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1 | /* | ||
2 | * AAC decoder | ||
3 | * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) | ||
4 | * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) | ||
5 | * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com> | ||
6 | * | ||
7 | * AAC LATM decoder | ||
8 | * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz> | ||
9 | * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net> | ||
10 | * | ||
11 | * This file is part of FFmpeg. | ||
12 | * | ||
13 | * FFmpeg is free software; you can redistribute it and/or | ||
14 | * modify it under the terms of the GNU Lesser General Public | ||
15 | * License as published by the Free Software Foundation; either | ||
16 | * version 2.1 of the License, or (at your option) any later version. | ||
17 | * | ||
18 | * FFmpeg is distributed in the hope that it will be useful, | ||
19 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
20 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
21 | * Lesser General Public License for more details. | ||
22 | * | ||
23 | * You should have received a copy of the GNU Lesser General Public | ||
24 | * License along with FFmpeg; if not, write to the Free Software | ||
25 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
26 | */ | ||
27 | |||
28 | /** | ||
29 | * @file | ||
30 | * AAC decoder | ||
31 | * @author Oded Shimon ( ods15 ods15 dyndns org ) | ||
32 | * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) | ||
33 | */ | ||
34 | |||
35 | #define USE_FIXED 0 | ||
36 | #define TX_TYPE AV_TX_FLOAT_MDCT | ||
37 | |||
38 | #include "libavutil/float_dsp.h" | ||
39 | #include "libavutil/opt.h" | ||
40 | #include "avcodec.h" | ||
41 | #include "codec_internal.h" | ||
42 | #include "get_bits.h" | ||
43 | #include "lpc.h" | ||
44 | #include "kbdwin.h" | ||
45 | #include "sinewin.h" | ||
46 | |||
47 | #include "aac.h" | ||
48 | #include "aactab.h" | ||
49 | #include "aacdectab.h" | ||
50 | #include "adts_header.h" | ||
51 | #include "cbrt_data.h" | ||
52 | #include "sbr.h" | ||
53 | #include "aacsbr.h" | ||
54 | #include "mpeg4audio.h" | ||
55 | #include "profiles.h" | ||
56 | #include "libavutil/intfloat.h" | ||
57 | |||
58 | #include <errno.h> | ||
59 | #include <math.h> | ||
60 | #include <stdint.h> | ||
61 | #include <string.h> | ||
62 | |||
63 | #if ARCH_ARM | ||
64 | # include "arm/aac.h" | ||
65 | #elif ARCH_MIPS | ||
66 | # include "mips/aacdec_mips.h" | ||
67 | #endif | ||
68 | |||
69 | DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_120))[120]; | ||
70 | DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(sine_960))[960]; | ||
71 | DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_long_960))[960]; | ||
72 | DECLARE_ALIGNED(32, static INTFLOAT, AAC_RENAME(aac_kbd_short_120))[120]; | ||
73 | |||
74 | 38542 | static av_always_inline void reset_predict_state(PredictorState *ps) | |
75 | { | ||
76 | 38542 | ps->r0 = 0.0f; | |
77 | 38542 | ps->r1 = 0.0f; | |
78 | 38542 | ps->cor0 = 0.0f; | |
79 | 38542 | ps->cor1 = 0.0f; | |
80 | 38542 | ps->var0 = 1.0f; | |
81 | 38542 | ps->var1 = 1.0f; | |
82 | 38542 | } | |
83 | |||
84 | #ifndef VMUL2 | ||
85 | 1815116 | static inline float *VMUL2(float *dst, const float *v, unsigned idx, | |
86 | const float *scale) | ||
87 | { | ||
88 | 1815116 | float s = *scale; | |
89 | 1815116 | *dst++ = v[idx & 15] * s; | |
90 | 1815116 | *dst++ = v[idx>>4 & 15] * s; | |
91 | 1815116 | return dst; | |
92 | } | ||
93 | #endif | ||
94 | |||
95 | #ifndef VMUL4 | ||
96 | 2273024 | static inline float *VMUL4(float *dst, const float *v, unsigned idx, | |
97 | const float *scale) | ||
98 | { | ||
