| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * Digital Speech Standard (DSS) demuxer | ||
| 3 | * Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de> | ||
| 4 | * | ||
| 5 | * This file is part of FFmpeg. | ||
| 6 | * | ||
| 7 | * FFmpeg is free software; you can redistribute it and/or | ||
| 8 | * modify it under the terms of the GNU Lesser General Public | ||
| 9 | * License as published by the Free Software Foundation; either | ||
| 10 | * version 2.1 of the License, or (at your option) any later version. | ||
| 11 | * | ||
| 12 | * FFmpeg is distributed in the hope that it will be useful, | ||
| 13 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
| 14 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
| 15 | * Lesser General Public License for more details. | ||
| 16 | * | ||
| 17 | * You should have received a copy of the GNU Lesser General Public | ||
| 18 | * License along with FFmpeg; if not, write to the Free Software | ||
| 19 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
| 20 | */ | ||
| 21 | |||
| 22 | #include "libavutil/channel_layout.h" | ||
| 23 | #include "libavutil/intreadwrite.h" | ||
| 24 | #include "libavutil/mem.h" | ||
| 25 | |||
| 26 | #include "avformat.h" | ||
| 27 | #include "demux.h" | ||
| 28 | #include "internal.h" | ||
| 29 | #include "avio_internal.h" | ||
| 30 | |||
| 31 | #define DSS_HEAD_OFFSET_AUTHOR 0xc | ||
| 32 | #define DSS_AUTHOR_SIZE 16 | ||
| 33 | |||
| 34 | #define DSS_HEAD_OFFSET_START_TIME 0x26 | ||
| 35 | #define DSS_HEAD_OFFSET_END_TIME 0x32 | ||
| 36 | #define DSS_TIME_SIZE 12 | ||
| 37 | |||
| 38 | #define DSS_HEAD_OFFSET_ACODEC 0x2a4 | ||
| 39 | #define DSS_ACODEC_DSS_SP 0x0 /* SP mode */ | ||
| 40 | #define DSS_ACODEC_G723_1 0x2 /* LP mode */ | ||
| 41 | |||
| 42 | #define DSS_HEAD_OFFSET_COMMENT 0x31e | ||
| 43 | #define DSS_COMMENT_SIZE 64 | ||
| 44 | |||
| 45 | #define DSS_BLOCK_SIZE 512 | ||
| 46 | #define DSS_AUDIO_BLOCK_HEADER_SIZE 6 | ||
| 47 | #define DSS_FRAME_SIZE 42 | ||
| 48 | |||
| 49 | static const uint8_t frame_size[4] = { 24, 20, 4, 1 }; | ||
| 50 | |||
| 51 | typedef struct DSSDemuxContext { | ||
| 52 | unsigned int audio_codec; | ||
| 53 | int counter; | ||
| 54 | int swap; | ||
| 55 | int dss_sp_swap_byte; | ||
| 56 | |||
| 57 | int packet_size; | ||
| 58 | int dss_header_size; | ||
| 59 | } DSSDemuxContext; | ||
| 60 | |||
| 61 | 7480 | static int dss_probe(const AVProbeData *p) | |
| 62 | { | ||
| 63 |
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7480 | if ( AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's') |
| 64 |
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7478 | && AV_RL32(p->buf) != MKTAG(0x3, 'd', 's', 's')) |
| 65 | 7478 | return 0; | |
| 66 | |||
| 67 | 2 | return AVPROBE_SCORE_MAX; | |
| 68 | } | ||
| 69 | |||
| 70 | 2 | static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset, | |
| 71 | const char *key) | ||
| 72 | { | ||
| 73 | 2 | AVIOContext *pb = s->pb; | |
| 74 | 2 | char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 }; | |
| 75 | int y, month, d, h, minute, sec; | ||
| 76 | int ret; | ||
| 77 | |||
| 78 | 2 | avio_seek(pb, offset, SEEK_SET); | |
| 79 | |||
| 80 | 2 | ret = avio_read(s->pb, string, DSS_TIME_SIZE); | |
| 81 |
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2 | if (ret < DSS_TIME_SIZE) |
| 82 | ✗ | return ret < 0 ? ret : AVERROR_EOF; | |
| 83 | |||
| 84 |
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2 | if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6) |
| 85 | ✗ | return AVERROR_INVALIDDATA; | |
| 86 | /* We deal with a two-digit year here, so set the default date to 2000 | ||
| 87 | * and hope it will never be used in the next century. */ | ||
| 88 | 2 | snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d", | |
| 89 | y + 2000, month, d, h, minute, sec); | ||
| 90 | 2 | return av_dict_set(&s->metadata, key, datetime, 0); | |
