[FFmpeg-cvslog] lavf: Add DSS demuxer
Oleksij Rempel
git at videolan.org
Thu Feb 19 21:53:21 CET 2015
ffmpeg | branch: master | Oleksij Rempel <linux at rempel-privat.de> | Fri Feb 13 08:36:17 2015 +0100| [062cd5a975ff7bd6fb91f9b4d1d9d102a7545499] | committer: Vittorio Giovara
lavf: Add DSS demuxer
Signed-off-by: Oleksij Rempel <linux at rempel-privat.de>
Signed-off-by: Luca Barbato <lu_zero at gentoo.org>
Signed-off-by: Vittorio Giovara <vittorio.giovara at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=062cd5a975ff7bd6fb91f9b4d1d9d102a7545499
---
Changelog | 2 +-
doc/general.texi | 1 +
libavformat/Makefile | 1 +
libavformat/allformats.c | 1 +
libavformat/dss.c | 342 ++++++++++++++++++++++++++++++++++++++++++++++
libavformat/version.h | 2 +-
6 files changed, 347 insertions(+), 2 deletions(-)
diff --git a/Changelog b/Changelog
index dea9435..641f048 100644
--- a/Changelog
+++ b/Changelog
@@ -15,7 +15,7 @@ version <next>:
- DXVA2-accelerated HEVC decoding
- AAC ELD 480 decoding
- Intel QSV-accelerated H.264 decoding
-- DSS SP decoder
+- DSS SP decoder and DSS demuxer
version 11:
diff --git a/doc/general.texi b/doc/general.texi
index 76b2f63..397a4b7 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -221,6 +221,7 @@ library:
@tab Used in the game Cyberia from Interplay.
@item Delphine Software International CIN @tab @tab X
@tab Multimedia format used by Delphine Software games.
+ at item Digital Speech Standard (DSS) @tab @tab X
@item CD+G @tab @tab X
@tab Video format used by CD+G karaoke disks
@item Commodore CDXL @tab @tab X
diff --git a/libavformat/Makefile b/libavformat/Makefile
index ea91a6c..edb50d7 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -100,6 +100,7 @@ OBJS-$(CONFIG_DIRAC_MUXER) += rawenc.o
OBJS-$(CONFIG_DNXHD_DEMUXER) += dnxhddec.o rawdec.o
OBJS-$(CONFIG_DNXHD_MUXER) += rawenc.o
OBJS-$(CONFIG_DSICIN_DEMUXER) += dsicin.o
+OBJS-$(CONFIG_DSS_DEMUXER) += dss.o
OBJS-$(CONFIG_DTS_DEMUXER) += dtsdec.o rawdec.o
OBJS-$(CONFIG_DTS_MUXER) += rawenc.o
OBJS-$(CONFIG_DV_DEMUXER) += dv.o
diff --git a/libavformat/allformats.c b/libavformat/allformats.c
index cb22ae3..f4be81a 100644
--- a/libavformat/allformats.c
+++ b/libavformat/allformats.c
@@ -94,6 +94,7 @@ void av_register_all(void)
REGISTER_MUXDEMUX(DIRAC, dirac);
REGISTER_MUXDEMUX(DNXHD, dnxhd);
REGISTER_DEMUXER (DSICIN, dsicin);
+ REGISTER_DEMUXER (DSS, dss);
REGISTER_MUXDEMUX(DTS, dts);
REGISTER_MUXDEMUX(DV, dv);
REGISTER_DEMUXER (DXA, dxa);
diff --git a/libavformat/dss.c b/libavformat/dss.c
new file mode 100644
index 0000000..f7d0ead
--- /dev/null
+++ b/libavformat/dss.c
@@ -0,0 +1,342 @@
+/*
+ * Digital Speech Standard (DSS) demuxer
+ * Copyright (c) 2014 Oleksij Rempel <linux at rempel-privat.de>
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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_HEADER_SIZE (DSS_BLOCK_SIZE * 2)
+#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;
+} DSSDemuxContext;
+
+static int dss_probe(AVProbeData *p)
+{
+ if (AV_RL32(p->buf) != MKTAG(0x2, '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;
+
+ sscanf(string, "%2d%2d%2d%2d%2d%2d", &y, &month, &d, &h, &minute, &sec);
+ /* 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;
+
+ st = avformat_new_stream(s, NULL);
+ if (!st)
+ return AVERROR(ENOMEM);
+
+ 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->codec->codec_id = AV_CODEC_ID_DSS_SP;
+ st->codec->sample_rate = 12000;
+ } else if (ctx->audio_codec == DSS_ACODEC_G723_1) {
+ st->codec->codec_id = AV_CODEC_ID_G723_1;
+ st->codec->sample_rate = 8000;
+ } else {
+ avpriv_request_sample(s, "Support for codec %x in DSS",
+ ctx->audio_codec);
+ return AVERROR_PATCHWELCOME;
+ }
+
+ st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codec->channel_layout = AV_CH_LAYOUT_MONO;
+ st->codec->channels = 1;
+
+ avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+ st->start_time = 0;
+
+ /* Jump over header */
+
+ if (avio_seek(pb, DSS_HEADER_SIZE, SEEK_SET) != 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;
+ int read_size, ret, offset = 0, buff_offset = 0;
+
+ if (ctx->counter == 0)
+ dss_skip_audio_header(s, pkt);
+
+ pkt->pos = avio_tell(s->pb);
+
+ if (ctx->swap) {
+ read_size = DSS_FRAME_SIZE - 2;
+ buff_offset = 3;
+ } else
+ read_size = DSS_FRAME_SIZE;
+
+ ctx->counter -= read_size;
+
+ ret = av_new_packet(pkt, DSS_FRAME_SIZE);
+ if (ret < 0)
+ return ret;
+
+ pkt->duration = 0;
+ pkt->stream_index = 0;
+
+ 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 (pkt->data[0] == 0xff)
+ return AVERROR_INVALIDDATA;
+
+ return pkt->size;
+
+error_eof:
+ av_free_packet(pkt);
+ return ret < 0 ? ret : AVERROR_EOF;
+}
+
+static int dss_723_1_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ DSSDemuxContext *ctx = s->priv_data;
+ int size, byte, ret, offset;
+
+ if (ctx->counter == 0)
+ dss_skip_audio_header(s, pkt);
+
+ pkt->pos = avio_tell(s->pb);
+ /* 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->counter -= size;
+
+ ret = av_new_packet(pkt, size);
+ if (ret < 0)
+ return ret;
+
+ pkt->data[0] = byte;
+ offset = 1;
+ pkt->duration = 240;
+
+ 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_free_packet(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_free_packet(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_free(ctx->dss_sp_buf);
+
+ 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,
+ .extensions = "dss"
+};
diff --git a/libavformat/version.h b/libavformat/version.h
index 8ec3a28..427565f 100644
--- a/libavformat/version.h
+++ b/libavformat/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFORMAT_VERSION_MAJOR 56
-#define LIBAVFORMAT_VERSION_MINOR 11
+#define LIBAVFORMAT_VERSION_MINOR 12
#define LIBAVFORMAT_VERSION_MICRO 0
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
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