99 | 2273024 | float s = *scale; | |
100 | 2273024 | *dst++ = v[idx & 3] * s; | |
101 | 2273024 | *dst++ = v[idx>>2 & 3] * s; | |
102 | 2273024 | *dst++ = v[idx>>4 & 3] * s; | |
103 | 2273024 | *dst++ = v[idx>>6 & 3] * s; | |
104 | 2273024 | return dst; | |
105 | } | ||
106 | #endif | ||
107 | |||
108 | #ifndef VMUL2S | ||
109 | 2274420 | static inline float *VMUL2S(float *dst, const float *v, unsigned idx, | |
110 | unsigned sign, const float *scale) | ||
111 | { | ||
112 | union av_intfloat32 s0, s1; | ||
113 | |||
114 | 2274420 | s0.f = s1.f = *scale; | |
115 | 2274420 | s0.i ^= sign >> 1 << 31; | |
116 | 2274420 | s1.i ^= sign << 31; | |
117 | |||
118 | 2274420 | *dst++ = v[idx & 15] * s0.f; | |
119 | 2274420 | *dst++ = v[idx>>4 & 15] * s1.f; | |
120 | |||
121 | 2274420 | return dst; | |
122 | } | ||
123 | #endif | ||
124 | |||
125 | #ifndef VMUL4S | ||
126 | 1809834 | static inline float *VMUL4S(float *dst, const float *v, unsigned idx, | |
127 | unsigned sign, const float *scale) | ||
128 | { | ||
129 | 1809834 | unsigned nz = idx >> 12; | |
130 | 1809834 | union av_intfloat32 s = { .f = *scale }; | |
131 | union av_intfloat32 t; | ||
132 | |||
133 | 1809834 | t.i = s.i ^ (sign & 1U<<31); | |
134 | 1809834 | *dst++ = v[idx & 3] * t.f; | |
135 | |||
136 | 1809834 | sign <<= nz & 1; nz >>= 1; | |
137 | 1809834 | t.i = s.i ^ (sign & 1U<<31); | |
138 | 1809834 | *dst++ = v[idx>>2 & 3] * t.f; | |
139 | |||
140 | 1809834 | sign <<= nz & 1; nz >>= 1; | |
141 | 1809834 | t.i = s.i ^ (sign & 1U<<31); | |
142 | 1809834 | *dst++ = v[idx>>4 & 3] * t.f; | |
143 | |||
144 | 1809834 | sign <<= nz & 1; | |
145 | 1809834 | t.i = s.i ^ (sign & 1U<<31); | |
146 | 1809834 | *dst++ = v[idx>>6 & 3] * t.f; | |
147 | |||
148 | 1809834 | return dst; | |
149 | } | ||
150 | #endif | ||
151 | |||
152 | 4047936 | static av_always_inline float flt16_round(float pf) | |
153 | { | ||
154 | union av_intfloat32 tmp; | ||
155 | 4047936 | tmp.f = pf; | |
156 | 4047936 | tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; | |
157 | 4047936 | return tmp.f; | |
158 | } | ||
159 | |||
160 | 5085872 | static av_always_inline float flt16_even(float pf) | |
161 | { | ||
162 | union av_intfloat32 tmp; | ||
163 | 5085872 | tmp.f = pf; | |
164 | 5085872 | tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; | |
165 | 5085872 | return tmp.f; | |
166 | } | ||
167 | |||
168 | 24287616 | static av_always_inline float flt16_trunc(float pf) | |
169 | { | ||
170 | union av_intfloat32 pun; | ||
171 | 24287616 | pun.f = pf; | |
172 | 24287616 | pun.i &= 0xFFFF0000U; | |
173 | 24287616 | return pun.f; | |
174 | } | ||
175 | |||
176 | 4047936 | static av_always_inline void predict(PredictorState *ps, float *coef, | |
177 | int output_enable) | ||
178 | { | ||
179 | 4047936 | const float a = 0.953125; // 61.0 / 64 | |
180 | 4047936 | const float alpha = 0.90625; // 29.0 / 32 | |
181 | float e0, e1; | ||
182 | float pv; | ||
183 | float k1, k2; | ||
184 | 4047936 | float r0 = ps->r0, r1 = ps->r1; | |
185 | 4047936 | float cor0 = ps->cor0, cor1 = ps->cor1; | |
186 | 4047936 | float var0 = ps->var0, var1 = ps->var1; | |
187 | |||
188 |
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4047936 | k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0; |
189 |
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4047936 | k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0; |
190 | |||
191 | 4047936 | pv = flt16_round(k1 * r0 + k2 * r1); | |
192 |
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4047936 | if (output_enable) |
193 | 271896 | *coef += pv; | |
194 | |||
195 | 4047936 | e0 = *coef; | |
