||
573 |
tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff; |
||
574 |
} |
||
575 |
tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1]; |
||
576 |
} |
||
577 |
} |
||
578 |
acc = 0; |
||
579 |
for (ch = 0; ch < nb_channels; ch++) |
||
580 |
for (sb = 0; sb < 30; sb++) |
||
581 |
for (j = 0; j < 64; j++) |
||
582 |
acc += tone_level_idx_temp[ch][sb][j]; |
||
583 |
|||
584 |
multres = 0x66666667LL * (acc * 10); |
||
585 |
esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31); |
||
586 |
for (ch = 0; ch < nb_channels; ch++) |
||
587 |
for (sb = 0; sb < 30; sb++) |
||
588 |
for (j = 0; j < 64; j++) { |
||
589 |
comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10; |
||
590 |
if (comp < 0) |
||
591 |
comp += 0xff; |
||
592 |
comp /= 256; // signed shift |
||
593 |
switch(sb) { |
||
594 |
case 0: |
||
595 |
if (comp < 30) |
||
596 |
comp = 30; |
||
597 |
comp += 15; |
||
598 |
break; |
||
599 |
case 1: |
||
600 |
if (comp < 24) |
||
601 |
comp = 24; |
||
602 |
comp += 10; |
||
603 |
break; |
||
604 |
case 2: |
||
605 |
case 3: |
||
606 |
case 4: |
||
607 |
if (comp < 16) |
||
608 |
comp = 16; |
||
609 |
} |
||
610 |
if (comp <= 5) |
||
611 |
tmp = 0; |
||
612 |
else if (comp <= 10) |
||
613 |
tmp = 10; |
||
614 |
else if (comp <= 16) |
||
615 |
tmp = 16; |
||
616 |
else if (comp <= 24) |
||
617 |
tmp = -1; |
||
618 |
else |
||
619 |
tmp = 0; |
||
620 |
coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff; |
||
621 |
} |
||
622 |
for (sb = 0; sb < 30; sb++) |
||
623 |
fix_coding_method_array(sb, nb_channels, coding_method); |
||
624 |
for (ch = 0; ch < nb_channels; ch++) |
||
625 |
for (sb = 0; sb < 30; sb++) |
||
626 |
for (j = 0; j < 64; j++) |
||
627 |
if (sb >= 10) { |
||
628 |
if (coding_method[ch][sb][j] < 10) |
||
629 |
coding_method[ch][sb][j] = 10; |
||
630 |
} else { |
||
631 |
if (sb >= 2) { |
||
632 |
if (coding_method[ch][sb][j] < 16) |
||
633 |
coding_method[ch][sb][j] = 16; |
||
634 |
} else { |
||
635 |
if (coding_method[ch][sb][j] < 30) |
||
636 |
coding_method[ch][sb][j] = 30; |
||
637 |
} |
||
638 |
} |
||
639 |
} else { // superblocktype_2_3 != 0 |
||
640 |
for (ch = 0; ch < nb_channels; ch++) |
||
641 |
for (sb = 0; sb < 30; sb++) |
||
642 |
for (j = 0; j < 64; j++) |
||
643 |
coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb]; |
||
644 |
} |
||
645 |
} |
||
646 |
|||
647 |
/** |
||
648 |
* Called by process_subpacket_11 to process more data from subpacket 11 |
||
649 |
* with sb 0-8. |
||
650 |
* Called by process_subpacket_12 to process data from subpacket 12 with |
||
651 |
* sb 8-sb_used. |
||
652 |
* |
||
653 |
* @param q context |
||
654 |
* @param gb bitreader context |
||
655 |
* @param length packet length in bits |
||
656 |
* @param sb_min lower subband processed (sb_min included) |
||
657 |
* @param sb_max higher subband processed (sb_max excluded) |
||
658 |
*/ |
||
659 |
278 |
static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, |
|
660 |
int length, int sb_min, int sb_max) |
||
661 |
{ |
||
662 |
int sb, j, k, n, ch, run, channels; |
||
663 |
int joined_stereo, zero_encoding; |
||
664 |
int type34_first; |
||
665 |
278 |
float type34_div = 0; |
|
666 |
float type34_predictor; |
||
667 |
float samples[10]; |
||
668 |
278 |
int sign_bits[16] = {0}; |
|
669 |
|||
670 |
✓✗ | 278 |
if (length == 0) { |
671 |
// If no data use noise |
||
672 |
✓✓ | 4448 |
for (sb=sb_min; sb < sb_max; sb++) |
673 |
4170 |
build_sb_samples_from_noise(q, sb); |
|
674 |
|||
675 |
278 |
return 0; |
|
676 |
} |
||
677 |
|||
678 |
for (sb = sb_min; sb < sb_max; sb++) { |
||
679 |
channels = q->nb_channels; |
||
680 |
|||
681 |
if (q->nb_channels <= 1 || sb < 12) |
||
682 |
joined_stereo = 0; |
||
683 |
else if (sb >= 24) |
||
684 |
joined_stereo = 1; |
||
685 |
else |
||
686 |
joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; |
||
687 |
|||
688 |
if (joined_stereo) { |
||
689 |
if (get_bits_left(gb) >= 16) |
||
690 |
for (j = 0; j < 16; j++) |
||
691 |
sign_bits[j] = get_bits1(gb); |
||
692 |
|||
693 |
for (j = 0; j < 64; j++) |
||
694 |
if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j]) |
||
695 |
q->coding_method[0][sb][j] = q->coding_method[1][sb][j]; |
||
696 |
|||
697 |
if (fix_coding_method_array(sb, q->nb_channels, |
||
698 |
q->coding_method)) { |
||
699 |
av_log(NULL, AV_LOG_ERROR, "coding method invalid\n"); |
||
700 |
build_sb_samples_from_noise(q, sb); |
||
701 |
continue; |
||
702 |
} |
||
703 |
channels = 1; |
||
704 |
} |
||
705 |
|||
706 |
for (ch = 0; ch < channels; ch++) { |
||
707 |
FIX_NOISE_IDX(q->noise_idx); |
||
708 |
zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0; |
||
709 |
type34_predictor = 0.0; |
||
710 |
type34_first = 1; |
||
711 |
|||
712 |
for (j = 0; j < 128; ) { |
||
713 |
switch (q->coding_method[ch][sb][j / 2]) { |
||
714 |
case 8: |
||
715 |
if (get_bits_left(gb) >= 10) { |
||
716 |
if (zero_encoding) { |
||
717 |
for (k = 0; k < 5; k++) { |
||
718 |
if ((j + 2 * k) >= 128) |
||
719 |
break; |
||
720 |
samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0; |
||
721 |
} |
||
722 |
} else { |
||
723 |
n = get_bits(gb, 8); |
||
724 |
if (n >= 243) { |
||
725 |
av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); |
||
726 |
return AVERROR_INVALIDDATA; |
||
727 |
} |
||
728 |
|||
729 |
for (k = 0; k < 5; k++) |
||
730 |
samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; |
||
731 |
} |
||
732 |
for (k = 0; k < 5; k++) |
||
733 |
samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
734 |
} else { |
||
735 |
for (k = 0; k < 10; k++) |
||
736 |
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
737 |
} |
||
738 |
run = 10; |
||
739 |
break; |
||
740 |
|||
741 |
case 10: |
||
742 |
if (get_bits_left(gb) >= 1) { |
||
743 |
float f = 0.81; |
||
744 |
|||
745 |
if (get_bits1(gb)) |
||
746 |
f = -f; |
||
747 |
f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0; |
||
748 |
samples[0] = f; |
||
749 |
} else { |
||
750 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
751 |
} |
||
752 |
run = 1; |
||
753 |
break; |
||
754 |
|||
755 |
case 16: |
||
756 |
if (get_bits_left(gb) >= 10) { |
||
757 |
if (zero_encoding) { |
||
758 |
for (k = 0; k < 5; k++) { |
