RepoMirrors/ffmpeg
1
0
Fork 0
mirror of https://git.ffmpeg.org/ffmpeg.git synced 2024-12-27 18:02:11 +00:00
Code Issues Projects Releases Wiki Activity
42cbf66fff
ffmpeg/libavformat/dss.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

368 lines
10 KiB
C
Raw Blame History

/*
* 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/channel_layout.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/mem.h"
#include "avformat.h"
#include "demux.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;
int packet_size;
int dss_header_size;
} DSSDemuxContext;
static int dss_probe(const 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) {
av_free(value);
return ret < 0 ? ret : AVERROR_EOF;
}
return av_dict_set(&s->metadata, key, value, AV_DICT_DONT_STRDUP_VAL);
}
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;
s->bit_rate = 8 * (DSS_FRAME_SIZE - 1) * st->codecpar->sample_rate
* 512 / (506 * 264);
} 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->ch_layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
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;
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 *data)
{
int i;
if (ctx->swap) {
for (i = 0; i < DSS_FRAME_SIZE - 2; i += 2)
data[i] = data[i + 4];
/* Zero the padding. */
data[DSS_FRAME_SIZE] = 0;
data[1] = ctx->dss_sp_swap_byte;
} else {
ctx->dss_sp_swap_byte = data[DSS_FRAME_SIZE - 2];
}
/* make sure byte 40 is always 0 */
data[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;
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;
ret = av_new_packet(pkt, DSS_FRAME_SIZE);
if (ret < 0)
return ret;
pkt->duration = 264;
pkt->pos = pos;
pkt->stream_index = 0;
if (ctx->counter < read_size) {
ret = avio_read(s->pb, pkt->data + buff_offset,
ctx->counter);
if (ret < ctx->counter)
goto error_eof;
offset = ctx->counter;
dss_skip_audio_header(s, pkt);
}
ctx->counter -= read_size;
/* This will write one byte into pkt's padding if buff_offset == 3 */
ret = avio_read(s->pb, pkt->data + offset + buff_offset,
read_size - offset);
if (ret < read_size - offset)
goto error_eof;
dss_sp_byte_swap(ctx, pkt->data);
if (ctx->dss_sp_swap_byte < 0) {
return AVERROR(EAGAIN);
}
return 0;
error_eof:
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--;
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 < size) {
ret = avio_read(s->pb, pkt->data + offset,
ctx->counter);
if (ret < ctx->counter)
return ret < 0 ? ret : AVERROR_EOF;
offset += ctx->counter;
size -= ctx->counter;
ctx->counter = 0;
dss_skip_audio_header(s, pkt);
}
ctx->counter -= size;
ret = avio_read(s->pb, pkt->data + offset, size);
if (ret < size)
return ret < 0 ? ret : AVERROR_EOF;
return 0;
}
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_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;
}
const FFInputFormat ff_dss_demuxer = {
.p.name = "dss",
.p.long_name = NULL_IF_CONFIG_SMALL("Digital Speech Standard (DSS)"),
.p.extensions = "dss",
.priv_data_size = sizeof(DSSDemuxContext),
.read_probe = dss_probe,
.read_header = dss_read_header,
.read_packet = dss_read_packet,
.read_seek = dss_read_seek,
};
Reference in New Issue View Git Blame Copy Permalink