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https://github.com/FFmpeg/FFmpeg.git
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lavf: Add DSS demuxer
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de> Signed-off-by: Luca Barbato <lu_zero@gentoo.org> Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
This commit is contained in:
Oleksij Rempel
Vittorio Giovara
parent
c56b9b1eb2
commit
062cd5a975
@ -15,7 +15,7 @@ version <next>:
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- DXVA2-accelerated HEVC decoding
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- AAC ELD 480 decoding
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- Intel QSV-accelerated H.264 decoding
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- DSS SP decoder
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- DSS SP decoder and DSS demuxer
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version 11:
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@ -221,6 +221,7 @@ library:
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@tab Used in the game Cyberia from Interplay.
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@item Delphine Software International CIN @tab @tab X
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@tab Multimedia format used by Delphine Software games.
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@item Digital Speech Standard (DSS) @tab @tab X
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@item CD+G @tab @tab X
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@tab Video format used by CD+G karaoke disks
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@item Commodore CDXL @tab @tab X
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@ -100,6 +100,7 @@ OBJS-$(CONFIG_DIRAC_MUXER) += rawenc.o
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OBJS-$(CONFIG_DNXHD_DEMUXER) += dnxhddec.o rawdec.o
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OBJS-$(CONFIG_DNXHD_MUXER) += rawenc.o
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OBJS-$(CONFIG_DSICIN_DEMUXER) += dsicin.o
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OBJS-$(CONFIG_DSS_DEMUXER) += dss.o
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OBJS-$(CONFIG_DTS_DEMUXER) += dtsdec.o rawdec.o
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OBJS-$(CONFIG_DTS_MUXER) += rawenc.o
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OBJS-$(CONFIG_DV_DEMUXER) += dv.o
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@ -94,6 +94,7 @@ void av_register_all(void)
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REGISTER_MUXDEMUX(DIRAC, dirac);
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REGISTER_MUXDEMUX(DNXHD, dnxhd);
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REGISTER_DEMUXER (DSICIN, dsicin);
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REGISTER_DEMUXER (DSS, dss);
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REGISTER_MUXDEMUX(DTS, dts);
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REGISTER_MUXDEMUX(DV, dv);
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REGISTER_DEMUXER (DXA, dxa);
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342
libavformat/dss.c
Normal file
342
libavformat/dss.c
Normal file
@ -0,0 +1,342 @@
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/*
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* Digital Speech Standard (DSS) demuxer
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* Copyright (c) 2014 Oleksij Rempel <linux@rempel-privat.de>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/attributes.h"
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#include "libavutil/bswap.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/intreadwrite.h"
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#include "avformat.h"
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#include "internal.h"
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#define DSS_HEAD_OFFSET_AUTHOR 0xc
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#define DSS_AUTHOR_SIZE 16
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#define DSS_HEAD_OFFSET_START_TIME 0x26
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#define DSS_HEAD_OFFSET_END_TIME 0x32
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#define DSS_TIME_SIZE 12
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#define DSS_HEAD_OFFSET_ACODEC 0x2a4
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#define DSS_ACODEC_DSS_SP 0x0 /* SP mode */
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#define DSS_ACODEC_G723_1 0x2 /* LP mode */
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#define DSS_HEAD_OFFSET_COMMENT 0x31e
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#define DSS_COMMENT_SIZE 64
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#define DSS_BLOCK_SIZE 512
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#define DSS_HEADER_SIZE (DSS_BLOCK_SIZE * 2)
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#define DSS_AUDIO_BLOCK_HEADER_SIZE 6
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#define DSS_FRAME_SIZE 42
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static const uint8_t frame_size[4] = { 24, 20, 4, 1 };
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typedef struct DSSDemuxContext {
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unsigned int audio_codec;
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int counter;
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int swap;
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int dss_sp_swap_byte;
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int8_t *dss_sp_buf;
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} DSSDemuxContext;
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static int dss_probe(AVProbeData *p)
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{
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if (AV_RL32(p->buf) != MKTAG(0x2, 'd', 's', 's'))
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return 0;
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return AVPROBE_SCORE_MAX;
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}
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static int dss_read_metadata_date(AVFormatContext *s, unsigned int offset,
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const char *key)
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{
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AVIOContext *pb = s->pb;
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char datetime[64], string[DSS_TIME_SIZE + 1] = { 0 };
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int y, month, d, h, minute, sec;
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int ret;
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avio_seek(pb, offset, SEEK_SET);
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ret = avio_read(s->pb, string, DSS_TIME_SIZE);
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if (ret < DSS_TIME_SIZE)
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return ret < 0 ? ret : AVERROR_EOF;
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sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec);
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/* We deal with a two-digit year here, so set the default date to 2000
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* and hope it will never be used in the next century. */
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snprintf(datetime, sizeof(datetime), "%.4d-%.2d-%.2dT%.2d:%.2d:%.2d",
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y + 2000, month, d, h, minute, sec);