196 | 4047936 | e1 = e0 - k1 * r0; | |
197 | |||
198 | 4047936 | ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1); | |
199 | 4047936 | ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1)); | |
200 | 4047936 | ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0); | |
201 | 4047936 | ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0)); | |
202 | |||
203 | 4047936 | ps->r1 = flt16_trunc(a * (r0 - k1 * e0)); | |
204 | 4047936 | ps->r0 = flt16_trunc(a * e0); | |
205 | 4047936 | } | |
206 | |||
207 | /** | ||
208 | * Apply dependent channel coupling (applied before IMDCT). | ||
209 | * | ||
210 | * @param index index into coupling gain array | ||
211 | */ | ||
212 | 2743 | static void apply_dependent_coupling(AACContext *ac, | |
213 | SingleChannelElement *target, | ||
214 | ChannelElement *cce, int index) | ||
215 | { | ||
216 | 2743 | IndividualChannelStream *ics = &cce->ch[0].ics; | |
217 | 2743 | const uint16_t *offsets = ics->swb_offset; | |
218 | 2743 | float *dest = target->coeffs; | |
219 | 2743 | const float *src = cce->ch[0].coeffs; | |
220 | 2743 | int g, i, group, k, idx = 0; | |
221 |
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2743 | if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) { |
222 | ✗ | av_log(ac->avctx, AV_LOG_ERROR, | |
223 | "Dependent coupling is not supported together with LTP\n"); | ||
224 | ✗ | return; | |
225 | } | ||
226 |
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5486 | for (g = 0; g < ics->num_window_groups; g++) { |
227 |
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112463 | for (i = 0; i < ics->max_sfb; i++, idx++) { |
228 |
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109720 | if (cce->ch[0].band_type[idx] != ZERO_BT) { |
229 | 5196 | const float gain = cce->coup.gain[index][idx]; | |
230 |
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10392 | for (group = 0; group < ics->group_len[g]; group++) { |
231 |
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81964 | for (k = offsets[i]; k < offsets[i + 1]; k++) { |
232 | // FIXME: SIMDify | ||
233 | 76768 | dest[group * 128 + k] += gain * src[group * 128 + k]; | |
234 | } | ||
235 | } | ||
236 | } | ||
237 | } | ||
238 | 2743 | dest += ics->group_len[g] * 128; | |
239 | 2743 | src += ics->group_len[g] * 128; | |
240 | } | ||
241 | } | ||
242 | |||
243 | /** | ||
244 | * Apply independent channel coupling (applied after IMDCT). | ||
245 | * | ||
246 | * @param index index into coupling gain array | ||
247 | */ | ||
248 | 1315 | static void apply_independent_coupling(AACContext *ac, | |
249 | SingleChannelElement *target, | ||
250 | ChannelElement *cce, int index) | ||
251 | { | ||
252 | 1315 | const float gain = cce->coup.gain[index][0]; | |
253 | 1315 | const float *src = cce->ch[0].ret; | |
254 | 1315 | float *dest = target->ret; | |
255 |
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1315 | const int len = 1024 << (ac->oc[1].m4ac.sbr == 1); |
256 | |||
257 | 1315 | ac->fdsp->vector_fmac_scalar(dest, src, gain, len); | |
258 | 1315 | } | |
259 | |||
260 | #include "aacdec_template.c" | ||
261 | |||
262 | #define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word | ||
263 | |||
264 | struct LATMContext { | ||
265 | AACContext aac_ctx; ///< containing AACContext | ||
266 | int initialized; ///< initialized after a valid extradata was seen | ||
267 | |||
268 | // parser data | ||
269 | int audio_mux_version_A; ///< LATM syntax version | ||
270 | int frame_length_type; ///< 0/1 variable/fixed frame length | ||
271 | int frame_length; ///< frame length for fixed frame length | ||
272 | }; | ||
273 | |||
