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avformat: add LAF demuxer

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This commit is contained in:
Paul B Mahol
2022-09-11 20:10:27 +02:00
parent e91bc521af
commit dd2a01ef5c
6 changed files with 277 additions and 1 deletions
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1
Changelog
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@@ -13,6 +13,7 @@ version <next>:
- a3dscope filter
- bonk decoder and demuxer
- Micronas SC-4 audio decoder
- LAF demuxer
version 5.1:

2
doc/general_contents.texi
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@@ -510,6 +510,8 @@ library:
@tab A format used by libvpx
@item Internet Video Recording @tab @tab X
@item IRCAM @tab X @tab X
@item LAF @tab @tab X
@tab Limitless Audio Format
@item LATM @tab X @tab X
@item LMLM4 @tab @tab X
@tab Used by Linux Media Labs MPEG-4 PCI boards

1
libavformat/Makefile
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@@ -319,6 +319,7 @@ OBJS-$(CONFIG_JV_DEMUXER) += jvdec.o
OBJS-$(CONFIG_KUX_DEMUXER) += flvdec.o
OBJS-$(CONFIG_KVAG_DEMUXER) += kvag.o
OBJS-$(CONFIG_KVAG_MUXER) += kvag.o rawenc.o
OBJS-$(CONFIG_LAF_DEMUXER) += lafdec.o
OBJS-$(CONFIG_LATM_MUXER) += latmenc.o rawenc.o
OBJS-$(CONFIG_LMLM4_DEMUXER) += lmlm4.o
OBJS-$(CONFIG_LOAS_DEMUXER) += loasdec.o rawdec.o

1
libavformat/allformats.c
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@@ -236,6 +236,7 @@ extern const AVInputFormat ff_jv_demuxer;
extern const AVInputFormat ff_kux_demuxer;
extern const AVInputFormat ff_kvag_demuxer;
extern const AVOutputFormat ff_kvag_muxer;
extern const AVInputFormat ff_laf_demuxer;
extern const AVOutputFormat ff_latm_muxer;
extern const AVInputFormat ff_lmlm4_demuxer;
extern const AVInputFormat ff_loas_demuxer;

