mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-28 20:53:54 +02:00
21b23ceab3
and remove FF_CODEC_CAP_INIT_THREADSAFE All our native codecs are already init-threadsafe (only wrappers for external libraries and hwaccels are typically not marked as init-threadsafe yet), so it is only natural for this to also be the default state. Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
203 lines
6.0 KiB
C
203 lines
6.0 KiB
C
/*
|
|
* MOFLEX Fast Audio decoder
|
|
* Copyright (c) 2015-2016 Florian Nouwt
|
|
* Copyright (c) 2017 Adib Surani
|
|
* Copyright (c) 2020 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 "avcodec.h"
|
|
#include "bytestream.h"
|
|
#include "codec_internal.h"
|
|
#include "internal.h"
|
|
#include "mathops.h"
|
|
|
|
typedef struct ChannelItems {
|
|
float f[8];
|
|
float last;
|
|
} ChannelItems;
|
|
|
|
typedef struct FastAudioContext {
|
|
float table[8][64];
|
|
|
|
ChannelItems *ch;
|
|
} FastAudioContext;
|
|
|
|
static av_cold int fastaudio_init(AVCodecContext *avctx)
|
|
{
|
|
FastAudioContext *s = avctx->priv_data;
|
|
|
|
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
|
|
|
|
for (int i = 0; i < 8; i++)
|
|
s->table[0][i] = (i - 159.5f) / 160.f;
|
|
for (int i = 0; i < 11; i++)
|
|
s->table[0][i + 8] = (i - 37.5f) / 40.f;
|
|
for (int i = 0; i < 27; i++)
|
|
s->table[0][i + 8 + 11] = (i - 13.f) / 20.f;
|
|
for (int i = 0; i < 11; i++)
|
|
s->table[0][i + 8 + 11 + 27] = (i + 27.5f) / 40.f;
|
|
for (int i = 0; i < 7; i++)
|
|
s->table[0][i + 8 + 11 + 27 + 11] = (i + 152.5f) / 160.f;
|
|
|
|
memcpy(s->table[1], s->table[0], sizeof(s->table[0]));
|
|
|
|
for (int i = 0; i < 7; i++)
|
|
s->table[2][i] = (i - 33.5f) / 40.f;
|
|
for (int i = 0; i < 25; i++)
|
|
s->table[2][i + 7] = (i - 13.f) / 20.f;
|
|
|
|
for (int i = 0; i < 32; i++)
|
|
s->table[3][i] = -s->table[2][31 - i];
|
|
|
|
for (int i = 0; i < 16; i++)
|
|
s->table[4][i] = i * 0.22f / 3.f - 0.6f;
|
|
|
|
for (int i = 0; i < 16; i++)
|
|
s->table[5][i] = i * 0.20f / 3.f - 0.3f;
|
|
|
|
for (int i = 0; i < 8; i++)
|
|
s->table[6][i] = i * 0.36f / 3.f - 0.4f;
|
|
|
|
for (int i = 0; i < 8; i++)
|
|
s->table[7][i] = i * 0.34f / 3.f - 0.2f;
|
|
|
|
s->ch = av_calloc(avctx->ch_layout.nb_channels, sizeof(*s->ch));
|
|
if (!s->ch)
|
|
return AVERROR(ENOMEM);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int read_bits(int bits, int *ppos, unsigned *src)
|
|
{
|
|
int r, pos;
|
|
|
|
pos = *ppos;
|
|
pos += bits;
|
|
r = src[(pos - 1) / 32] >> ((-pos) & 31);
|
|
*ppos = pos;
|
|
|
|
return r & ((1 << bits) - 1);
|
|
}
|
|
|
|
static const uint8_t bits[8] = { 6, 6, 5, 5, 4, 0, 3, 3, };
|
|
|
|
static void set_sample(int i, int j, int v, float *result, int *pads, float value)
|
|
{
|
|
result[i * 64 + pads[i] + j * 3] = value * (2 * v - 7);
|
|
}
|
|
|
|
static int fastaudio_decode(AVCodecContext *avctx, AVFrame *frame,
|
|
int *got_frame, AVPacket *pkt)
|
|
{
|
|
FastAudioContext *s = avctx->priv_data;
|
|
GetByteContext gb;
|
|
int subframes;
|
|
int ret;
|
|
|
|
subframes = pkt->size / (40 * avctx->ch_layout.nb_channels);
|
|
frame->nb_samples = subframes * 256;
|
|
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
|
|
return ret;
|
|
|
|
bytestream2_init(&gb, pkt->data, pkt->size);
|
|
|
|
for (int subframe = 0; subframe < subframes; subframe++) {
|
|
for (int channel = 0; channel < avctx->ch_layout.nb_channels; channel++) {
|
|
ChannelItems *ch = &s->ch[channel];
|
|
float result[256] = { 0 };
|
|
unsigned src[10];
|
|
int inds[4], pads[4];
|
|
float m[8];
|
|
int pos = 0;
|
|
|
|
for (int i = 0; i < 10; i++)
|
|
src[i] = bytestream2_get_le32(&gb);
|
|
|
|
for (int i = 0; i < 8; i++)
|
|
m[7 - i] = s->table[i][read_bits(bits[i], &pos, src)];
|
|
|
|
for (int i = 0; i < 4; i++)
|
|
inds[3 - i] = read_bits(6, &pos, src);
|
|
|
|
for (int i = 0; i < 4; i++)
|
|
pads[3 - i] = read_bits(2, &pos, src);
|
|
|
|
for (int i = 0, index5 = 0; i < 4; i++) {
|
|
float value = av_int2float((inds[i] + 1) << 20) * powf(2.f, 116.f);
|
|
|
|
for (int j = 0, tmp = 0; j < 21; j++) {
|
|
set_sample(i, j, j == 20 ? tmp / 2 : read_bits(3, &pos, src), result, pads, value);
|
|
if (j % 10 == 9)
|
|
tmp = 4 * tmp + read_bits(2, &pos, src);
|
|
if (j == 20)
|
|
index5 = FFMIN(2 * index5 + tmp % 2, 63);
|
|
}
|
|
|
|
m[2] = s->table[5][index5];
|
|
}
|
|
|
|
for (int i = 0; i < 256; i++) {
|
|
float x = result[i];
|
|
|
|
for (int j = 0; j < 8; j++) {
|
|
x -= m[j] * ch->f[j];
|
|
ch->f[j] += m[j] * x;
|
|
}
|
|
|
|
memmove(&ch->f[0], &ch->f[1], sizeof(float) * 7);
|
|
ch->f[7] = x;
|
|
ch->last = x + ch->last * 0.86f;
|
|
result[i] = ch->last * 2.f;
|
|
}
|
|
|
|
memcpy(frame->extended_data[channel] + 1024 * subframe, result, 256 * sizeof(float));
|
|
}
|
|
}
|
|
|
|
*got_frame = 1;
|
|
|
|
return pkt->size;
|
|
}
|
|
|
|
static av_cold int fastaudio_close(AVCodecContext *avctx)
|
|
{
|
|
FastAudioContext *s = avctx->priv_data;
|
|
|
|
av_freep(&s->ch);
|
|
|
|
return 0;
|
|
}
|
|
|
|
const FFCodec ff_fastaudio_decoder = {
|
|
.p.name = "fastaudio",
|
|
.p.long_name = NULL_IF_CONFIG_SMALL("MobiClip FastAudio"),
|
|
.p.type = AVMEDIA_TYPE_AUDIO,
|
|
.p.id = AV_CODEC_ID_FASTAUDIO,
|
|
.priv_data_size = sizeof(FastAudioContext),
|
|
.init = fastaudio_init,
|
|
FF_CODEC_DECODE_CB(fastaudio_decode),
|
|
.close = fastaudio_close,
|
|
.p.capabilities = AV_CODEC_CAP_DR1,
|
|
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
};
|