274 | ✗ | static inline uint32_t latm_get_value(GetBitContext *b) | |
275 | { | ||
276 | ✗ | int length = get_bits(b, 2); | |
277 | |||
278 | ✗ | return get_bits_long(b, (length+1)*8); | |
279 | } | ||
280 | |||
281 | 322 | static int latm_decode_audio_specific_config(struct LATMContext *latmctx, | |
282 | GetBitContext *gb, int asclen) | ||
283 | { | ||
284 | 322 | AACContext *ac = &latmctx->aac_ctx; | |
285 | 322 | AVCodecContext *avctx = ac->avctx; | |
286 | 322 | MPEG4AudioConfig m4ac = { 0 }; | |
287 | GetBitContext gbc; | ||
288 | 322 | int config_start_bit = get_bits_count(gb); | |
289 | 322 | int sync_extension = 0; | |
290 | int bits_consumed, esize, i; | ||
291 | |||
292 |
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322 | if (asclen > 0) { |
293 | ✗ | sync_extension = 1; | |
294 | ✗ | asclen = FFMIN(asclen, get_bits_left(gb)); | |
295 | ✗ | init_get_bits(&gbc, gb->buffer, config_start_bit + asclen); | |
296 | ✗ | skip_bits_long(&gbc, config_start_bit); | |
297 |
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322 | } else if (asclen == 0) { |
298 | 322 | gbc = *gb; | |
299 | } else { | ||
300 | ✗ | return AVERROR_INVALIDDATA; | |
301 | } | ||
302 | |||
303 |
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322 | if (get_bits_left(gb) <= 0) |
304 | ✗ | return AVERROR_INVALIDDATA; | |
305 | |||
306 | 322 | bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac, | |
307 | &gbc, config_start_bit, | ||
308 | sync_extension); | ||
309 | |||
310 |
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322 | if (bits_consumed < config_start_bit) |
311 | ✗ | return AVERROR_INVALIDDATA; | |
312 | 322 | bits_consumed -= config_start_bit; | |
313 | |||
314 |
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322 | if (asclen == 0) |
315 | 322 | asclen = bits_consumed; | |
316 | |||
317 |
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322 | if (!latmctx->initialized || |
318 |
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316 | ac->oc[1].m4ac.sample_rate != m4ac.sample_rate || |
319 |
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316 | ac->oc[1].m4ac.chan_config != m4ac.chan_config) { |
320 | |||
321 |
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7 | if (latmctx->initialized) { |
322 | 1 | av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config); | |
323 | } else { | ||
324 | 6 | av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n"); | |
325 | } | ||
326 | 7 | latmctx->initialized = 0; | |
327 | |||
328 | 7 | esize = (asclen + 7) / 8; | |
329 | |||
330 |
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7 | if (avctx->extradata_size < esize) { |
331 | 6 | av_free(avctx->extradata); | |
332 | 6 | avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE); | |
333 |
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6 | if (!avctx->extradata) |
334 | ✗ | return AVERROR(ENOMEM); | |
335 | } | ||
336 | |||
337 | 7 | avctx->extradata_size = esize; | |
338 | 7 | gbc = *gb; | |
339 |
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28 | for (i = 0; i < esize; i++) { |
340 | 21 | avctx->extradata[i] = get_bits(&gbc, 8); | |
341 | } | ||
342 | 7 | memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE); | |
343 | } | ||
344 | 322 | skip_bits_long(gb, asclen); | |
345 | |||
346 | 322 | return 0; | |
347 | } | ||
348 | |||
349 | 322 | static int read_stream_mux_config(struct LATMContext *latmctx, | |
350 | GetBitContext *gb) | ||
351 | { | ||
352 | 322 | int ret, audio_mux_version = get_bits(gb, 1); | |
353 | |||
354 | 322 | latmctx->audio_mux_version_A = 0; | |
355 |
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322 | if (audio_mux_version) |
