/* * Digital Speech Standard (DSS) demuxer * Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de> * * 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/attributes.h" #include "libavutil/bswap.h" #include "libavutil/channel_layout.h" #include "libavutil/intreadwrite.h" #include "avformat.h" #include "internal.h" #define DSS_HEAD_OFFSET_AUTHOR 0xc #define DSS_AUTHOR_SIZE 16 #define DSS_HEAD_OFFSET_START_TIME 0x26 #define DSS_HEAD_OFFSET_END_TIME 0x32 #define DSS_TIME_SIZE 12 #define DSS_HEAD_OFFSET_ACODEC 0x2a4 #define DSS_ACODEC_DSS_SP 0x0 /* SP mode */ #define DSS_ACODEC_G723_1 0x2 /* LP mode */ #define DSS_HEAD_OFFSET_COMMENT 0x31e #define DSS_COMMENT_SIZE 64 #define DSS_BLOCK_SIZE 512 #define DSS_AUDIO_BLOCK_HEADER_SIZE 6 #define DSS_FRAME_SIZE 42 static const uint8_t frame_size[4] = { 24, 20, 4, 1 }; typedef struct DSSDemuxContext { unsigned int audio_codec; int counter; int swap; int dss_sp_swap_byte; int8_t *dss_sp_buf; int packet_size; int dss_header_size; } DSSDemuxContext; static int dss_probe(AVProbeData *p) { if ( AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's') && AV_RL32(p->buf) != MKTAG(0x3, 'd', 's', 's')) return 0; return AVPROBE_SCORE_MAX; } static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset, const char *key) { AVIOContext *pb = s->pb; char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 }; int y, month, d, h, minute, sec; int ret; avio_seek(pb, offset, SEEK_SET); ret = avio_read(s->pb, string, DSS_TIME_SIZE); if (ret < DSS_TIME_SIZE) return ret < 0 ? ret : AVERROR_EOF; if (sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec) != 6) return AVERROR_INVALIDDATA; /* We deal with a two-digit year here, so set the default date to 2000 * and hope it will never be used in the next century. */ snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d", y + 2000, month, d, h, minute, sec); return av_dict_set(&s->metadata, key, datetime, 0); } static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset, unsigned int size, const char *key) { AVIOContext *pb = s->pb; char *value; int ret; avio_seek(pb, offset, SEEK_SET); value = av_mallocz(size + 1); if (!value) return AVERROR(ENOMEM); ret = avio_read(s->pb, value, size); if (ret < size) { ret = ret < 0 ? ret : AVERROR_EOF; goto exit; } ret = av_dict_set(&s->metadata, key, value, 0); exit: av_free(value); return ret; } static int dss_read_header(AVFormatContext *s) { DSSDemuxContext *ctx = s->priv_data; AVIOContext *pb = s->pb; AVStream *st; int ret, version; st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); version = avio_r8(pb); ctx->dss_header_size = version * DSS_BLOCK_SIZE; ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR, DSS_AUTHOR_SIZE, "author"); if (ret) return ret; ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date"); if (ret) return ret; ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT, DSS_COMMENT_SIZE, "comment"); if (ret) return ret; avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET); ctx->audio_codec = avio_r8(pb); if (ctx->audio_codec == DSS_ACODEC_DSS_SP) { st->codecpar->codec_id = AV_CODEC_ID_DSS_SP; st->codecpar->sample_rate = 11025; } else if (ctx->audio_codec == DSS_ACODEC_G723_1) { st->codecpar->codec_id = AV_CODEC_ID_G723_1; st->codecpar->sample_rate = 8000; } else { avpriv_request_sample(s, "Support for codec %x in DSS", ctx->audio_codec); return AVERROR_PATCHWELCOME; } st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO; st->codecpar->channel_layout = AV_CH_LAYOUT_MONO; st->codecpar->channels = 1; avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate); st->start_time = 0; /* Jump over header */ if (avio_seek(pb, ctx->dss_header_size, SEEK_SET) != ctx->dss_header_size) return AVERROR(EIO); ctx->counter = 0; ctx->swap = 0; ctx->dss_sp_buf = av_malloc(DSS_FRAME_SIZE + 1); if (!ctx->dss_sp_buf) return AVERROR(ENOMEM); return 0; } static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt) { DSSDemuxContext *ctx = s->priv_data; AVIOContext *pb = s->pb; avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE); ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE; } static void dss_sp_byte_swap(DSSDemuxContext *ctx, uint8_t *dst, const uint8_t *src) { int i; if (ctx->swap) { for (i = 3; i < DSS_FRAME_SIZE; i += 