summaryrefslogtreecommitdiffstats
path: root/libavcodec/flac.c
blob: 5ffbf93190e2b4e270a5dfd5f840b7f6d95fddcb (plain)
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
/*
 * FLAC common code
 * Copyright (c) 2009 Justin Ruggles
 *
 * 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
 */

#include "libavutil/channel_layout.h"
#include "libavutil/crc.h"
#include "libavutil/log.h"
#include "bytestream.h"
#include "get_bits.h"
#include "flac.h"
#include "flacdata.h"

static const int8_t sample_size_table[] = { 0, 8, 12, 0, 16, 20, 24, 0 };

static const uint64_t flac_channel_layouts[8] = {
    AV_CH_LAYOUT_MONO,
    AV_CH_LAYOUT_STEREO,
    AV_CH_LAYOUT_SURROUND,
    AV_CH_LAYOUT_QUAD,
    AV_CH_LAYOUT_5POINT0,
    AV_CH_LAYOUT_5POINT1,
    AV_CH_LAYOUT_6POINT1,
    AV_CH_LAYOUT_7POINT1
};

static int64_t get_utf8(GetBitContext *gb)
{
    int64_t val;
    GET_UTF8(val, get_bits(gb, 8), return -1;)
    return val;
}

int ff_flac_decode_frame_header(AVCodecContext *avctx, GetBitContext *gb,
                                FLACFrameInfo *fi, int log_level_offset)
{
    int bs_code, sr_code, bps_code;

    /* frame sync code */
    if ((get_bits(gb, 15) & 0x7FFF) != 0x7FFC) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset, "invalid sync code\n");
        return AVERROR_INVALIDDATA;
    }

    /* variable block size stream code */
    fi->is_var_size = get_bits1(gb);

    /* block size and sample rate codes */
    bs_code = get_bits(gb, 4);
    sr_code = get_bits(gb, 4);

    /* channels and decorrelation */
    fi->ch_mode = get_bits(gb, 4);
    if (fi->ch_mode < FLAC_MAX_CHANNELS) {
        fi->channels = fi->ch_mode + 1;
        fi->ch_mode = FLAC_CHMODE_INDEPENDENT;
    } else if (fi->ch_mode < FLAC_MAX_CHANNELS + FLAC_CHMODE_MID_SIDE) {
        fi->channels = 2;
        fi->ch_mode -= FLAC_MAX_CHANNELS - 1;
    } else {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "invalid channel mode: %d\n", fi->ch_mode);
        return AVERROR_INVALIDDATA;
    }

    /* bits per sample */
    bps_code = get_bits(gb, 3);
    if (bps_code == 3 || bps_code == 7) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "invalid sample size code (%d)\n",
               bps_code);
        return AVERROR_INVALIDDATA;
    }
    fi->bps = sample_size_table[bps_code];

    /* reserved bit */
    if (get_bits1(gb)) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "broken stream, invalid padding\n");
        return AVERROR_INVALIDDATA;
    }

    /* sample or frame count */
    fi->frame_or_sample_num = get_utf8(gb);
    if (fi->frame_or_sample_num < 0) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "sample/frame number invalid; utf8 fscked\n");
        return AVERROR_INVALIDDATA;
    }

    /* blocksize */
    if (bs_code == 0) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "reserved blocksize code: 0\n");
        return AVERROR_INVALIDDATA;
    } else if (bs_code == 6) {
        fi->blocksize = get_bits(gb, 8) + 1;
    } else if (bs_code == 7) {
        fi->blocksize = get_bits(gb, 16) + 1;
    } else {
        fi->blocksize = ff_flac_blocksize_table[bs_code];
    }

    /* sample rate */
    if (sr_code < 12) {
        fi->samplerate = ff_flac_sample_rate_table[sr_code];
    } else if (sr_code == 12) {
        fi->samplerate = get_bits(gb, 8) * 1000;
    } else if (sr_code == 13) {
        fi->samplerate = get_bits(gb, 16);
    } else if (sr_code == 14) {
        fi->samplerate = get_bits(gb, 16) * 10;
    } else {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "illegal sample rate code %d\n",
               sr_code);
        return AVERROR_INVALIDDATA;
    }