| 91 | } | ||
| 92 | |||
| 93 | 4 | static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset, | |
| 94 | unsigned int size, const char *key) | ||
| 95 | { | ||
| 96 | 4 | AVIOContext *pb = s->pb; | |
| 97 | char *value; | ||
| 98 | int ret; | ||
| 99 | |||
| 100 | 4 | avio_seek(pb, offset, SEEK_SET); | |
| 101 | |||
| 102 | 4 | value = av_mallocz(size + 1); | |
| 103 |
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4 | if (!value) |
| 104 | ✗ | return AVERROR(ENOMEM); | |
| 105 | |||
| 106 | 4 | ret = avio_read(s->pb, value, size); | |
| 107 |
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4 | if (ret < size) { |
| 108 | ✗ | av_free(value); | |
| 109 | ✗ | return ret < 0 ? ret : AVERROR_EOF; | |
| 110 | } | ||
| 111 | |||
| 112 | 4 | return av_dict_set(&s->metadata, key, value, AV_DICT_DONT_STRDUP_VAL); | |
| 113 | } | ||
| 114 | |||
| 115 | 2 | static int dss_read_header(AVFormatContext *s) | |
| 116 | { | ||
| 117 | 2 | DSSDemuxContext *ctx = s->priv_data; | |
| 118 | 2 | AVIOContext *pb = s->pb; | |
| 119 | AVStream *st; | ||
| 120 | int64_t ret64; | ||
| 121 | int ret, version; | ||
| 122 | |||
| 123 | 2 | st = avformat_new_stream(s, NULL); | |
| 124 |
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2 | if (!st) |
| 125 | ✗ | return AVERROR(ENOMEM); | |
| 126 | |||
| 127 | 2 | version = avio_r8(pb); | |
| 128 | 2 | ctx->dss_header_size = version * DSS_BLOCK_SIZE; | |
| 129 | |||
| 130 | 2 | ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR, | |
| 131 | DSS_AUTHOR_SIZE, "author"); | ||
| 132 |
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2 | if (ret) |
| 133 | ✗ | return ret; | |
| 134 | |||
| 135 | 2 | ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date"); | |
| 136 |
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2 | if (ret) |
| 137 | ✗ | return ret; | |
| 138 | |||
| 139 | 2 | ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT, | |
| 140 | DSS_COMMENT_SIZE, "comment"); | ||
| 141 |
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2 | if (ret) |
| 142 | ✗ | return ret; | |
| 143 | |||
| 144 | 2 | avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET); | |
| 145 | 2 | ctx->audio_codec = avio_r8(pb); | |
| 146 | |||
| 147 |
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2 | if (ctx->audio_codec == DSS_ACODEC_DSS_SP) { |
| 148 | 1 | st->codecpar->codec_id = AV_CODEC_ID_DSS_SP; | |
| 149 | 1 | st->codecpar->sample_rate = 11025; | |
| 150 | 1 | s->bit_rate = 8 * (DSS_FRAME_SIZE - 1) * st->codecpar->sample_rate | |
| 151 | 1 | * 512 / (506 * 264); | |
| 152 |
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1 | } else if (ctx->audio_codec == DSS_ACODEC_G723_1) { |
| 153 | 1 | st->codecpar->codec_id = AV_CODEC_ID_G723_1; | |
| 154 | 1 | st->codecpar->sample_rate = 8000; | |
| 155 | } else { | ||
| 156 | ✗ | avpriv_request_sample(s, "Support for codec %x in DSS", | |
| 157 | ctx->audio_codec); | ||
| 158 | ✗ | return AVERROR_PATCHWELCOME; | |
| 159 | } | ||
| 160 | |||
| 161 | 2 | st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; | |
| 162 | 2 | st->codecpar->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO; | |
| 163 | |||
| 164 | 2 | avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate); | |
| 165 | 2 | st->start_time = 0; | |
| 166 | |||
| 167 | /* Jump over header */ | ||
| 168 | |||
| 169 |
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2 | if ((ret64 = avio_seek(pb, ctx->dss_header_size, SEEK_SET)) < 0) |
| 170 | ✗ | return (int)ret64; | |
| 171 | |||
| 172 | 2 | ctx->counter = 0; | |
| 173 | 2 | ctx->swap = 0; | |
| 174 | |||
| 175 | 2 | return 0; | |
| 176 | } | ||
| 177 | |||
| 178 | 8 | static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt) | |
| 179 | { | ||
| 180 | 8 | DSSDemuxContext *ctx = s->priv_data; | |
| 181 | 8 | AVIOContext *pb = s->pb; | |
| 182 | |||
| 183 | 8 | avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE); | |