271
libavformat/lafdec.c Normal file
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@@ -0,0 +1,271 @@
/*
* Limitless Audio Format demuxer
* Copyright (c) 2022 Paul B Mahol
*
* 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/intreadwrite.h"
#include "avformat.h"
#include "internal.h"
#define MAX_STREAMS 4096
typedef struct StreamParams {
AVChannelLayout layout;
float horizontal;
float vertical;
int lfe;
int stored;
} StreamParams;
typedef struct LAFContext {
uint8_t *data;
unsigned nb_stored;
unsigned stored_index;
unsigned index;
unsigned bpp;
StreamParams p[MAX_STREAMS];
int header_len;
uint8_t header[(MAX_STREAMS + 7) / 8];
} LAFContext;
static int laf_probe(const AVProbeData *p)
{
if (memcmp(p->buf, "LIMITLESS", 9))
return 0;
if (memcmp(p->buf + 9, "HEAD", 4))
return 0;
return AVPROBE_SCORE_MAX;
}
static int laf_read_header(AVFormatContext *ctx)
{
LAFContext *s = ctx->priv_data;
AVIOContext *pb = ctx->pb;
unsigned st_count, mode;
unsigned sample_rate;
int64_t duration;
int codec_id;
int quality;
int bpp;
avio_skip(pb, 9);
if (avio_rb32(pb) != MKBETAG('H','E','A','D'))
return AVERROR_INVALIDDATA;
quality = avio_r8(pb);
if (quality > 3)
return AVERROR_INVALIDDATA;
mode = avio_r8(pb);
if (mode > 1)
return AVERROR_INVALIDDATA;
st_count = avio_rl32(pb);
if (st_count == 0 || st_count > MAX_STREAMS)
return AVERROR_INVALIDDATA;
for (int i = 0; i < st_count; i++) {
StreamParams *stp = &s->p[i];
stp->vertical = av_int2float(avio_rl32(pb));
stp->horizontal = av_int2float(avio_rl32(pb));
stp->lfe = avio_r8(pb);
if (stp->lfe) {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_LOW_FREQUENCY));
} else if (stp->vertical == 0.f &&
stp->horizontal == 0.f) {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_CENTER));
} else if (stp->vertical == 0.f &&
stp->horizontal == -30.f) {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_LEFT));
} else if (stp->vertical == 0.f &&
stp->horizontal == 30.f) {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_FRONT_RIGHT));
} else if (stp->vertical == 0.f &&
stp->horizontal == -110.f) {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_SIDE_LEFT));
} else if (stp->vertical == 0.f &&
stp->horizontal == 110.f) {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MASK(1, (AV_CH_SIDE_RIGHT));
} else {
stp->layout = (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO;
}
}
sample_rate = avio_rl32(pb);
duration = avio_rl64(pb) / st_count;
switch (quality) {
case 0:
codec_id = AV_CODEC_ID_PCM_U8;
bpp = 1;
break;
case 1:
codec_id = AV_CODEC_ID_PCM_S16LE;
bpp = 2;
break;
case 2:
codec_id = AV_CODEC_ID_PCM_F32LE;
bpp = 4;
break;
case 3:
codec_id = AV_CODEC_ID_PCM_S24LE;
bpp = 3;
break;
}
s->index = 0;
s->stored_index = 0;
s->bpp = bpp;
if ((int64_t)bpp * st_count * (int64_t)sample_rate >= INT32_MAX)
return AVERROR_INVALIDDATA;
s->data = av_calloc(st_count * sample_rate, bpp);
if (!s->data)
return AVERROR(ENOMEM);
for (int st = 0; st < st_count; st++) {
StreamParams *stp = &s->p[st];
AVCodecParameters *par;
AVStream *st = avformat_new_stream(ctx, NULL);
if (!st)
return AVERROR(ENOMEM);
par = st->codecpar;
par->codec_id = codec_id;
par->codec_type = AVMEDIA_TYPE_AUDIO;
par->ch_layout.nb_channels = 1;
par->ch_layout = stp->layout;
par->sample_rate = sample_rate;
st->duration = duration;
avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
}
s->header_len = (ctx->nb_streams + 7) / 8;
return 0;
}
static int laf_read_packet(AVFormatContext *ctx, AVPacket *pkt)
{
AVIOContext *pb = ctx->pb;
LAFContext *s = ctx->priv_data;
AVStream *st = ctx->streams[0];
const int bpp = s->bpp;
StreamParams *stp;
int64_t pos;
int ret;
pos = avio_tell(pb);
again:
if (avio_feof(pb))
return AVERROR_EOF;
if (s->index >= ctx->nb_streams) {
int cur_st = 0, st_count = 0, st_index = 0;
avio_read(pb, s->header, s->header_len);
for (int i = 0; i < s->header_len; i++) {
uint8_t val = s->header[i];
for (int j = 0; j < 8 && cur_st < ctx->nb_streams; j++, cur_st++) {
StreamParams *stp = &s->p[st_index];
stp->stored = 0;
if (val & 1) {
stp->stored = 1;
st_count++;
}
val >>= 1;
st_index++;
}
}
s->index = s->stored_index = 0;
s->nb_stored = st_count;
if (!st_count)
return AVERROR_INVALIDDATA;
ret = avio_read(pb, s->data, st_count * st->codecpar->sample_rate * bpp);
if (ret < 0)
return ret;
}
st = ctx->streams[s->index];
stp = &s->p[s->index];
while (!stp->stored) {
s->index++;
if (s->index >= ctx->nb_streams)
goto again;
stp = &s->p[s->index];
}
st = ctx->streams[s->index];
ret = av_new_packet(pkt, st->codecpar->sample_rate * bpp);
if (ret < 0)
return ret;
switch (bpp) {
case 1:
for (int n = 0; n < st->codecpar->sample_rate; n++)
pkt->data[n] = s->data[n * s->nb_stored + s->stored_index];
break;
case 2:
for (int n = 0; n < st->codecpar->sample_rate; n++)
AV_WN16(pkt->data + n * 2, AV_RN16(s->data + n * s->nb_stored * 2 + s->stored_index * 2));
break;
case 3:
for (int n = 0; n < st->codecpar->sample_rate; n++)
AV_WL24(pkt->data + n * 3, AV_RL24(s->data + n * s->nb_stored * 3 + s->stored_index * 3));
break;
case 4:
for (int n = 0; n < st->codecpar->sample_rate; n++)
AV_WN32(pkt->data + n * 4, AV_RN32(s->data + n * s->nb_stored * 4 + s->stored_index * 4));
break;
}
pkt->stream_index = s->index;
pkt->pos = pos;
s->index++;
s->stored_index++;
return 0;
}
static int laf_read_seek(AVFormatContext *ctx, int stream_index,
int64_t timestamp, int flags)
{
LAFContext *s = ctx->priv_data;
s->stored_index = s->index = s->nb_stored = 0;
return -1;
}
const AVInputFormat ff_laf_demuxer = {
.name = "laf",
.long_name = NULL_IF_CONFIG_SMALL("LAF (Limitless Audio Format)"),
.priv_data_size = sizeof(LAFContext),
.read_probe = laf_probe,
.read_header = laf_read_header,
.read_packet = laf_read_packet,
.read_seek = laf_read_seek,
.extensions = "laf",
.flags = AVFMT_GENERIC_INDEX,
};

2
libavformat/version.h
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@@ -31,7 +31,7 @@
#include "version_major.h"
#define LIBAVFORMAT_VERSION_MINOR 31
#define LIBAVFORMAT_VERSION_MINOR 32
#define LIBAVFORMAT_VERSION_MICRO 100
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \