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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavformat/lafdec.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

295 lines
8.2 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 "libavutil/mem.h"
#include "avformat.h"
#include "avio_internal.h"
#include "demux.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 ||
(int64_t)bpp * st_count * (int64_t)sample_rate == 0
)
return AVERROR_INVALIDDATA;
s->data = av_calloc(st_count * sample_rate, bpp);
if (!s->data)
return AVERROR(ENOMEM);
for (unsigned i = 0; i < st_count; i++) {
StreamParams *stp = &s->p[i];
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 FFInputFormat ff_laf_demuxer = {
.p.name = "laf",
.p.long_name = NULL_IF_CONFIG_SMALL("LAF (Limitless Audio Format)"),
.p.extensions = "laf",
.p.flags = AVFMT_GENERIC_INDEX,
.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,
.flags_internal = FF_INFMT_FLAG_INIT_CLEANUP,
};