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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/dfpwmdec.c
Andreas Rheinhardt cd335de53f avcodec/dfpwmdec: Remove always-false check
This decoder does not have the AV_CODEC_CAP_CHANNEL_CONF set,
so that number of channels has to be set by the user before
avcodec_open2().

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-09-22 01:05:59 +02:00

133 lines
3.6 KiB
C

/*
* DFPWM decoder
* Copyright (c) 2022 Jack Bruienne
* Copyright (c) 2012, 2016 Ben "GreaseMonkey" Russell
*
* 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
*/
/**
* @file
* DFPWM1a decoder
*/
#include "libavutil/internal.h"
#include "avcodec.h"
#include "codec_id.h"
#include "codec_internal.h"
#include "decode.h"
typedef struct {
int fq, q, s, lt;
} DFPWMState;
// DFPWM codec from https://github.com/ChenThread/dfpwm/blob/master/1a/
// Licensed in the public domain
static void au_decompress(DFPWMState *state, int fs, int len,
uint8_t *outbuf, const uint8_t *inbuf)
{
unsigned d;
for (int i = 0; i < len; i++) {
// get bits
d = *(inbuf++);
for (int j = 0; j < 8; j++) {
int nq, lq, st, ns, ov;
// set target
int t = ((d&1) ? 127 : -128);
d >>= 1;
// adjust charge
nq = state->q + ((state->s * (t-state->q) + 512)>>10);
if(nq == state->q && nq != t)
nq += (t == 127 ? 1 : -1);
lq = state->q;
state->q = nq;
// adjust strength
st = (t != state->lt ? 0 : 1023);
ns = state->s;
if(ns != st)
ns += (st != 0 ? 1 : -1);
if(ns < 8) ns = 8;
state->s = ns;
// FILTER: perform antijerk
ov = (t != state->lt ? (nq+lq+1)>>1 : nq);
// FILTER: perform LPF
state->fq += ((fs*(ov-state->fq) + 0x80)>>8);
ov = state->fq;
// output sample
*(outbuf++) = ov + 128;
state->lt = t;
}
}
}
static av_cold int dfpwm_dec_init(struct AVCodecContext *ctx)
{
DFPWMState *state = ctx->priv_data;
state->fq = 0;
state->q = 0;
state->s = 0;
state->lt = -128;
ctx->sample_fmt = AV_SAMPLE_FMT_U8;
ctx->bits_per_raw_sample = 8;
return 0;
}
static int dfpwm_dec_frame(struct AVCodecContext *ctx, AVFrame *frame,
int *got_frame, struct AVPacket *packet)
{
DFPWMState *state = ctx->priv_data;
int ret;
if (packet->size * 8LL % ctx->ch_layout.nb_channels)
return AVERROR_PATCHWELCOME;
frame->nb_samples = packet->size * 8LL / ctx->ch_layout.nb_channels;
if (frame->nb_samples <= 0) {
av_log(ctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
return AVERROR_INVALIDDATA;
}
if ((ret = ff_get_buffer(ctx, frame, 0)) < 0)
return ret;
au_decompress(state, 140, packet->size, frame->data[0], packet->data);
*got_frame = 1;
return packet->size;
}
const FFCodec ff_dfpwm_decoder = {
.p.name = "dfpwm",
CODEC_LONG_NAME("DFPWM1a audio"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_DFPWM,
.priv_data_size = sizeof(DFPWMState),
.init = dfpwm_dec_init,
FF_CODEC_DECODE_CB(dfpwm_dec_frame),
.p.capabilities = AV_CODEC_CAP_DR1,
};