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return av_dict_set(&s->metadata, key, datetime, 0);
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}
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static int dss_read_metadata_string(AVFormatContext *s, unsigned int offset,
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unsigned int size, const char *key)
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{
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AVIOContext *pb = s->pb;
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char *value;
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int ret;
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avio_seek(pb, offset, SEEK_SET);
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value = av_mallocz(size + 1);
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if (!value)
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return AVERROR(ENOMEM);
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ret = avio_read(s->pb, value, size);
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if (ret < size) {
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ret = ret < 0 ? ret : AVERROR_EOF;
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goto exit;
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}
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ret = av_dict_set(&s->metadata, key, value, 0);
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exit:
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av_free(value);
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return ret;
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}
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static int dss_read_header(AVFormatContext *s)
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{
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DSSDemuxContext *ctx = s->priv_data;
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AVIOContext *pb = s->pb;
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AVStream *st;
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int ret;
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_AUTHOR,
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DSS_AUTHOR_SIZE, "author");
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if (ret)
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return ret;
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ret = dss_read_metadata_date(s, DSS_HEAD_OFFSET_END_TIME, "date");
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if (ret)
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return ret;
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ret = dss_read_metadata_string(s, DSS_HEAD_OFFSET_COMMENT,
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DSS_COMMENT_SIZE, "comment");
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if (ret)
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return ret;
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avio_seek(pb, DSS_HEAD_OFFSET_ACODEC, SEEK_SET);
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ctx->audio_codec = avio_r8(pb);
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if (ctx->audio_codec == DSS_ACODEC_DSS_SP) {
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st->codec->codec_id = AV_CODEC_ID_DSS_SP;
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st->codec->sample_rate = 12000;
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} else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
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st->codec->codec_id = AV_CODEC_ID_G723_1;
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st->codec->sample_rate = 8000;
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} else {
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avpriv_request_sample(s, "Support for codec %x in DSS",
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ctx->audio_codec);
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return AVERROR_PATCHWELCOME;
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}
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->channel_layout = AV_CH_LAYOUT_MONO;
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st->codec->channels = 1;
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avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
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st->start_time = 0;
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/* Jump over header */
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if (avio_seek(pb, DSS_HEADER_SIZE, SEEK_SET) != DSS_HEADER_SIZE)
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return AVERROR(EIO);
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ctx->counter = 0;
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ctx->swap = 0;
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ctx->dss_sp_buf = av_malloc(DSS_FRAME_SIZE + 1);
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if (!ctx->dss_sp_buf)
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return AVERROR(ENOMEM);
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return 0;
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}
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static void dss_skip_audio_header(AVFormatContext *s, AVPacket *pkt)
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{
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DSSDemuxContext *ctx = s->priv_data;
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AVIOContext *pb = s->pb;
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avio_skip(pb, DSS_AUDIO_BLOCK_HEADER_SIZE);
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ctx->counter += DSS_BLOCK_SIZE - DSS_AUDIO_BLOCK_HEADER_SIZE;
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}
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static void dss_sp_byte_swap(DSSDemuxContext *ctx,
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uint8_t *dst, const uint8_t *src)
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{
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int i;
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if (ctx->swap) {
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for (i = 3; i < DSS_FRAME_SIZE; i += 2)
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dst[i] = src[i];
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for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
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dst[i] = src[i + 4];
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dst[1] = ctx->dss_sp_swap_byte;
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} else {
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memcpy(dst, src, DSS_FRAME_SIZE);
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ctx->dss_sp_swap_byte = src[DSS_FRAME_SIZE - 2];
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}
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/* make sure byte 40 is always 0 */
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dst[DSS_FRAME_SIZE - 2] = 0;
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ctx->swap ^= 1;
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}
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static int dss_sp_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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DSSDemuxContext *ctx = s->priv_data;
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int read_size, ret, offset = 0, buff_offset = 0;
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if (ctx->counter == 0)
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dss_skip_audio_header(s, pkt);
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pkt->pos = avio_tell(s->pb);