    /* header CRC-8 check */
    skip_bits(gb, 8);
    if (av_crc(av_crc_get_table(AV_CRC_8_ATM), 0, gb->buffer,
               get_bits_count(gb)/8)) {
        av_log(avctx, AV_LOG_ERROR + log_level_offset,
               "header crc mismatch\n");
        return AVERROR_INVALIDDATA;
    }

    return 0;
}

int ff_flac_get_max_frame_size(int blocksize, int ch, int bps)
{
    /* Technically, there is no limit to FLAC frame size, but an encoder
       should not write a frame that is larger than if verbatim encoding mode
       were to be used. */

    int count;

    count = 16;                  /* frame header */
    count += ch * ((7+bps+7)/8); /* subframe headers */
    if (ch == 2) {
        /* for stereo, need to account for using decorrelation */
        count += (( 2*bps+1) * blocksize + 7) / 8;
    } else {
        count += ( ch*bps    * blocksize + 7) / 8;
    }
    count += 2; /* frame footer */

    return count;
}

int ff_flac_is_extradata_valid(AVCodecContext *avctx,
                               enum FLACExtradataFormat *format,
                               uint8_t **streaminfo_start)
{
    if (!avctx->extradata || avctx->extradata_size < FLAC_STREAMINFO_SIZE) {
        av_log(avctx, AV_LOG_ERROR, "extradata NULL or too small.\n");
        return 0;
    }
    if (AV_RL32(avctx->extradata) != MKTAG('f','L','a','C')) {
        /* extradata contains STREAMINFO only */
        if (avctx->extradata_size != FLAC_STREAMINFO_SIZE) {
            av_log(avctx, AV_LOG_WARNING, "extradata contains %d bytes too many.\n",
                   FLAC_STREAMINFO_SIZE-avctx->extradata_size);
        }
        *format = FLAC_EXTRADATA_FORMAT_STREAMINFO;
        *streaminfo_start = avctx->extradata;
    } else {
        if (avctx->extradata_size < 8+FLAC_STREAMINFO_SIZE) {
            av_log(avctx, AV_LOG_ERROR, "extradata too small.\n");
            return 0;
        }
        *format = FLAC_EXTRADATA_FORMAT_FULL_HEADER;
        *streaminfo_start = &avctx->extradata[8];
    }
    return 1;
}

void ff_flac_set_channel_layout(AVCodecContext *avctx)
{
    if (avctx->channels <= FF_ARRAY_ELEMS(flac_channel_layouts))
        avctx->channel_layout = flac_channel_layouts[avctx->channels - 1];
    else
        avctx->channel_layout = 0;
}

int ff_flac_parse_streaminfo(AVCodecContext *avctx, struct FLACStreaminfo *s,
                              const uint8_t *buffer)
{
    GetBitContext gb;
    init_get_bits(&gb, buffer, FLAC_STREAMINFO_SIZE*8);

    skip_bits(&gb, 16); /* skip min blocksize */
    s->max_blocksize = get_bits(&gb, 16);
    if (s->max_blocksize < FLAC_MIN_BLOCKSIZE) {
        av_log(avctx, AV_LOG_WARNING, "invalid max blocksize: %d\n",
               s->max_blocksize);
        s->max_blocksize = 16;
        return AVERROR_INVALIDDATA;
    }

    skip_bits(&gb, 24); /* skip min frame size */
    s->max_framesize = get_bits_long(&gb, 24);

    s->samplerate = get_bits_long(&gb, 20);
    s->channels = get_bits(&gb, 3) + 1;
    s->bps = get_bits(&gb, 5) + 1;

    if (s->bps < 4) {
        av_log(avctx, AV_LOG_ERROR, "invalid bps: %d\n", s->bps);
        s->bps = 16;
        return AVERROR_INVALIDDATA;
    }

    avctx->channels = s->channels;
    avctx->sample_rate = s->samplerate;
    avctx->bits_per_raw_sample = s->bps;

    if (!avctx->channel_layout ||
        av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
        ff_flac_set_channel_layout(avctx);

    s->samples = get_bits64(&gb, 36);

    skip_bits_long(&gb, 64); /* md5 sum */
    skip_bits_long(&gb, 64); /* md5 sum */

    return 0;
}
OpenPOWER on IntegriCloud