356 | ✗ | latmctx->audio_mux_version_A = get_bits(gb, 1); | |
357 | |||
358 |
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322 | if (!latmctx->audio_mux_version_A) { |
359 | |||
360 |
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322 | if (audio_mux_version) |
361 | ✗ | latm_get_value(gb); // taraFullness | |
362 | |||
363 | 322 | skip_bits(gb, 1); // allStreamSameTimeFraming | |
364 | 322 | skip_bits(gb, 6); // numSubFrames | |
365 | // numPrograms | ||
366 |
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322 | if (get_bits(gb, 4)) { // numPrograms |
367 | ✗ | avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs"); | |
368 | ✗ | return AVERROR_PATCHWELCOME; | |
369 | } | ||
370 | |||
371 | // for each program (which there is only one in DVB) | ||
372 | |||
373 | // for each layer (which there is only one in DVB) | ||
374 |
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322 | if (get_bits(gb, 3)) { // numLayer |
375 | ✗ | avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers"); | |
376 | ✗ | return AVERROR_PATCHWELCOME; | |
377 | } | ||
378 | |||
379 | // for all but first stream: use_same_config = get_bits(gb, 1); | ||
380 |
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322 | if (!audio_mux_version) { |
381 |
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322 | if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0) |
382 | ✗ | return ret; | |
383 | } else { | ||
384 | ✗ | int ascLen = latm_get_value(gb); | |
385 | ✗ | if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0) | |
386 | ✗ | return ret; | |
387 | } | ||
388 | |||
389 | 322 | latmctx->frame_length_type = get_bits(gb, 3); | |
390 |
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322 | switch (latmctx->frame_length_type) { |
391 | 322 | case 0: | |
392 | 322 | skip_bits(gb, 8); // latmBufferFullness | |
393 | 322 | break; | |
394 | ✗ | case 1: | |
395 | ✗ | latmctx->frame_length = get_bits(gb, 9); | |
396 | ✗ | break; | |
397 | ✗ | case 3: | |
398 | case 4: | ||
399 | case 5: | ||
400 | ✗ | skip_bits(gb, 6); // CELP frame length table index | |
401 | ✗ | break; | |
402 | ✗ | case 6: | |
403 | case 7: | ||
404 | ✗ | skip_bits(gb, 1); // HVXC frame length table index | |
405 | ✗ | break; | |
406 | } | ||
407 | |||
408 |
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322 | if (get_bits(gb, 1)) { // other data |
409 | ✗ | if (audio_mux_version) { | |
410 | ✗ | latm_get_value(gb); // other_data_bits | |
411 | } else { | ||
412 | int esc; | ||
413 | do { | ||
414 | ✗ | if (get_bits_left(gb) < 9) | |
415 | ✗ | return AVERROR_INVALIDDATA; | |
416 | ✗ | esc = get_bits(gb, 1); | |
417 | ✗ | skip_bits(gb, 8); | |
418 | ✗ | } while (esc); | |
419 | } | ||
420 | } | ||
421 | |||
422 |
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322 | if (get_bits(gb, 1)) // crc present |
423 | ✗ | skip_bits(gb, 8); // config_crc | |
424 | } | ||
425 | |||
426 | 322 | return 0; | |
427 | } | ||
428 | |||
429 | 515 | static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb) | |
430 | { | ||
431 | uint8_t tmp; | ||
432 | |||
433 |
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515 | if (ctx->frame_length_type == 0) { |
434 | 515 | int mux_slot_length = 0; | |
435 | do { | ||
436 |
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1504 | if (get_bits_left(gb) < 8) |
437 | ✗ | return AVERROR_INVALIDDATA; | |
438 | 1504 | tmp = get_bits(gb, 8); | |
439 | 1504 | mux_slot_length += tmp; | |
440 |
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1504 | } while (tmp == 255); |
441 | 515 | return mux_slot_length; | |
442 | ✗ | } else if (ctx->frame_length_type == 1) { | |
