ffmpeg/libavcodec/fastaudio.c

200 lines
5.9 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 "avcodec.h"
#include "bytestream.h"
#include "codec_internal.h"
#include "decode.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",
CODEC_LONG_NAME("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 },
};