aboutsummaryrefslogtreecommitdiffstats
path: root/libavcodec/aac/aacdec_latm.h
blob: 047c11e0fb554247b498031847b709dd0f262155 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
/*
 * AAC decoder
 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
 * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
 *
 * AAC LATM decoder
 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
 * Copyright (c) 2010      Janne Grunau <janne-libav@jannau.net>
 *
 * AAC decoder fixed-point implementation
 * Copyright (c) 2013
 *      MIPS Technologies, Inc., California.
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#ifndef AVCODEC_AAC_AACDEC_LATM_H
#define AVCODEC_AAC_AACDEC_LATM_H

#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word

struct LATMContext {
    AACDecContext aac_ctx;  ///< containing AACContext
    int initialized;        ///< initialized after a valid extradata was seen

    // parser data
    int audio_mux_version_A; ///< LATM syntax version
    int frame_length_type;   ///< 0/1 variable/fixed frame length
    int frame_length;        ///< frame length for fixed frame length
};

static inline uint32_t latm_get_value(GetBitContext *b)
{
    int length = get_bits(b, 2);

    return get_bits_long(b, (length+1)*8);
}

static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
                                             GetBitContext *gb, int asclen)
{
    AACDecContext *ac     = &latmctx->aac_ctx;
    AVCodecContext *avctx = ac->avctx;
    OutputConfiguration oc = { 0 };
    MPEG4AudioConfig *m4ac = &oc.m4ac;
    GetBitContext gbc;
    int config_start_bit  = get_bits_count(gb);
    int sync_extension    = 0;
    int bits_consumed, esize, i;

    if (asclen > 0) {
        sync_extension = 1;
        asclen         = FFMIN(asclen, get_bits_left(gb));
        init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
        skip_bits_long(&gbc, config_start_bit);
    } else if (asclen == 0) {
        gbc = *gb;
    } else {
        return AVERROR_INVALIDDATA;
    }

    if (get_bits_left(gb) <= 0)
        return AVERROR_INVALIDDATA;

    bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &oc,
                                                    &gbc, config_start_bit,
                                                    sync_extension);

    if (bits_consumed < config_start_bit)
        return AVERROR_INVALIDDATA;
    bits_consumed -= config_start_bit;

    if (asclen == 0)
      asclen = bits_consumed;

    if (!latmctx->initialized ||
        ac->oc[1].m4ac.sample_rate != m4ac->sample_rate ||
        ac->oc[1].m4ac.chan_config != m4ac->chan_config) {

        if (latmctx->initialized) {
            av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n",
                   m4ac->sample_rate, m4ac->chan_config);
        } else {
            av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
        }
        latmctx->initialized = 0;

        esize = (asclen + 7) / 8;

        if (avctx->extradata_size < esize) {
            av_free(avctx->extradata);
            avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
            if (!avctx->extradata)
                return AVERROR(ENOMEM);
        }

        avctx->extradata_size = esize;
        gbc = *gb;
        for (i = 0; i < esize; i++) {
          avctx->extradata[i] = get_bits(&gbc, 8);
        }
        memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
    }
    skip_bits_long(gb, asclen);

    return 0;
}

static int read_stream_mux_config(struct LATMContext *latmctx,
                                  GetBitContext *gb)
{
    int ret, audio_mux_version = get_bits(gb, 1);

    latmctx->audio_mux_version_A = 0;
    if (audio_mux_version)
        latmctx->audio_mux_version_A = get_bits(gb, 1);

    if (!latmctx->audio_mux_version_A) {

        if (audio_mux_version)
            latm_get_value(gb);                 // taraFullness

        skip_bits(gb, 1);                       // allStreamSameTimeFraming
        skip_bits(gb, 6);                       // numSubFrames
        // numPrograms
        if (get_bits(gb, 4)) {                  // numPrograms
            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
            return AVERROR_PATCHWELCOME;
        }

        // for each program (which there is only one in DVB)

        // for each layer (which there is only one in DVB)
        if (get_bits(gb, 3)) {                   // numLayer
            avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
            return AVERROR_PATCHWELCOME;
        }