||
759 |
if ((j + k) >= 128) |
||
760 |
break; |
||
761 |
samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)]; |
||
762 |
} |
||
763 |
} else { |
||
764 |
n = get_bits (gb, 8); |
||
765 |
if (n >= 243) { |
||
766 |
av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n"); |
||
767 |
return AVERROR_INVALIDDATA; |
||
768 |
} |
||
769 |
|||
770 |
for (k = 0; k < 5; k++) |
||
771 |
samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]]; |
||
772 |
} |
||
773 |
} else { |
||
774 |
for (k = 0; k < 5; k++) |
||
775 |
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
776 |
} |
||
777 |
run = 5; |
||
778 |
break; |
||
779 |
|||
780 |
case 24: |
||
781 |
if (get_bits_left(gb) >= 7) { |
||
782 |
n = get_bits(gb, 7); |
||
783 |
if (n >= 125) { |
||
784 |
av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n"); |
||
785 |
return AVERROR_INVALIDDATA; |
||
786 |
} |
||
787 |
|||
788 |
for (k = 0; k < 3; k++) |
||
789 |
samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5; |
||
790 |
} else { |
||
791 |
for (k = 0; k < 3; k++) |
||
792 |
samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
793 |
} |
||
794 |
run = 3; |
||
795 |
break; |
||
796 |
|||
797 |
case 30: |
||
798 |
if (get_bits_left(gb) >= 4) { |
||
799 |
unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1); |
||
800 |
if (index >= FF_ARRAY_ELEMS(type30_dequant)) { |
||
801 |
av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index); |
||
802 |
return AVERROR_INVALIDDATA; |
||
803 |
} |
||
804 |
samples[0] = type30_dequant[index]; |
||
805 |
} else |
||
806 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
807 |
|||
808 |
run = 1; |
||
809 |
break; |
||
810 |
|||
811 |
case 34: |
||
812 |
if (get_bits_left(gb) >= 7) { |
||
813 |
if (type34_first) { |
||
814 |
type34_div = (float)(1 << get_bits(gb, 2)); |
||
815 |
samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0; |
||
816 |
type34_predictor = samples[0]; |
||
817 |
type34_first = 0; |
||
818 |
} else { |
||
819 |
unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1); |
||
820 |
if (index >= FF_ARRAY_ELEMS(type34_delta)) { |
||
821 |
av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index); |
||
822 |
return AVERROR_INVALIDDATA; |
||
823 |
} |
||
824 |
samples[0] = type34_delta[index] / type34_div + type34_predictor; |
||
825 |
type34_predictor = samples[0]; |
||
826 |
} |
||
827 |
} else { |
||
828 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
829 |
} |
||
830 |
run = 1; |
||
831 |
break; |
||
832 |
|||
833 |
default: |
||
834 |
samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx); |
||
835 |
run = 1; |
||
836 |
break; |
||
837 |
} |
||
838 |
|||
839 |
if (joined_stereo) { |
||
840 |
for (k = 0; k < run && j + k < 128; k++) { |
||
841 |
q->sb_samples[0][j + k][sb] = |
||
842 |
q->tone_level[0][sb][(j + k) / 2] * samples[k]; |
||
843 |
if (q->nb_channels == 2) { |
||
844 |
if (sign_bits[(j + k) / 8]) |
||
845 |
q->sb_samples[1][j + k][sb] = |
||
846 |
q->tone_level[1][sb][(j + k) / 2] * -samples[k]; |
||
847 |
else |
||
848 |
q->sb_samples[1][j + k][sb] = |
||
849 |
q->tone_level[1][sb][(j + k) / 2] * samples[k]; |
||
850 |
} |
||
851 |
} |
||
852 |
} else { |
||
853 |
for (k = 0; k < run; k++) |
||
854 |
if ((j + k) < 128) |
||
855 |
q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k]; |
||
856 |
} |
||
857 |
|||
858 |
j += run; |
||
859 |
} // j loop |
||
860 |
} // channel loop |
||
861 |
} // subband loop |
||
862 |
return 0; |
||
863 |
} |
||
864 |
|||
865 |
/** |
||
866 |
* Init the first element of a channel in quantized_coeffs with data |
||
867 |
* from packet 10 (quantized_coeffs[ch][0]). |
||
868 |
* This is similar to process_subpacket_9, but for a single channel |
||
869 |
* and for element [0] |
||
870 |
* same VLC tables as process_subpacket_9 are used. |
||
871 |
* |
||
872 |
* @param quantized_coeffs pointer to quantized_coeffs[ch][0] |
||
873 |
* @param gb bitreader context |
||
874 |
*/ |
||
875 |
static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, |
||
876 |
GetBitContext *gb) |
||
877 |
{ |
||
878 |
int i, k, run, level, diff; |
||
879 |
|||
880 |
if (get_bits_left(gb) < 16) |
||
881 |
return -1; |
||
882 |
level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2); |
||
883 |
|||
884 |
quantized_coeffs[0] = level; |
||
885 |
|||
886 |
for (i = 0; i < 7; ) { |
||
887 |
if (get_bits_left(gb) < 16) |
||
888 |
return -1; |
||
889 |
run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1; |
||
890 |
|||
891 |
if (i + run >= 8) |
||
892 |
return -1; |
||
893 |
|||
894 |
if (get_bits_left(gb) < 16) |
||
895 |
return -1; |
||
896 |
diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2); |
||
897 |
|||
898 |
for (k = 1; k <= run; k++) |
||
899 |
quantized_coeffs[i + k] = (level + ((k * diff) / run)); |
||
900 |
|||
901 |
level += diff; |
||
902 |
i += run; |
||
903 |
} |
||
904 |
return 0; |
||
905 |
} |
||
906 |
|||
907 |
/** |
||
908 |
* Related to synthesis filter, process data from packet 10 |
||
909 |
* Init part of quantized_coeffs via function init_quantized_coeffs_elem0 |
||
910 |
* Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with |
||
911 |
* data from packet 10 |
||
912 |
* |
||
913 |
* @param q context |
||
914 |
* @param gb bitreader context |
||
915 |
*/ |
||
916 |
static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb) |
||
917 |
{ |
||
918 |
int sb, j, k, n, ch; |
||
919 |
|||
920 |
for (ch = 0; ch < q->nb_channels; ch++) { |
||
921 |
init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb); |
||
922 |
|||
923 |
if (get_bits_left(gb) < 16) { |
||
924 |
memset(q->quantized_coeffs[ch][0], 0, 8); |
||
925 |
break; |
||
926 |
} |
||
927 |
} |
||
928 |
|||
929 |
n = q->sub_sampling + 1; |
||
930 |
|||
931 |
for (sb = 0; sb < n; sb++) |
||
932 |
for (ch = 0; ch < q->nb_channels; ch++) |
||
933 |
for (j = 0; j < 8; j++) { |
||
934 |
if (get_bits_left(gb) < 1) |
||
935 |
break; |
||
936 |
if (get_bits1(gb)) { |
||
937 |
for (k=0; k < 8; k++) { |
||
938 |
if (get_bits_left(gb) < 16) |
||
939 |
break; |
||
940 |
q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2); |
||
941 |
} |
||
942 |
} else { |
||
943 |
for (k=0; k < 8; k++) |
||