| 184 | 8 | ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE; | |
| 185 | 8 | } | |
| 186 | |||
| 187 | 50 | static void dss_sp_byte_swap(DSSDemuxContext *ctx, uint8_t *data) | |
| 188 | { | ||
| 189 | int i; | ||
| 190 | |||
| 191 |
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50 | if (ctx->swap) { |
| 192 |
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525 | for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2) |
| 193 | 500 | data[i] = data[i + 4]; | |
| 194 | |||
| 195 | /* Zero the padding. */ | ||
| 196 | 25 | data[DSS_FRAME_SIZE] = 0; | |
| 197 | 25 | data[1] = ctx->dss_sp_swap_byte; | |
| 198 | } else { | ||
| 199 | 25 | ctx->dss_sp_swap_byte = data[DSS_FRAME_SIZE - 2]; | |
| 200 | } | ||
| 201 | |||
| 202 | /* make sure byte 40 is always 0 */ | ||
| 203 | 50 | data[DSS_FRAME_SIZE - 2] = 0; | |
| 204 | 50 | ctx->swap ^= 1; | |
| 205 | 50 | } | |
| 206 | |||
| 207 | 50 | static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt) | |
| 208 | { | ||
| 209 | 50 | DSSDemuxContext *ctx = s->priv_data; | |
| 210 | 50 | int read_size, ret, offset = 0, buff_offset = 0; | |
| 211 | 50 | int64_t pos = avio_tell(s->pb); | |
| 212 | |||
| 213 |
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50 | if (ctx->counter == 0) |
| 214 | 2 | dss_skip_audio_header(s, pkt); | |
| 215 | |||
| 216 |
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50 | if (ctx->swap) { |
| 217 | 25 | read_size = DSS_FRAME_SIZE - 2; | |
| 218 | 25 | buff_offset = 3; | |
| 219 | } else | ||
| 220 | 25 | read_size = DSS_FRAME_SIZE; | |
| 221 | |||
| 222 | 50 | ret = av_new_packet(pkt, DSS_FRAME_SIZE); | |
| 223 |
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50 | if (ret < 0) |
| 224 | ✗ | return ret; | |
| 225 | |||
| 226 | 50 | pkt->duration = 264; | |
| 227 | 50 | pkt->pos = pos; | |
| 228 | 50 | pkt->stream_index = 0; | |
| 229 | |||
| 230 |
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50 | if (ctx->counter < read_size) { |
| 231 | 3 | ret = avio_read(s->pb, pkt->data + buff_offset, | |
| 232 | ctx->counter); | ||
| 233 |
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3 | if (ret < ctx->counter) |
| 234 | ✗ | goto error_eof; | |
| 235 | |||
| 236 | 3 | offset = ctx->counter; | |
| 237 | 3 | dss_skip_audio_header(s, pkt); | |
| 238 | } | ||
| 239 | 50 | ctx->counter -= read_size; | |
| 240 | |||
| 241 | /* This will write one byte into pkt's padding if buff_offset == 3 */ | ||
| 242 | 50 | ret = avio_read(s->pb, pkt->data + offset + buff_offset, | |
| 243 | read_size - offset); | ||
| 244 |
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50 | if (ret < read_size - offset) |
| 245 | ✗ | goto error_eof; | |
| 246 | |||
| 247 | 50 | dss_sp_byte_swap(ctx, pkt->data); | |
| 248 | |||
| 249 |
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50 | if (ctx->dss_sp_swap_byte < 0) { |
| 250 | ✗ | return AVERROR(EAGAIN); | |
| 251 | } | ||
| 252 | |||
| 253 | 50 | return 0; | |
| 254 | |||
| 255 | ✗ | error_eof: | |
| 256 | ✗ | return ret < 0 ? ret : AVERROR_EOF; | |
| 257 | } | ||
| 258 | |||
| 259 | 50 | static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt) | |
| 260 | { | ||
| 261 | 50 | DSSDemuxContext *ctx = s->priv_data; | |
| 262 | 50 | AVStream *st = s->streams[0]; | |
| 263 | int size, byte, ret, offset; | ||
| 264 | 50 | int64_t pos = avio_tell(s->pb); | |
| 265 | |||
| 266 |
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50 | if (ctx->counter == 0) |
| 267 | 1 | dss_skip_audio_header(s, pkt); | |
| 268 | |||
| 269 | /* We make one byte-step here. Don't forget to add offset. */ | ||
| 270 | 50 | byte = avio_r8(s->pb); | |
| 271 |
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50 | if (byte == 0xff) |
| 272 | ✗ | return AVERROR_INVALIDDATA; | |
| 273 | |||
| 274 | 50 | size = frame_size[byte & 3]; | |
| 275 | |||