2) dst[i] = src[i]; for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2) dst[i] = src[i + 4]; dst[1] = ctx->dss_sp_swap_byte; } else { memcpy(dst, src, DSS_FRAME_SIZE); ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2]; } /* make sure byte 40 is always 0 */ dst[DSS_FRAME_SIZE - 2] = 0; ctx->swap ^= 1; } static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt) { DSSDemuxContext *ctx = s->priv_data; AVStream *st = s->streams[0]; int read_size, ret, offset = 0, buff_offset = 0; int64_t pos = avio_tell(s->pb); if (ctx->counter == 0) dss_skip_audio_header(s, pkt); if (ctx->swap) { read_size = DSS_FRAME_SIZE - 2; buff_offset = 3; } else read_size = DSS_FRAME_SIZE; ctx->counter -= read_size; ctx->packet_size = DSS_FRAME_SIZE - 1; ret = av_new_packet(pkt, DSS_FRAME_SIZE); if (ret < 0) return ret; pkt->duration = 264; pkt->pos = pos; pkt->stream_index = 0; s->bit_rate = 8LL * ctx->packet_size * st->codecpar->sample_rate * 512 / (506 * pkt->duration); if (ctx->counter < 0) { int size2 = ctx->counter + read_size; ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset, size2 - offset); if (ret < size2 - offset) goto error_eof; dss_skip_audio_header(s, pkt); offset = size2; } ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset, read_size - offset); if (ret < read_size - offset) goto error_eof; dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf); if (ctx->dss_sp_swap_byte < 0) { ret = AVERROR(EAGAIN); goto error_eof; } return pkt->size; error_eof: av_packet_unref(pkt); return ret < 0 ? ret : AVERROR_EOF; } static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt) { DSSDemuxContext *ctx = s->priv_data; AVStream *st = s->streams[0]; int size, byte, ret, offset; int64_t pos = avio_tell(s->pb); if (ctx->counter == 0) dss_skip_audio_header(s, pkt); /* We make one byte-step here. Don't forget to add offset. */ byte = avio_r8(s->pb); if (byte == 0xff) return AVERROR_INVALIDDATA; size = frame_size[byte & 3]; ctx->packet_size = size; ctx->counter -= size; ret = av_new_packet(pkt, size); if (ret < 0) return ret; pkt->pos = pos; pkt->data[0] = byte; offset = 1; pkt->duration = 240; s->bit_rate = 8LL * size * st->codecpar->sample_rate * 512 / (506 * pkt->duration); pkt->stream_index = 0; if (ctx->counter < 0) { int size2 = ctx->counter + size; ret = avio_read(s->pb, pkt->data + offset, size2 - offset); if (ret < size2 - offset) { av_packet_unref(pkt); return ret < 0 ? ret : AVERROR_EOF; } dss_skip_audio_header(s, pkt); offset = size2; } ret = avio_read(s->pb, pkt->data + offset, size - offset); if (ret < size - offset) { av_packet_unref(pkt); return ret < 0 ? ret : AVERROR_EOF; } return pkt->size; } static int dss_read_packet(AVFormatContext *s, AVPacket *pkt) { DSSDemuxContext *ctx = s->priv_data; if (ctx->audio_codec == DSS_ACODEC_DSS_SP) return dss_sp_read_packet(s, pkt); else return dss_723_1_read_packet(s, pkt); } static int dss_read_close(AVFormatContext *s) { DSSDemuxContext *ctx = s->priv_data; av_freep(&ctx->dss_sp_buf); return 0; } static int dss_read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags) { DSSDemuxContext *ctx = s->priv_data; int64_t ret, seekto; uint8_t header[DSS_AUDIO_BLOCK_HEADER_SIZE]; int offset; if (ctx->audio_codec == DSS_ACODEC_DSS_SP) seekto = timestamp / 264 * 41 / 506 * 512; else seekto = timestamp / 240 * ctx->packet_size / 506 * 512; if (seekto < 0) seekto = 0; seekto += ctx->dss_header_size; ret = avio_seek(s->pb, seekto, SEEK_SET); if (ret < 0) return ret; avio_read(s->pb, header, DSS_AUDIO_BLOCK_HEADER_SIZE); ctx->swap = !!(header[0] & 0x80); offset = 2*header[1] + 2*ctx->swap; if (offset < DSS_AUDIO_BLOCK_HEADER_SIZE) return AVERROR_INVALIDDATA; if (offset == DSS_AUDIO_BLOCK_HEADER_SIZE) { ctx->counter = 0; offset = avio_skip(s->pb, -DSS_AUDIO_BLOCK_HEADER_SIZE); } else { ctx->counter = DSS_BLOCK_SIZE - offset; offset = avio_skip(s->pb, offset - DSS_AUDIO_BLOCK_HEADER_SIZE); } ctx->dss_sp_swap_byte = -1; return 0; } AVInputFormat ff_dss_demuxer = { .name = "dss", .long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"), .priv_data_size = sizeof(DSSDemuxContext), .read_probe = dss_probe, .read_header = dss_read_header, .read_packet = dss_read_packet, .read_close = dss_read_close, .read_seek = dss_read_seek, .extensions = "dss" };