944 |
q->tone_level_idx_hi1[ch][sb][j][k] = 0; |
||
945 |
} |
||
946 |
} |
||
947 |
|||
948 |
n = QDM2_SB_USED(q->sub_sampling) - 4; |
||
949 |
|||
950 |
for (sb = 0; sb < n; sb++) |
||
951 |
for (ch = 0; ch < q->nb_channels; ch++) { |
||
952 |
if (get_bits_left(gb) < 16) |
||
953 |
break; |
||
954 |
q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2); |
||
955 |
if (sb > 19) |
||
956 |
q->tone_level_idx_hi2[ch][sb] -= 16; |
||
957 |
else |
||
958 |
for (j = 0; j < 8; j++) |
||
959 |
q->tone_level_idx_mid[ch][sb][j] = -16; |
||
960 |
} |
||
961 |
|||
962 |
n = QDM2_SB_USED(q->sub_sampling) - 5; |
||
963 |
|||
964 |
for (sb = 0; sb < n; sb++) |
||
965 |
for (ch = 0; ch < q->nb_channels; ch++) |
||
966 |
for (j = 0; j < 8; j++) { |
||
967 |
if (get_bits_left(gb) < 16) |
||
968 |
break; |
||
969 |
q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32; |
||
970 |
} |
||
971 |
} |
||
972 |
|||
973 |
/** |
||
974 |
* Process subpacket 9, init quantized_coeffs with data from it |
||
975 |
* |
||
976 |
* @param q context |
||
977 |
* @param node pointer to node with packet |
||
978 |
*/ |
||
979 |
139 |
static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node) |
|
980 |
{ |
||
981 |
GetBitContext gb; |
||
982 |
int i, j, k, n, ch, run, level, diff; |
||
983 |
|||
984 |
139 |
init_get_bits(&gb, node->packet->data, node->packet->size * 8); |
|
985 |
|||
986 |
✗✓ | 139 |
n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; |
987 |
|||
988 |
✓✓ | 1390 |
for (i = 1; i < n; i++) |
989 |
✓✓ | 3753 |
for (ch = 0; ch < q->nb_channels; ch++) { |
990 |
2502 |
level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2); |
|
991 |
2502 |
q->quantized_coeffs[ch][i][0] = level; |
|
992 |
|||
993 |
✓✓ | 13498 |
for (j = 0; j < (8 - 1); ) { |
994 |
10996 |
run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1; |
|
995 |
10996 |
diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2); |
|
996 |
|||
997 |
✗✓ | 10996 |
if (j + run >= 8) |
998 |
return -1; |
||
999 |
|||
1000 |
✓✓ | 28510 |
for (k = 1; k <= run; k++) |
1001 |
17514 |
q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run)); |
|
1002 |
|||
1003 |
10996 |
level += diff; |
|
1004 |
10996 |
j += run; |
|
1005 |
} |
||
1006 |
} |
||
1007 |
|||
1008 |
✓✓ | 417 |
for (ch = 0; ch < q->nb_channels; ch++) |
1009 |
✓✓ | 2502 |
for (i = 0; i < 8; i++) |
1010 |
2224 |
q->quantized_coeffs[ch][0][i] = 0; |
|
1011 |
|||
1012 |
139 |
return 0; |
|
1013 |
} |
||
1014 |
|||
1015 |
/** |
||
1016 |
* Process subpacket 10 if not null, else |
||
1017 |
* |
||
1018 |
* @param q context |
||
1019 |
* @param node pointer to node with packet |
||
1020 |
*/ |
||
1021 |
139 |
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node) |
|
1022 |
{ |
||
1023 |
GetBitContext gb; |
||
1024 |
|||
1025 |
✗✓ | 139 |
if (node) { |
1026 |
init_get_bits(&gb, node->packet->data, node->packet->size * 8); |
||
1027 |
init_tone_level_dequantization(q, &gb); |
||
1028 |
fill_tone_level_array(q, 1); |
||
1029 |
} else { |
||
1030 |
139 |
fill_tone_level_array(q, 0); |
|
1031 |
} |
||
1032 |
139 |
} |
|
1033 |
|||
1034 |
/** |
||
1035 |
* Process subpacket 11 |
||
1036 |
* |
||
1037 |
* @param q context |
||
1038 |
* @param node pointer to node with packet |
||
1039 |
*/ |
||
1040 |
139 |
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node) |
|
1041 |
{ |
||
1042 |
GetBitContext gb; |
||
1043 |
139 |
int length = 0; |
|
1044 |
|||
1045 |
✗✓ | 139 |
if (node) { |
1046 |
length = node->packet->size * 8; |
||
1047 |
init_get_bits(&gb, node->packet->data, length); |
||
1048 |
} |
||
1049 |
|||
1050 |
✗✓ | 139 |
if (length >= 32) { |
1051 |
int c = get_bits(&gb, 13); |
||
1052 |
|||
1053 |
if (c > 3) |
||
1054 |
fill_coding_method_array(q->tone_level_idx, |
||
1055 |
q->tone_level_idx_temp, q->coding_method, |
||
1056 |
q->nb_channels, 8 * c, |
||
1057 |
q->superblocktype_2_3, q->cm_table_select); |
||
1058 |
} |
||
1059 |
|||
1060 |
139 |
synthfilt_build_sb_samples(q, &gb, length, 0, 8); |
|
1061 |
139 |
} |
|
1062 |
|||
1063 |
/** |
||
1064 |
* Process subpacket 12 |
||
1065 |
* |
||
1066 |
* @param q context |
||
1067 |
* @param node pointer to node with packet |
||
1068 |
*/ |
||
1069 |
139 |
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node) |
|
1070 |
{ |
||
1071 |
GetBitContext gb; |
||
1072 |
139 |
int length = 0; |
|
1073 |
|||
1074 |
✗✓ | 139 |
if (node) { |
1075 |
length = node->packet->size * 8; |
||
1076 |
init_get_bits(&gb, node->packet->data, length); |
||
1077 |
} |
||
1078 |
|||
1079 |
✗✓ | 139 |
synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling)); |
1080 |
139 |
} |
|
1081 |
|||
1082 |
/** |
||
1083 |
* Process new subpackets for synthesis filter |
||
1084 |
* |
||
1085 |
* @param q context |
||
1086 |
* @param list list with synthesis filter packets (list D) |
||
1087 |
*/ |
||
1088 |
139 |
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list) |
|
1089 |
{ |
||
1090 |
QDM2SubPNode *nodes[4]; |
||
1091 |
|||
1092 |
139 |
nodes[0] = qdm2_search_subpacket_type_in_list(list, 9); |
|
1093 |
✓✗ | 139 |
if (nodes[0]) |
1094 |
139 |
process_subpacket_9(q, nodes[0]); |
|
1095 |
|||
1096 |
139 |
nodes[1] = qdm2_search_subpacket_type_in_list(list, 10); |
|
1097 |
✗✓ | 139 |
if (nodes[1]) |
1098 |
process_subpacket_10(q, nodes[1]); |
||
1099 |
else |
||
1100 |
139 |
process_subpacket_10(q, NULL); |
|
1101 |
|||
1102 |
139 |
nodes[2] = qdm2_search_subpacket_type_in_list(list, 11); |
|
1103 |
✓✗✗✓ ✗✗ |
139 |
if (nodes[0] && nodes[1] && nodes[2]) |
1104 |
process_subpacket_11(q, nodes[2]); |
||
1105 |
else |
||
1106 |
139 |
process_subpacket_11(q, NULL); |
|
1107 |
|||
1108 |
139 |
nodes[3] = qdm2_search_subpacket_type_in_list(list, 12); |
|
1109 |
✓✗✗✓ ✗✗ |
139 |
if (nodes[0] && nodes[1] && nodes[3]) |
1110 |
process_subpacket_12(q, nodes[3]); |
||
1111 |
else |
||
1112 |
139 |
process_subpacket_12(q, NULL); |
|
1113 |
139 |
} |
|
1114 |
|||
1115 |
/** |
||
1116 |
* Decode superblock, fill packet lists. |
||
1117 |
* |
||
1118 |
* @param q context |
||
1119 |
*/ |
||
1120 |
139 |
static void qdm2_decode_super_block(QDM2Context *q) |
|
1121 |
{ |
||
1122 |
GetBitContext gb; |
||
1123 |
QDM2SubPacket header, *packet; |