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if (ctx->swap) {
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read_size = DSS_FRAME_SIZE - 2;
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buff_offset = 3;
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} else
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read_size = DSS_FRAME_SIZE;
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ctx->counter -= read_size;
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ret = av_new_packet(pkt, DSS_FRAME_SIZE);
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if (ret < 0)
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return ret;
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pkt->duration = 0;
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pkt->stream_index = 0;
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if (ctx->counter < 0) {
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int size2 = ctx->counter + read_size;
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ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
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size2 - offset);
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if (ret < size2 - offset)
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goto error_eof;
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dss_skip_audio_header(s, pkt);
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offset = size2;
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}
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ret = avio_read(s->pb, ctx->dss_sp_buf + offset + buff_offset,
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read_size - offset);
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if (ret < read_size - offset)
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goto error_eof;
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dss_sp_byte_swap(ctx, pkt->data, ctx->dss_sp_buf);
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if (pkt->data[0] == 0xff)
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return AVERROR_INVALIDDATA;
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return pkt->size;
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error_eof:
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av_free_packet(pkt);
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return ret < 0 ? ret : AVERROR_EOF;
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}
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static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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DSSDemuxContext *ctx = s->priv_data;
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int size, byte, ret, offset;
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if (ctx->counter == 0)
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dss_skip_audio_header(s, pkt);
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pkt->pos = avio_tell(s->pb);
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/* We make one byte-step here. Don't forget to add offset. */
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byte = avio_r8(s->pb);
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if (byte == 0xff)
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return AVERROR_INVALIDDATA;
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size = frame_size[byte & 3];
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ctx->counter -= size;
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ret = av_new_packet(pkt, size);
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if (ret < 0)
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return ret;
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pkt->data[0] = byte;
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offset = 1;
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pkt->duration = 240;
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pkt->stream_index = 0;
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if (ctx->counter < 0) {
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int size2 = ctx->counter + size;
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ret = avio_read(s->pb, pkt->data + offset,
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size2 - offset);
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if (ret < size2 - offset) {
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av_free_packet(pkt);
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return ret < 0 ? ret : AVERROR_EOF;
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}
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dss_skip_audio_header(s, pkt);
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offset = size2;
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}
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ret = avio_read(s->pb, pkt->data + offset, size - offset);
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if (ret < size - offset) {
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av_free_packet(pkt);
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return ret < 0 ? ret : AVERROR_EOF;
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}
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return pkt->size;
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}
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static int dss_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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DSSDemuxContext *ctx = s->priv_data;
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if (ctx->audio_codec == DSS_ACODEC_DSS_SP)
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return dss_sp_read_packet(s, pkt);
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else
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return dss_723_1_read_packet(s, pkt);
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}
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static int dss_read_close(AVFormatContext *s)
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{
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DSSDemuxContext *ctx = s->priv_data;
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av_free(ctx->dss_sp_buf);
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return 0;
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}
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AVInputFormat ff_dss_demuxer = {
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.name = "dss",
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.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
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.priv_data_size = sizeof(DSSDemuxContext),
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.read_probe = dss_probe,
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.read_header = dss_read_header,
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.read_packet = dss_read_packet,
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.read_close = dss_read_close,
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.extensions = "dss"
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};
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFORMAT_VERSION_MAJOR 56
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#define LIBAVFORMAT_VERSION_MINOR 11
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#define LIBAVFORMAT_VERSION_MINOR 12
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#define LIBAVFORMAT_VERSION_MICRO 0
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#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
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