443 | ✗ | return ctx->frame_length; | |
444 | ✗ | } else if (ctx->frame_length_type == 3 || | |
445 | ✗ | ctx->frame_length_type == 5 || | |
446 | ✗ | ctx->frame_length_type == 7) { | |
447 | ✗ | skip_bits(gb, 2); // mux_slot_length_coded | |
448 | } | ||
449 | ✗ | return 0; | |
450 | } | ||
451 | |||
452 | 521 | static int read_audio_mux_element(struct LATMContext *latmctx, | |
453 | GetBitContext *gb) | ||
454 | { | ||
455 | int err; | ||
456 | 521 | uint8_t use_same_mux = get_bits(gb, 1); | |
457 |
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521 | if (!use_same_mux) { |
458 |
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322 | if ((err = read_stream_mux_config(latmctx, gb)) < 0) |
459 | ✗ | return err; | |
460 |
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199 | } else if (!latmctx->aac_ctx.avctx->extradata) { |
461 | 6 | av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG, | |
462 | "no decoder config found\n"); | ||
463 | 6 | return 1; | |
464 | } | ||
465 |
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515 | if (latmctx->audio_mux_version_A == 0) { |
466 | 515 | int mux_slot_length_bytes = read_payload_length_info(latmctx, gb); | |
467 |
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515 | if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) { |
468 | ✗ | av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n"); | |
469 | ✗ | return AVERROR_INVALIDDATA; | |
470 |
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515 | } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) { |
471 | ✗ | av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, | |
472 | "frame length mismatch %d << %d\n", | ||
473 | mux_slot_length_bytes * 8, get_bits_left(gb)); | ||
474 | ✗ | return AVERROR_INVALIDDATA; | |
475 | } | ||
476 | } | ||
477 | 515 | return 0; | |
478 | } | ||
479 | |||
480 | |||
481 | 523 | static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out, | |
482 | int *got_frame_ptr, AVPacket *avpkt) | ||
483 | { | ||
484 | 523 | struct LATMContext *latmctx = avctx->priv_data; | |
485 | int muxlength, err; | ||
486 | GetBitContext gb; | ||
487 | |||
488 |
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523 | if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0) |
489 | ✗ | return err; | |
490 | |||
491 | // check for LOAS sync word | ||
492 |
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523 | if (get_bits(&gb, 11) != LOAS_SYNC_WORD) |
493 | ✗ | return AVERROR_INVALIDDATA; | |
494 | |||
495 | 523 | muxlength = get_bits(&gb, 13) + 3; | |
496 | // not enough data, the parser should have sorted this out | ||
497 |
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523 | if (muxlength > avpkt->size) |
498 | 2 | return AVERROR_INVALIDDATA; | |
499 | |||
500 |
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521 | if ((err = read_audio_mux_element(latmctx, &gb))) |
501 |
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6 | return (err < 0) ? err : avpkt->size; |
502 | |||
503 |
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515 | if (!latmctx->initialized) { |
504 |
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7 | if (!avctx->extradata) { |
505 | ✗ | *got_frame_ptr = 0; | |
506 | ✗ | return avpkt->size; | |
507 | } else { | ||
508 | 7 | push_output_configuration(&latmctx->aac_ctx); | |
509 |
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7 | if ((err = decode_audio_specific_config( |
510 | &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac, | ||
511 | 7 | avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) { | |
512 | ✗ | pop_output_configuration(&latmctx->aac_ctx); | |
