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