||
1124 |
int i, packet_bytes, sub_packet_size, sub_packets_D; |
||
1125 |
139 |
unsigned int next_index = 0; |
|
1126 |
|||
1127 |
139 |
memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1)); |
|
1128 |
139 |
memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid)); |
|
1129 |
139 |
memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2)); |
|
1130 |
|||
1131 |
139 |
q->sub_packets_B = 0; |
|
1132 |
139 |
sub_packets_D = 0; |
|
1133 |
|||
1134 |
139 |
average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8] |
|
1135 |
|||
1136 |
139 |
init_get_bits(&gb, q->compressed_data, q->compressed_size * 8); |
|
1137 |
139 |
qdm2_decode_sub_packet_header(&gb, &header); |
|
1138 |
|||
1139 |
✓✗✗✓ |
139 |
if (header.type < 2 || header.type >= 8) { |
1140 |
q->has_errors = 1; |
||
1141 |
av_log(NULL, AV_LOG_ERROR, "bad superblock type\n"); |
||
1142 |
return; |
||
1143 |
} |
||
1144 |
|||
1145 |
✗✓✗✗ |
139 |
q->superblocktype_2_3 = (header.type == 2 || header.type == 3); |
1146 |
139 |
packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8); |
|
1147 |
|||
1148 |
139 |
init_get_bits(&gb, header.data, header.size * 8); |
|
1149 |
|||
1150 |
✗✓✗✗ ✗✗ |
139 |
if (header.type == 2 || header.type == 4 || header.type == 5) { |
1151 |
139 |
int csum = 257 * get_bits(&gb, 8); |
|
1152 |
139 |
csum += 2 * get_bits(&gb, 8); |
|
1153 |
|||
1154 |
139 |
csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum); |
|
1155 |
|||
1156 |
✗✓ | 139 |
if (csum != 0) { |
1157 |
q->has_errors = 1; |
||
1158 |
av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n"); |
||
1159 |
return; |
||
1160 |
} |
||
1161 |
} |
||
1162 |
|||
1163 |
139 |
q->sub_packet_list_B[0].packet = NULL; |
|
1164 |
139 |
q->sub_packet_list_D[0].packet = NULL; |
|
1165 |
|||
1166 |
✓✓ | 973 |
for (i = 0; i < 6; i++) |
1167 |
✓✗ | 834 |
if (--q->fft_level_exp[i] < 0) |
1168 |
834 |
q->fft_level_exp[i] = 0; |
|
1169 |
|||
1170 |
✓✗ | 973 |
for (i = 0; packet_bytes > 0; i++) { |
1171 |
int j; |
||
1172 |
|||
1173 |
✗✓ | 973 |
if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) { |
1174 |
SAMPLES_NEEDED_2("too many packet bytes"); |
||
1175 |
return; |
||
1176 |
} |
||
1177 |
|||
1178 |
973 |
q->sub_packet_list_A[i].next = NULL; |
|
1179 |
|||
1180 |
✓✓ | 973 |
if (i > 0) { |
1181 |
834 |
q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i]; |
|
1182 |
|||
1183 |
/* seek to next block */ |
||
1184 |
834 |
init_get_bits(&gb, header.data, header.size * 8); |
|
1185 |
834 |
skip_bits(&gb, next_index * 8); |
|
1186 |
|||
1187 |
✗✓ | 834 |
if (next_index >= header.size) |
1188 |
break; |
||
1189 |
} |
||
1190 |
|||
1191 |
/* decode subpacket */ |
||
1192 |
973 |
packet = &q->sub_packets[i]; |
|
1193 |
973 |
qdm2_decode_sub_packet_header(&gb, packet); |
|
1194 |
973 |
next_index = packet->size + get_bits_count(&gb) / 8; |
|
1195 |
✗✓ | 973 |
sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2; |
1196 |
|||
1197 |
✓✓ | 973 |
if (packet->type == 0) |
1198 |
139 |
break; |
|
1199 |
|||
1200 |
✗✓ | 834 |
if (sub_packet_size > packet_bytes) { |
1201 |
if (packet->type != 10 && packet->type != 11 && packet->type != 12) |
||
1202 |
break; |
||
1203 |
packet->size += packet_bytes - sub_packet_size; |
||
1204 |
} |
||
1205 |
|||
1206 |
834 |
packet_bytes -= sub_packet_size; |
|
1207 |
|||
1208 |
/* add subpacket to 'all subpackets' list */ |
||
1209 |
834 |
q->sub_packet_list_A[i].packet = packet; |
|
1210 |
|||
1211 |
/* add subpacket to related list */ |
||
1212 |
✗✓ | 834 |
if (packet->type == 8) { |
1213 |
SAMPLES_NEEDED_2("packet type 8"); |
||
1214 |
return; |
||
1215 |
✓✗✓✓ |
834 |
} else if (packet->type >= 9 && packet->type <= 12) { |
1216 |
/* packets for MPEG Audio like Synthesis Filter */ |
||
1217 |
✗✓ | 139 |
QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet); |
1218 |
✗✓ | 695 |
} else if (packet->type == 13) { |
1219 |
for (j = 0; j < 6; j++) |
||
1220 |
q->fft_level_exp[j] = get_bits(&gb, 6); |
||
1221 |
✗✓ | 695 |
} else if (packet->type == 14) { |
1222 |
for (j = 0; j < 6; j++) |
||
1223 |
q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2); |
||
1224 |
✗✓ | 695 |
} else if (packet->type == 15) { |
1225 |
SAMPLES_NEEDED_2("packet type 15") |
||
1226 |
return; |
||
1227 |
✓✗✓✗ |
695 |
} else if (packet->type >= 16 && packet->type < 48 && |
1228 |
✓✗ | 695 |
!fft_subpackets[packet->type - 16]) { |
1229 |
/* packets for FFT */ |
||
1230 |
✓✓ | 695 |
QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet); |
1231 |
} |
||
1232 |
} // Packet bytes loop |
||
1233 |
|||
1234 |
✓✗ | 139 |
if (q->sub_packet_list_D[0].packet) { |
1235 |
139 |
process_synthesis_subpackets(q, q->sub_packet_list_D); |
|
1236 |
139 |
q->do_synth_filter = 1; |
|
1237 |
} else if (q->do_synth_filter) { |
||
1238 |
process_subpacket_10(q, NULL); |
||
1239 |
process_subpacket_11(q, NULL); |
||
1240 |
process_subpacket_12(q, NULL); |
||
1241 |
} |
||
1242 |
} |
||
1243 |
|||
1244 |
20097 |
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, |
|
1245 |
int offset, int duration, int channel, |
||
1246 |
int exp, int phase) |
||
1247 |
{ |
||
1248 |
✓✓ | 20097 |
if (q->fft_coefs_min_index[duration] < 0) |
1249 |
545 |
q->fft_coefs_min_index[duration] = q->fft_coefs_index; |
|
1250 |
|||
1251 |
✓✓ | 20097 |
q->fft_coefs[q->fft_coefs_index].sub_packet = |
1252 |
288 |
((sub_packet >= 16) ? (sub_packet - 16) : sub_packet); |
|
1253 |
20097 |
q->fft_coefs[q->fft_coefs_index].channel = channel; |
|
1254 |
20097 |
q->fft_coefs[q->fft_coefs_index].offset = offset; |
|
1255 |
20097 |
q->fft_coefs[q->fft_coefs_index].exp = exp; |
|
1256 |
20097 |
q->fft_coefs[q->fft_coefs_index].phase = phase; |
|
1257 |
20097 |
q->fft_coefs_index++; |
|
1258 |
20097 |
} |
|
1259 |
|||
1260 |
556 |
static void qdm2_fft_decode_tones(QDM2Context *q, int duration, |
|
1261 |
GetBitContext *gb, int b) |
||
1262 |
{ |
||
1263 |
int channel, stereo, phase, exp; |
||
1264 |
int local_int_4, local_int_8, stereo_phase, local_int_10; |
||
1265 |
int local_int_14, stereo_exp, local_int_20, local_int_28; |
||
1266 |
int n, offset; |
||
1267 |
|||
1268 |
556 |
local_int_4 = 0; |
|
1269 |
556 |
local_int_28 = 0; |
|
1270 |
556 |
local_int_20 = 2; |