| 276 | 50 | ctx->packet_size = size; | |
| 277 | 50 | ctx->counter--; | |
| 278 | |||
| 279 | 50 | ret = av_new_packet(pkt, size); | |
| 280 |
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50 | if (ret < 0) |
| 281 | ✗ | return ret; | |
| 282 | 50 | pkt->pos = pos; | |
| 283 | |||
| 284 | 50 | pkt->data[0] = byte; | |
| 285 | 50 | offset = 1; | |
| 286 | 50 | pkt->duration = 240; | |
| 287 | 50 | s->bit_rate = 8LL * size-- * st->codecpar->sample_rate * 512 / (506 * pkt->duration); | |
| 288 | |||
| 289 | 50 | pkt->stream_index = 0; | |
| 290 | |||
| 291 |
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50 | if (ctx->counter < size) { |
| 292 | 2 | ret = avio_read(s->pb, pkt->data + offset, | |
| 293 | ctx->counter); | ||
| 294 |
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2 | if (ret < ctx->counter) |
| 295 | ✗ | return ret < 0 ? ret : AVERROR_EOF; | |
| 296 | |||
| 297 | 2 | offset += ctx->counter; | |
| 298 | 2 | size -= ctx->counter; | |
| 299 | 2 | ctx->counter = 0; | |
| 300 | 2 | dss_skip_audio_header(s, pkt); | |
| 301 | } | ||
| 302 | 50 | ctx->counter -= size; | |
| 303 | |||
| 304 | 50 | ret = avio_read(s->pb, pkt->data + offset, size); | |
| 305 |
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50 | if (ret < size) |
| 306 | ✗ | return ret < 0 ? ret : AVERROR_EOF; | |
| 307 | |||
| 308 | 50 | return 0; | |
| 309 | } | ||
| 310 | |||
| 311 | 100 | static int dss_read_packet(AVFormatContext *s, AVPacket *pkt) | |
| 312 | { | ||
| 313 | 100 | DSSDemuxContext *ctx = s->priv_data; | |
| 314 | |||
| 315 |
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100 | if (ctx->audio_codec == DSS_ACODEC_DSS_SP) |
| 316 | 50 | return dss_sp_read_packet(s, pkt); | |
| 317 | else | ||
| 318 | 50 | return dss_723_1_read_packet(s, pkt); | |
| 319 | } | ||
| 320 | |||
| 321 | ✗ | static int dss_read_seek(AVFormatContext *s, int stream_index, | |
| 322 | int64_t timestamp, int flags) | ||
| 323 | { | ||
| 324 | ✗ | DSSDemuxContext *ctx = s->priv_data; | |
| 325 | int64_t ret, seekto; | ||
| 326 | uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE]; | ||
| 327 | int offset; | ||
| 328 | |||
| 329 | ✗ | if (ctx->audio_codec == DSS_ACODEC_DSS_SP) | |
| 330 | ✗ | seekto = timestamp / 264 * 41 / 506 * 512; | |
| 331 | else | ||
| 332 | ✗ | seekto = timestamp / 240 * ctx->packet_size / 506 * 512; | |
| 333 | |||
| 334 | ✗ | if (seekto < 0) | |
| 335 | ✗ | seekto = 0; | |
| 336 | |||
| 337 | ✗ | seekto += ctx->dss_header_size; | |
| 338 | |||
| 339 | ✗ | ret = avio_seek(s->pb, seekto, SEEK_SET); | |
| 340 | ✗ | if (ret < 0) | |
| 341 | ✗ | return ret; | |
| 342 | |||
| 343 | ✗ | ret = ffio_read_size(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE); | |
| 344 | ✗ | if (ret < 0) | |
| 345 | ✗ | return ret; | |
| 346 | ✗ | ctx->swap = !!(header[0] & 0x80); | |
| 347 | ✗ | offset = 2*header[1] + 2*ctx->swap; | |
| 348 | ✗ | if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE) | |
| 349 | ✗ | return AVERROR_INVALIDDATA; | |
| 350 | ✗ | if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) { | |
| 351 | ✗ | ctx->counter = 0; | |
| 352 | ✗ | offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE); | |
| 353 | } else { | ||
| 354 | ✗ | ctx->counter = DSS_BLOCK_SIZE - offset; | |
| 355 | ✗ | offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE); | |
| 356 | } | ||
| 357 | ✗ | ctx->dss_sp_swap_byte = -1; | |
| 358 | ✗ | return 0; | |
| 359 | } | ||
| 360 | |||
| 361 | |||
| 362 | const FFInputFormat ff_dss_demuxer = { | ||
| 363 | .p.name = "dss", | ||
| 364 | .p.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"), | ||
| 365 | .p.extensions = "dss", | ||
| 366 | .priv_data_size = sizeof(DSSDemuxContext), | ||
| 367 | .read_probe = dss_probe, | ||
| 368 | .read_header = dss_read_header, | ||
| 369 | .read_packet = dss_read_packet, | ||
| 370 | .read_seek = dss_read_seek, | ||
| 371 | }; | ||
| 372 |