513 | ✗ | return err; | |
514 | } | ||
515 | 7 | latmctx->initialized = 1; | |
516 | } | ||
517 | } | ||
518 | |||
519 |
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515 | if (show_bits(&gb, 12) == 0xfff) { |
520 | ✗ | av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, | |
521 | "ADTS header detected, probably as result of configuration " | ||
522 | "misparsing\n"); | ||
523 | ✗ | return AVERROR_INVALIDDATA; | |
524 | } | ||
525 | |||
526 |
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515 | switch (latmctx->aac_ctx.oc[1].m4ac.object_type) { |
527 | ✗ | case AOT_ER_AAC_LC: | |
528 | case AOT_ER_AAC_LTP: | ||
529 | case AOT_ER_AAC_LD: | ||
530 | case AOT_ER_AAC_ELD: | ||
531 | ✗ | err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb); | |
532 | ✗ | break; | |
533 | 515 | default: | |
534 | 515 | err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt); | |
535 | } | ||
536 |
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515 | if (err < 0) |
537 | ✗ | return err; | |
538 | |||
539 | 515 | return muxlength; | |
540 | } | ||
541 | |||
542 | 11 | static av_cold int latm_decode_init(AVCodecContext *avctx) | |
543 | { | ||
544 | 11 | struct LATMContext *latmctx = avctx->priv_data; | |
545 | 11 | int ret = aac_decode_init(avctx); | |
546 | |||
547 |
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11 | if (avctx->extradata_size > 0) |
548 | 5 | latmctx->initialized = !ret; | |
549 | |||
550 | 11 | return ret; | |
551 | } | ||
552 | |||
553 | const FFCodec ff_aac_decoder = { | ||
554 | .p.name = "aac", | ||
555 | CODEC_LONG_NAME("AAC (Advanced Audio Coding)"), | ||
556 | .p.type = AVMEDIA_TYPE_AUDIO, | ||
557 | .p.id = AV_CODEC_ID_AAC, | ||
558 | .priv_data_size = sizeof(AACContext), | ||
559 | .init = aac_decode_init, | ||
560 | .close = aac_decode_close, | ||
561 | FF_CODEC_DECODE_CB(aac_decode_frame), | ||
562 | .p.sample_fmts = (const enum AVSampleFormat[]) { | ||
563 | AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE | ||
564 | }, | ||
565 | .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, | ||
566 | .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, | ||
567 | CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(aac_channel_layout) | ||
568 | .p.ch_layouts = aac_ch_layout, | ||
569 | .flush = flush, | ||
570 | .p.priv_class = &aac_decoder_class, | ||
571 | .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), | ||
572 | }; | ||
573 | |||
574 | /* | ||
575 | Note: This decoder filter is intended to decode LATM streams transferred | ||
576 | in MPEG transport streams which only contain one program. | ||
577 | To do a more complex LATM demuxing a separate LATM demuxer should be used. | ||
578 | */ | ||
579 | const FFCodec ff_aac_latm_decoder = { | ||
580 | .p.name = "aac_latm", | ||
581 | CODEC_LONG_NAME("AAC LATM (Advanced Audio Coding LATM syntax)"), | ||
582 | .p.type = AVMEDIA_TYPE_AUDIO, | ||
583 | .p.id = AV_CODEC_ID_AAC_LATM, | ||
584 | .priv_data_size = sizeof(struct LATMContext), | ||
585 | .init = latm_decode_init, | ||
586 | .close = aac_decode_close, | ||
587 | FF_CODEC_DECODE_CB(latm_decode_frame), | ||
588 | .p.sample_fmts = (const enum AVSampleFormat[]) { | ||
589 | AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE | ||
590 | }, | ||
591 | .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1, | ||
592 | .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, | ||
593 | CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(aac_channel_layout) | ||
594 | .p.ch_layouts = aac_ch_layout, | ||
595 | .flush = flush, | ||
596 | .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles), | ||
597 | }; | ||
598 |