/*
* 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"
};