mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-11-26 19:01:44 +02:00
291 lines
8.1 KiB
C
291 lines
8.1 KiB
C
/*
|
|
* 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 "avio_internal.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;
|
|
|
|
if (avio_feof(pb))
|
|
return AVERROR_INVALIDDATA;
|
|
|
|
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;
|
|
default:
|
|
return AVERROR_INVALIDDATA;
|
|
}
|
|
|
|
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;
|
|
|
|
ret = ffio_read_size(pb, s->header, s->header_len);
|
|
if (ret < 0)
|
|
return ret;
|
|
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 = ffio_read_size(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_close(AVFormatContext *ctx)
|
|
{
|
|
LAFContext *s = ctx->priv_data;
|
|
|
|
av_freep(&s->data);
|
|
|
|
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_close = laf_read_close,
|
|
.read_seek = laf_read_seek,
|
|
.extensions = "laf",
|
|
.flags = AVFMT_GENERIC_INDEX,
|
|
.flags_internal = FF_FMT_INIT_CLEANUP,
|
|
};
|