|
1271 |
556 |
local_int_8 = (4 - duration); |
|
1272 |
556 |
local_int_10 = 1 << (q->group_order - duration - 1); |
|
1273 |
556 |
offset = 1; |
|
1274 |
|||
1275 |
✓✗ | 16211 |
while (get_bits_left(gb)>0) { |
1276 |
✓✗ | 16211 |
if (q->superblocktype_2_3) { |
1277 |
✓✓ | 18212 |
while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) { |
1278 |
✓✓ | 2068 |
if (get_bits_left(gb)<0) { |
1279 |
✗✓ | 67 |
if(local_int_4 < q->group_size) |
1280 |
av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n"); |
||
1281 |
67 |
return; |
|
1282 |
} |
||
1283 |
2001 |
offset = 1; |
|
1284 |
✓✓ | 2001 |
if (n == 0) { |
1285 |
1989 |
local_int_4 += local_int_10; |
|
1286 |
1989 |
local_int_28 += (1 << local_int_8); |
|
1287 |
} else { |
||
1288 |
12 |
local_int_4 += 8 * local_int_10; |
|
1289 |
12 |
local_int_28 += (8 << local_int_8); |
|
1290 |
} |
||
1291 |
} |
||
1292 |
16144 |
offset += (n - 2); |
|
1293 |
} else { |
||
1294 |
if (local_int_10 <= 2) { |
||
1295 |
av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n"); |
||
1296 |
return; |
||
1297 |
} |
||
1298 |
offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2); |
||
1299 |
while (offset >= (local_int_10 - 1)) { |
||
1300 |
offset += (1 - (local_int_10 - 1)); |
||
1301 |
local_int_4 += local_int_10; |
||
1302 |
local_int_28 += (1 << local_int_8); |
||
1303 |
} |
||
1304 |
} |
||
1305 |
|||
1306 |
✓✓ | 16144 |
if (local_int_4 >= q->group_size) |
1307 |
489 |
return; |
|
1308 |
|||
1309 |
15655 |
local_int_14 = (offset >> local_int_8); |
|
1310 |
✗✓ | 15655 |
if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table)) |
1311 |
return; |
||
1312 |
|||
1313 |
✓✗ | 15655 |
if (q->nb_channels > 1) { |
1314 |
15655 |
channel = get_bits1(gb); |
|
1315 |
15655 |
stereo = get_bits1(gb); |
|
1316 |
} else { |
||
1317 |
channel = 0; |
||
1318 |
stereo = 0; |
||
1319 |
} |
||
1320 |
|||
1321 |
✗✓ | 15655 |
exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2); |
1322 |
15655 |
exp += q->fft_level_exp[fft_level_index_table[local_int_14]]; |
|
1323 |
15655 |
exp = (exp < 0) ? 0 : exp; |
|
1324 |
|||
1325 |
15655 |
phase = get_bits(gb, 3); |
|
1326 |
15655 |
stereo_exp = 0; |
|
1327 |
15655 |
stereo_phase = 0; |
|
1328 |
|||
1329 |
✓✓ | 15655 |
if (stereo) { |
1330 |
4442 |
stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1)); |
|
1331 |
4442 |
stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1)); |
|
1332 |
✓✓ | 4442 |
if (stereo_phase < 0) |
1333 |
497 |
stereo_phase += 8; |
|
1334 |
} |
||
1335 |
|||
1336 |
✓✗ | 15655 |
if (q->frequency_range > (local_int_14 + 1)) { |
1337 |
15655 |
int sub_packet = (local_int_20 + local_int_28); |
|
1338 |
|||
1339 |
✗✓ | 15655 |
if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs)) |
1340 |
return; |
||
1341 |
|||
1342 |
15655 |
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, |
|
1343 |
channel, exp, phase); |
||
1344 |
✓✓ | 15655 |
if (stereo) |
1345 |
4442 |
qdm2_fft_init_coefficient(q, sub_packet, offset, duration, |
|
1346 |
1 - channel, |
||
1347 |
stereo_exp, stereo_phase); |
||
1348 |
} |
||
1349 |
15655 |
offset++; |
|
1350 |
} |
||
1351 |
} |
||
1352 |
|||
1353 |
139 |
static void qdm2_decode_fft_packets(QDM2Context *q) |
|
1354 |
{ |
||
1355 |
int i, j, min, max, value, type, unknown_flag; |
||
1356 |
GetBitContext gb; |
||
1357 |
|||
1358 |
✗✓ | 139 |
if (!q->sub_packet_list_B[0].packet) |
1359 |
return; |
||
1360 |
|||
1361 |
/* reset minimum indexes for FFT coefficients */ |
||
1362 |
139 |
q->fft_coefs_index = 0; |
|
1363 |
✓✓ | 834 |
for (i = 0; i < 5; i++) |
1364 |
695 |
q->fft_coefs_min_index[i] = -1; |
|
1365 |
|||
1366 |
/* process subpackets ordered by type, largest type first */ |
||
1367 |
✓✓ | 834 |
for (i = 0, max = 256; i < q->sub_packets_B; i++) { |
1368 |
695 |
QDM2SubPacket *packet = NULL; |
|
1369 |
|||
1370 |
/* find subpacket with largest type less than max */ |
||
1371 |
✓✓ | 4170 |
for (j = 0, min = 0; j < q->sub_packets_B; j++) { |
1372 |
3475 |
value = q->sub_packet_list_B[j].packet->type; |
|
1373 |
✓✓✓✓ |
3475 |
if (value > min && value < max) { |
1374 |
695 |
min = value; |
|
1375 |
695 |
packet = q->sub_packet_list_B[j].packet; |
|
1376 |
} |
||
1377 |
} |
||
1378 |
|||
1379 |
695 |
max = min; |
|
1380 |
|||
1381 |
/* check for errors (?) */ |
||
1382 |
✗✓ | 695 |
if (!packet) |
1383 |
return; |
||
1384 |
|||
1385 |
✓✓ | 695 |
if (i == 0 && |
1386 |
✓✗✓✗ |
139 |
(packet->type < 16 || packet->type >= 48 || |
1387 |
✗✓ | 139 |
fft_subpackets[packet->type - 16])) |
1388 |
return; |
||
1389 |
|||
1390 |
/* decode FFT tones */ |
||
1391 |
695 |
init_get_bits(&gb, packet->data, packet->size * 8); |
|
1392 |
|||
1393 |
✗✓✗✗ ✗✗ |
695 |
if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16]) |
1394 |
unknown_flag = 1; |
||
1395 |
else |
||
1396 |
695 |
unknown_flag = 0; |
|
1397 |
|||
1398 |
695 |
type = packet->type; |
|
1399 |
|||
1400 |
✓✗✗✓ ✗✗✗✗ |
1390 |
if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) { |
1401 |
695 |
int duration = q->sub_sampling + 5 - (type & 15); |
|
1402 |
|||
1403 |
✓✗✓✓ |
695 |
if (duration >= 0 && duration < 4) |
1404 |
556 |
qdm2_fft_decode_tones(q, duration, &gb, unknown_flag); |
|
1405 |
} else if (type == 31) { |
||
1406 |
for (j = 0; j < 4; j++) |
||
1407 |
qdm2_fft_decode_tones(q, j, &gb, unknown_flag); |
||
1408 |
} else if (type == 46) { |
||
1409 |
for (j = 0; j < 6; j++) |
||
1410 |
q->fft_level_exp[j] = get_bits(&gb, 6); |
||
1411 |
for (j = 0; j < 4; j++) |
||
1412 |
qdm2_fft_decode_tones(q, j, &gb, unknown_flag); |
||
1413 |
} |
||
1414 |
} // Loop on B packets |
||
1415 |
|||
1416 |
/* calculate maximum indexes for FFT coefficients */ |
||
1417 |
✓✓ | 834 |
for (i = 0, j = -1; i < 5; i++) |
1418 |
✓✓ | 695 |
if (q->fft_coefs_min_index[i] >= 0) { |
1419 |
✓✓ | 545 |
if (j >= 0) |
1420 |
406 |
q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i]; |
|
1421 |
545 |
j = i; |
|
1422 |
} |
||
1423 |
✓✗ | 139 |
if (j >= 0) |
1424 |
139 |
q->fft_coefs_max_index[j] = q->fft_coefs_index; |
|
1425 |
} |
||
1426 |
|||
1427 |
341224 |
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone) |
|
1428 |
{ |
||
1429 |
float level, f[6]; |
||
1430 |
int i; |
||
1431 |
QDM2Complex c; |
||
1432 |
341224 |
const double iscale = 2.0 * M_PI / 512.0; |
|
1433 |
|||
1434 |
341224 |
tone->phase += tone->phase_shift; |
|
1435 |
|||
1436 |
/* calculate current level (maximum amplitude) of tone */ |
||
1437 |
341224 |
level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level; |
|
1438 |
341224 |
c.im = level * sin(tone->phase * iscale); |
|
1439 |
341224 |
c.re = level * cos(tone->phase * iscale); |
|
1440 |
|||
1441 |
/* generate FFT coefficients for tone */ |
||
1442 |
✓✓✓✓ |
341224 |
if (tone->duration >= 3 || tone->cutoff >= 3) { |
1443 |
46517 |
tone->complex[0].im += c.im; |
|
1444 |
46517 |
tone->complex[0].re += c.re; |
|
1445 |
46517 |
tone->complex[1].im -= c.im; |
|
1446 |
46517 |
tone->complex[1].re -= c.re; |
|
1447 |
} else { |
||
1448 |
294707 |
f[1] = -tone->table[4]; |
|
1449 |
294707 |
f[0] = tone->table[3] - tone->table[0]; |
|
1450 |
294707 |
f[2] = 1.0 - tone->table[2] - tone->table[3]; |
|
1451 |
294707 |
f[3] = tone->table[1] + tone->table[4] - 1.0; |
|
1452 |
294707 |
f[4] = tone->table[0] - tone->table[1]; |
|
1453 |
294707 |
f[5] = tone->table[2]; |
|
1454 |
✓✓ | 884121 |
for (i = 0; i < 2; i++) { |
1455 |
589414 |
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += |
|
1456 |
589414 |
c.re * f[i]; |
|
1457 |
1178828 |
tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += |
|
1458 |
✓✓ | 589414 |
c.im * ((tone->cutoff <= i) ? -f[i] : f[i]); |
1459 |
} |
||
1460 |
✓✓ | 1473535 |
for (i = 0; i < 4; i++) { |
1461 |
1178828 |
tone->complex[i].re += c.re * f[i + 2]; |
|
1462 |
1178828 |
tone->complex[i].im += c.im * f[i + 2]; |
|
1463 |
} |
||
1464 |
} |
||
1465 |
|||
1466 |
/* copy the tone if it has not yet died out */ |
||
1467 |
✓✓ | 341224 |
if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) { |
1468 |
321446 |
memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone)); |
|
1469 |
321446 |
q->fft_tone_end = (q->fft_tone_end + 1) % 1000; |
|
1470 |
} |
||
1471 |
341224 |
} |
|
1472 |
|||
1473 |
2224 |
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet) |
|
1474 |
{ |
||
1475 |
int i, j, ch; |
||
1476 |
2224 |
const double iscale = 0.25 * M_PI; |
|
1477 |
|||
1478 |
✓✓ | 6672 |
for (ch = 0; ch < q->channels; ch++) { |
1479 |
4448 |
memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex)); |
|
1480 |
} |
||
1481 |
|||
1482 |
|||
1483 |
/* apply FFT tones with duration 4 (1 FFT period) */ |
||
1484 |
✓✓ | 2224 |
if (q->fft_coefs_min_index[4] >= 0) |
1485 |
✗✓ | 4 |
for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) { |
1486 |
float level; |
||
1487 |
QDM2Complex c; |
||
1488 |
|||
1489 |
if (q->fft_coefs[i].sub_packet != sub_packet) |
||
1490 |
break; |
||
1491 |
|||
1492 |
ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel; |
||
1493 |
level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63]; |
||
1494 |
|||
1495 |
c.re = level * cos(q->fft_coefs[i].phase * iscale); |
||
1496 |
c.im = level * sin(q->fft_coefs[i].phase * iscale); |
||
1497 |
q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re; |
||
1498 |
q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im; |
||
1499 |
q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re; |
||
1500 |
q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im; |
||
1501 |
} |
||
1502 |
|||
1503 |
/* generate existing FFT tones */ |
||
1504 |
✓✓ | 323355 |
for (i = q->fft_tone_end; i != q->fft_tone_start; ) { |
1505 |
321131 |
qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]); |
|
1506 |
321131 |
q->fft_tone_start = (q->fft_tone_start + 1) % 1000; |
|
1507 |
} |
||
1508 |
|||
1509 |
/* create and generate new FFT tones with duration 0 (long) to 3 (short) */ |
||
1510 |
✓✓ | 11120 |
for (i = 0; i < 4; i++) |
1511 |
✓✓ | 8896 |
if (q->fft_coefs_min_index[i] >= 0) { |
1512 |
✓✓ | 28817 |
for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) { |
1513 |
int offset, four_i; |
||
1514 |
FFTTone tone; |
||
1515 |
|||
1516 |
✓✓ | 24369 |
if (q->fft_coefs[j].sub_packet != sub_packet) |
1517 |
4276 |
break; |
|
1518 |
|||
1519 |
20093 |
four_i = (4 - i); |
|
1520 |
20093 |
offset = q->fft_coefs[j].offset >> four_i; |
|
1521 |
✓✗ | 20093 |
ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel; |
1522 |
|||
1523 |
✓✗ | 20093 |
if (offset < q->frequency_range) { |
1524 |
✓✓ | 20093 |
if (offset < 2) |
1525 |
2785 |
tone.cutoff = offset; |
|
1526 |
else |
||
1527 |
✓✓ | 17308 |
tone.cutoff = (offset >= 60) ? 3 : 2; |
1528 |
|||
1529 |
✓✗ | 20093 |
tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63]; |
1530 |
20093 |
tone.complex = &q->fft.complex[ch][offset]; |
|
1531 |
20093 |
tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)]; |
|
1532 |
20093 |
tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128; |
|
1533 |
20093 |
tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i); |
|
1534 |
20093 |
tone.duration = i; |
|
1535 |
20093 |
tone.time_index = 0; |
|
1536 |
|||
1537 |
20093 |
qdm2_fft_generate_tone(q, &tone); |
|
1538 |
} |
||
1539 |
} |
||
1540 |
8724 |
q->fft_coefs_min_index[i] = j; |
|
1541 |
} |
||
1542 |
2224 |
} |
|
1543 |
|||
1544 |
4448 |
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet) |
|
1545 |
{ |
||
1546 |
✗✓✗✗ |
4448 |
const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f; |
1547 |
4448 |
float *out = q->output_buffer + channel; |
|
1548 |
int i; |
||
1549 |
4448 |
q->fft.complex[channel][0].re *= 2.0f; |
|
1550 |
4448 |
q->fft.complex[channel][0].im = 0.0f; |
|
1551 |
4448 |
q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]); |
|
1552 |
/* add samples to output buffer */ |
||
1553 |
✓✓ | 1143136 |
for (i = 0; i < FFALIGN(q->fft_size, 8); i++) { |
1554 |
1138688 |
out[0] += q->fft.complex[channel][i].re * gain; |
|
1555 |
1138688 |
out[q->channels] += q->fft.complex[channel][i].im * gain; |
|
1556 |
1138688 |
out += 2 * q->channels; |
|
1557 |
} |
||
1558 |
4448 |
} |
|
1559 |
|||
1560 |
/** |
||
1561 |
* @param q context |
||
1562 |
* @param index subpacket number |
||
1563 |
*/ |
||
1564 |
2224 |
static void qdm2_synthesis_filter(QDM2Context *q, int index) |
|
1565 |
{ |
||
1566 |
2224 |
int i, k, ch, sb_used, sub_sampling, dither_state = 0; |
|
1567 |
|||
1568 |
/* copy sb_samples */ |
||
1569 |
✗✓ | 2224 |
sb_used = QDM2_SB_USED(q->sub_sampling); |
1570 |
|||
1571 |
✓✓ | 6672 |
for (ch = 0; ch < q->channels; ch++) |
1572 |
✓✓ | 40032 |
for (i = 0; i < 8; i++) |
1573 |
✓✓ | 106752 |
for (k = sb_used; k < SBLIMIT; k++) |
1574 |
71168 |
q->sb_samples[ch][(8 * index) + i][k] = 0; |
|
1575 |
|||
1576 |
✓✓ | 6672 |
for (ch = 0; ch < q->nb_channels; ch++) { |
1577 |
4448 |
float *samples_ptr = q->samples + ch; |
|
1578 |
|||
1579 |
✓✓ | 40032 |
for (i = 0; i < 8; i++) { |
1580 |
35584 |
ff_mpa_synth_filter_float(&q->mpadsp, |
|
1581 |
35584 |
q->synth_buf[ch], &(q->synth_buf_offset[ch]), |
|
1582 |
ff_mpa_synth_window_float, &dither_state, |
||
1583 |
35584 |
samples_ptr, q->nb_channels, |
|
1584 |
35584 |
q->sb_samples[ch][(8 * index) + i]); |
|
1585 |
35584 |
samples_ptr += 32 * q->nb_channels; |
|
1586 |
} |
||
1587 |
} |
||
1588 |
|||
1589 |
/* add samples to output buffer */ |
||
1590 |
2224 |
sub_sampling = (4 >> q->sub_sampling); |
|
1591 |
|||
1592 |
✓✓ | 6672 |
for (ch = 0; ch < q->channels; ch++) |
1593 |
✓✓ | 1143136 |
for (i = 0; i < q->frame_size; i++) |
1594 |
1138688 |
q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch]; |
|
1595 |
2224 |
} |
|
1596 |
|||
1597 |
/** |
||
1598 |
* Init static data (does not depend on specific file) |
||
1599 |
*/ |
||
1600 |
1 |
static av_cold void qdm2_init_static_data(void) { |
|
1601 |
1 |
qdm2_init_vlc(); |
|
1602 |
1 |
softclip_table_init(); |
|
1603 |
1 |
rnd_table_init(); |
|
1604 |
1 |
init_noise_samples(); |
|
1605 |
|||
1606 |
1 |
ff_mpa_synth_init_float(); |
|
1607 |
1 |
} |
|
1608 |
|||
1609 |
/** |
||
1610 |
* Init parameters from codec extradata |
||
1611 |
*/ |
||
1612 |
2 |
static av_cold int qdm2_decode_init(AVCodecContext *avctx) |
|
1613 |
{ |
||
1614 |
static AVOnce init_static_once = AV_ONCE_INIT; |
||
1615 |
2 |
QDM2Context *s = avctx->priv_data; |
|
1616 |
int tmp_val, tmp, size; |
||
1617 |
GetByteContext gb; |
||
1618 |
|||
1619 |
/* extradata parsing |
||
1620 |
|||
1621 |
Structure: |
||
1622 |
wave { |
||
1623 |
frma (QDM2) |
||
1624 |
QDCA |
||
1625 |
QDCP |
||
1626 |
} |
||
1627 |
|||
1628 |
32 size (including this field) |
||
1629 |
32 tag (=frma) |
||
1630 |
32 type (=QDM2 or QDMC) |
||
1631 |
|||
1632 |
32 size (including this field, in bytes) |
||
1633 |
32 tag (=QDCA) // maybe mandatory parameters |
||
1634 |
32 unknown (=1) |
||
1635 |
32 channels (=2) |
||
1636 |
32 samplerate (=44100) |
||
1637 |
32 bitrate (=96000) |
||
1638 |
32 block size (=4096) |
||
1639 |
32 frame size (=256) (for one channel) |
||
1640 |
32 packet size (=1300) |
||
1641 |
|||
1642 |
32 size (including this field, in bytes) |
||
1643 |
32 tag (=QDCP) // maybe some tuneable parameters |
||
1644 |
32 float1 (=1.0) |
||
1645 |
32 zero ? |
||
1646 |
32 float2 (=1.0) |
||
1647 |
32 float3 (=1.0) |
||
1648 |
32 unknown (27) |
||
1649 |
32 unknown (8) |
||
1650 |
32 zero ? |
||
1651 |
*/ |
||
1652 |
|||
1653 |
✓✗✗✓ |
2 |
if (!avctx->extradata || (avctx->extradata_size < 48)) { |
1654 |
av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n"); |
||
1655 |
return AVERROR_INVALIDDATA; |
||
1656 |
} |
||
1657 |
|||
1658 |
2 |
bytestream2_init(&gb, avctx->extradata, avctx->extradata_size); |
|
1659 |
|||
1660 |
✓✗ | 10 |
while (bytestream2_get_bytes_left(&gb) > 8) { |
1661 |
✓✓ | 10 |
if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) | |
1662 |
(uint64_t)MKBETAG('Q','D','M','2'))) |
||
1663 |
2 |
break; |
|
1664 |
8 |
bytestream2_skip(&gb, 1); |
|
1665 |
} |
||
1666 |
|||
1667 |
✗✓ | 2 |
if (bytestream2_get_bytes_left(&gb) < 12) { |
1668 |
av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n", |
||
1669 |
bytestream2_get_bytes_left(&gb)); |
||
1670 |
return AVERROR_INVALIDDATA; |
||
1671 |
} |
||
1672 |
|||
1673 |
2 |
bytestream2_skip(&gb, 8); |
|
1674 |
2 |
size = bytestream2_get_be32(&gb); |
|
1675 |
|||
1676 |
✗✓ | 2 |
if (size > bytestream2_get_bytes_left(&gb)) { |
1677 |
av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n", |
||
1678 |
bytestream2_get_bytes_left(&gb), size); |
||
1679 |
return AVERROR_INVALIDDATA; |
||
1680 |
} |
||
1681 |
|||
1682 |
2 |
av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size); |
|
1683 |
✗✓ | 2 |
if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) { |
1684 |
av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n"); |
||
1685 |
return AVERROR_INVALIDDATA; |
||
1686 |
} |
||
1687 |
|||
1688 |
2 |
bytestream2_skip(&gb, 4); |
|
1689 |
|||
1690 |
2 |
avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb); |
|
1691 |
✓✗✗✓ |
2 |
if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) { |
1692 |
av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n"); |
||
1693 |
return AVERROR_INVALIDDATA; |
||
1694 |
} |
||
1695 |
✓✗ | 2 |
avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO : |
1696 |
AV_CH_LAYOUT_MONO; |
||
1697 |
|||
1698 |
2 |
avctx->sample_rate = bytestream2_get_be32(&gb); |
|
1699 |
2 |
avctx->bit_rate = bytestream2_get_be32(&gb); |
|
1700 |
2 |
s->group_size = bytestream2_get_be32(&gb); |
|
1701 |
2 |
s->fft_size = bytestream2_get_be32(&gb); |
|
1702 |
2 |
s->checksum_size = bytestream2_get_be32(&gb); |
|
1703 |
✓✗✗✓ |
2 |
if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) { |
1704 |
av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size); |
||
1705 |
return AVERROR_INVALIDDATA; |
||
1706 |
} |
||
1707 |
|||
1708 |
2 |
s->fft_order = av_log2(s->fft_size) + 1; |
|
1709 |
|||
1710 |
// Fail on unknown fft order |
||
1711 |
✓✗✗✓ |
2 |
if ((s->fft_order < 7) || (s->fft_order > 9)) { |
1712 |
avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order); |
||
1713 |
return AVERROR_PATCHWELCOME; |
||
1714 |
} |
||
1715 |
|||
1716 |
// something like max decodable tones |
||
1717 |
2 |
s->group_order = av_log2(s->group_size) + 1; |
|
1718 |
2 |
s->frame_size = s->group_size / 16; // 16 iterations per super block |
|
1719 |
|||
1720 |
✗✓ | 2 |
if (s->frame_size > QDM2_MAX_FRAME_SIZE) |
1721 |
return AVERROR_INVALIDDATA; |
||
1722 |
|||
1723 |
2 |
s->sub_sampling = s->fft_order - 7; |
|
1724 |
2 |
s->frequency_range = 255 / (1 << (2 - s->sub_sampling)); |
|
1725 |
|||
1726 |
✗✓ | 2 |
if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) { |
1727 |
avpriv_request_sample(avctx, "large frames"); |
||
1728 |
return AVERROR_PATCHWELCOME; |
||
1729 |
} |
||
1730 |
|||
1731 |
✗✗✗✗ ✗✓✗ |
2 |
switch ((s->sub_sampling * 2 + s->channels - 1)) { |
1732 |
case 0: tmp = 40; break; |
||
1733 |
case 1: tmp = 48; break; |
||
1734 |
case 2: tmp = 56; break; |
||
1735 |
case 3: tmp = 72; break; |
||
1736 |
case 4: tmp = 80; break; |
||
1737 |
2 |
case 5: tmp = 100;break; |
|
1738 |
default: tmp=s->sub_sampling; break; |
||
1739 |
} |
||
1740 |
2 |
tmp_val = 0; |
|
1741 |
✗✓ | 2 |
if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1; |
1742 |
✗✓ | 2 |
if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2; |
1743 |
✗✓ | 2 |
if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3; |
1744 |
✗✓ | 2 |
if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4; |
1745 |
2 |
s->cm_table_select = tmp_val; |
|
1746 |
|||
1747 |
✗✓ | 2 |
if (avctx->bit_rate <= 8000) |
1748 |
s->coeff_per_sb_select = 0; |
||
1749 |
✗✓ | 2 |
else if (avctx->bit_rate < 16000) |
1750 |
s->coeff_per_sb_select = 1; |
||
1751 |
else |
||
1752 |
2 |
s->coeff_per_sb_select = 2; |
|
1753 |
|||
1754 |
✗✓ | 2 |
if (s->fft_size != (1 << (s->fft_order - 1))) { |
1755 |
av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size); |
||
1756 |
return AVERROR_INVALIDDATA; |
||
1757 |
} |
||
1758 |
|||
1759 |
2 |
ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); |
|
1760 |
2 |
ff_mpadsp_init(&s->mpadsp); |
|
1761 |
|||
1762 |
2 |
avctx->sample_fmt = AV_SAMPLE_FMT_S16; |
|
1763 |
|||
1764 |
2 |
ff_thread_once(&init_static_once, qdm2_init_static_data); |
|
1765 |
|||
1766 |
2 |
return 0; |
|
1767 |
} |
||
1768 |
|||
1769 |
2 |
static av_cold int qdm2_decode_close(AVCodecContext *avctx) |
|
1770 |
{ |
||
1771 |
2 |
QDM2Context *s = avctx->priv_data; |
|
1772 |
|||
1773 |
2 |
ff_rdft_end(&s->rdft_ctx); |
|
1774 |
|||
1775 |
2 |
return 0; |
|
1776 |
} |
||
1777 |
|||
1778 |
2224 |
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out) |
|
1779 |
{ |
||
1780 |
int ch, i; |
||
1781 |
2224 |
const int frame_size = (q->frame_size * q->channels); |
|
1782 |
|||
1783 |
✗✓ | 2224 |
if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2) |
1784 |
return -1; |
||
1785 |
|||
1786 |
/* select input buffer */ |
||
1787 |
2224 |
q->compressed_data = in; |
|
1788 |
2224 |
q->compressed_size = q->checksum_size; |
|
1789 |
|||
1790 |
/* copy old block, clear new block of output samples */ |
||
1791 |
2224 |
memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float)); |
|
1792 |
2224 |
memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float)); |
|
1793 |
|||
1794 |
/* decode block of QDM2 compressed data */ |
||
1795 |
✓✓ | 2224 |
if (q->sub_packet == 0) { |
1796 |
139 |
q->has_errors = 0; // zero it for a new super block |
|
1797 |
139 |
av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n"); |
|
1798 |
139 |
qdm2_decode_super_block(q); |
|
1799 |
} |
||
1800 |
|||
1801 |
/* parse subpackets */ |
||
1802 |
✓✗ | 2224 |
if (!q->has_errors) { |
1803 |
✓✓ | 2224 |
if (q->sub_packet == 2) |
1804 |
139 |
qdm2_decode_fft_packets(q); |
|
1805 |
|||
1806 |
2224 |
qdm2_fft_tone_synthesizer(q, q->sub_packet); |
|
1807 |
} |
||
1808 |
|||
1809 |
/* sound synthesis stage 1 (FFT) */ |
||
1810 |
✓✓ | 6672 |
for (ch = 0; ch < q->channels; ch++) { |
1811 |
4448 |
qdm2_calculate_fft(q, ch, q->sub_packet); |
|
1812 |
|||
1813 |
✓✗✗✓ |
4448 |
if (!q->has_errors && q->sub_packet_list_C[0].packet) { |
1814 |
SAMPLES_NEEDED_2("has errors, and C list is not empty") |
||
1815 |
return -1; |
||
1816 |
} |
||
1817 |
} |
||
1818 |
|||
1819 |
/* sound synthesis stage 2 (MPEG audio like synthesis filter) */ |
||
1820 |
✓✗✓✗ |
2224 |
if (!q->has_errors && q->do_synth_filter) |
1821 |
2224 |
qdm2_synthesis_filter(q, q->sub_packet); |
|
1822 |
|||
1823 |
2224 |
q->sub_packet = (q->sub_packet + 1) % 16; |
|
1824 |
|||
1825 |
/* clip and convert output float[] to 16-bit signed samples */ |
||
1826 |
✓✓ | 1140912 |
for (i = 0; i < frame_size; i++) { |
1827 |
1138688 |
int value = (int)q->output_buffer[i]; |
|
1828 |
|||
1829 |
✓✓ | 1138688 |
if (value > SOFTCLIP_THRESHOLD) |
1830 |
✓✓ | 245 |
value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD]; |
1831 |
✓✓ | 1138443 |
else if (value < -SOFTCLIP_THRESHOLD) |
1832 |
✓✓ | 839 |
value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD]; |
1833 |
|||
1834 |
1138688 |
out[i] = value; |
|
1835 |
} |
||
1836 |
|||
1837 |
2224 |
return 0; |
|
1838 |
} |
||
1839 |
|||
1840 |
139 |
static int qdm2_decode_frame(AVCodecContext *avctx, void *data, |
|
1841 |
int *got_frame_ptr, AVPacket *avpkt) |
||
1842 |
{ |
||
1843 |
139 |
AVFrame *frame = data; |
|
1844 |
139 |
const uint8_t *buf = avpkt->data; |
|
1845 |
139 |
int buf_size = avpkt->size; |
|
1846 |
139 |
QDM2Context *s = avctx->priv_data; |
|
1847 |
int16_t *out; |
||
1848 |
int i, ret; |
||
1849 |
|||
1850 |
✗✓ | 139 |
if(!buf) |
1851 |
return 0; |
||
1852 |
✗✓ | 139 |
if(buf_size < s->checksum_size) |
1853 |
return -1; |
||
1854 |
|||
1855 |
/* get output buffer */ |
||
1856 |
139 |
frame->nb_samples = 16 * s->frame_size; |
|
1857 |
✗✓ | 139 |
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
1858 |
return ret; |
||
1859 |
139 |
out = (int16_t *)frame->data[0]; |
|
1860 |
|||
1861 |
✓✓ | 2363 |
for (i = 0; i < 16; i++) { |
1862 |
✗✓ | 2224 |
if ((ret = qdm2_decode(s, buf, out)) < 0) |
1863 |
return ret; |
||
1864 |
2224 |
out += s->channels * s->frame_size; |
|
1865 |
} |
||
1866 |
|||
1867 |
139 |
*got_frame_ptr = 1; |
|
1868 |
|||
1869 |
139 |
return s->checksum_size; |
|
1870 |
} |
||
1871 |
|||
1872 |
AVCodec ff_qdm2_decoder = { |
||
1873 |
.name = "qdm2", |
||
1874 |
.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"), |
||
1875 |
.type = AVMEDIA_TYPE_AUDIO, |
||
1876 |
.id = AV_CODEC_ID_QDM2, |
||
1877 |
.priv_data_size = sizeof(QDM2Context), |
||
1878 |
.init = qdm2_decode_init, |
||
1879 |
.close = qdm2_decode_close, |
||
1880 |
.decode = qdm2_decode_frame, |
||
1881 |
.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, |
||
1882 |
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE, |
||
1883 |
}; |
Generated by: GCOVR (Version 4.2) |