        // for all but first stream: use_same_config = get_bits(gb, 1);
        if (!audio_mux_version) {
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
                return ret;
        } else {
            int ascLen = latm_get_value(gb);
            if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
                return ret;
        }

        latmctx->frame_length_type = get_bits(gb, 3);
        switch (latmctx->frame_length_type) {
        case 0:
            skip_bits(gb, 8);       // latmBufferFullness
            break;
        case 1:
            latmctx->frame_length = get_bits(gb, 9);
            break;
        case 3:
        case 4:
        case 5:
            skip_bits(gb, 6);       // CELP frame length table index
            break;
        case 6:
        case 7:
            skip_bits(gb, 1);       // HVXC frame length table index
            break;
        }

        if (get_bits(gb, 1)) {                  // other data
            if (audio_mux_version) {
                latm_get_value(gb);             // other_data_bits
            } else {
                int esc;
                do {
                    if (get_bits_left(gb) < 9)
                        return AVERROR_INVALIDDATA;
                    esc = get_bits(gb, 1);
                    skip_bits(gb, 8);
                } while (esc);
            }
        }

        if (get_bits(gb, 1))                     // crc present
            skip_bits(gb, 8);                    // config_crc
    }

    return 0;
}

static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
{
    uint8_t tmp;

    if (ctx->frame_length_type == 0) {
        int mux_slot_length = 0;
        do {
            if (get_bits_left(gb) < 8)
                return AVERROR_INVALIDDATA;
            tmp = get_bits(gb, 8);
            mux_slot_length += tmp;
        } while (tmp == 255);
        return mux_slot_length;
    } else if (ctx->frame_length_type == 1) {
        return ctx->frame_length;
    } else if (ctx->frame_length_type == 3 ||
               ctx->frame_length_type == 5 ||
               ctx->frame_length_type == 7) {
        skip_bits(gb, 2);          // mux_slot_length_coded
    }
    return 0;
}

static int read_audio_mux_element(struct LATMContext *latmctx,
                                  GetBitContext *gb)
{
    int err;
    uint8_t use_same_mux = get_bits(gb, 1);
    if (!use_same_mux) {
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
            return err;
    } else if (!latmctx->aac_ctx.avctx->extradata) {
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
               "no decoder config found\n");
        return 1;
    }
    if (latmctx->audio_mux_version_A == 0) {
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
        if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
            return AVERROR_INVALIDDATA;
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
                   "frame length mismatch %d << %d\n",
                   mux_slot_length_bytes * 8, get_bits_left(gb));
            return AVERROR_INVALIDDATA;
        }
    }
    return 0;
}


static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out,
                             int *got_frame_ptr, AVPacket *avpkt)
{
    struct LATMContext *latmctx = avctx->priv_data;
    int                 muxlength, err;
    GetBitContext       gb;

    if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
        return err;

    // check for LOAS sync word
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
        return AVERROR_INVALIDDATA;

    muxlength = get_bits(&gb, 13) + 3;
    // not enough data, the parser should have sorted this out
    if (muxlength > avpkt->size)
        return AVERROR_INVALIDDATA;

    if ((err = read_audio_mux_element(latmctx, &gb)))
        return (err < 0) ? err : avpkt->size;

    if (!latmctx->initialized) {
        if (!avctx->extradata) {
            *got_frame_ptr = 0;
            return avpkt->size;
        } else {
            push_output_configuration(&latmctx->aac_ctx);
            if ((err = decode_audio_specific_config(
                    &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1],
                    avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
                pop_output_configuration(&latmctx->aac_ctx);
                return err;
            }
            latmctx->initialized = 1;
        }
    }

    if (show_bits(&gb, 12) == 0xfff) {
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
               "ADTS header detected, probably as result of configuration "
               "misparsing\n");
        return AVERROR_INVALIDDATA;
    }

    switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
    case AOT_ER_AAC_LC:
    case AOT_ER_AAC_LTP:
    case AOT_ER_AAC_LD:
    case AOT_ER_AAC_ELD:
        err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
        break;
    default:
        err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
    }
    if (err < 0)
        return err;

    return muxlength;
}

static av_cold int latm_decode_init(AVCodecContext *avctx)
{
    struct LATMContext *latmctx = avctx->priv_data;
    int ret = ff_aac_decode_init_float(avctx);

    if (avctx->extradata_size > 0)
        latmctx->initialized = !ret;

    return ret;
}

/*
    Note: This decoder filter is intended to decode LATM streams transferred
    in MPEG transport streams which only contain one program.
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
*/
const FFCodec ff_aac_latm_decoder = {
    .p.name          = "aac_latm",
    CODEC_LONG_NAME("AAC LATM (Advanced Audio Coding LATM syntax)"),
    .p.type          = AVMEDIA_TYPE_AUDIO,
    .p.id            = AV_CODEC_ID_AAC_LATM,
    .priv_data_size  = sizeof(struct LATMContext),
    .init            = latm_decode_init,
    .close           = decode_close,
    FF_CODEC_DECODE_CB(latm_decode_frame),
    .p.sample_fmts   = (const enum AVSampleFormat[]) {
        AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
    },
    .p.capabilities  = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
    .caps_internal   = FF_CODEC_CAP_INIT_CLEANUP,
    .p.ch_layouts    = ff_aac_ch_layout,
    .flush = flush,
    .p.profiles      = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
};

#endif /* AVCODEC_AAC_AACDEC_LATM_H */