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https://github.com/FFmpeg/FFmpeg.git
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cd335de53f
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>
133 lines
3.6 KiB
C
133 lines
3.6 KiB
C
/*
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* DFPWM decoder
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* Copyright (c) 2022 Jack Bruienne
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* Copyright (c) 2012, 2016 Ben "GreaseMonkey" Russell
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* DFPWM1a decoder
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*/
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#include "libavutil/internal.h"
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#include "avcodec.h"
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#include "codec_id.h"
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#include "codec_internal.h"
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#include "decode.h"
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typedef struct {
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int fq, q, s, lt;
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} DFPWMState;
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// DFPWM codec from https://github.com/ChenThread/dfpwm/blob/master/1a/
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// Licensed in the public domain
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static void au_decompress(DFPWMState *state, int fs, int len,
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uint8_t *outbuf, const uint8_t *inbuf)
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{
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unsigned d;
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for (int i = 0; i < len; i++) {
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// get bits
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d = *(inbuf++);
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for (int j = 0; j < 8; j++) {
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int nq, lq, st, ns, ov;
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// set target
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int t = ((d&1) ? 127 : -128);
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d >>= 1;
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// adjust charge
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nq = state->q + ((state->s * (t-state->q) + 512)>>10);
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if(nq == state->q && nq != t)
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nq += (t == 127 ? 1 : -1);
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lq = state->q;
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state->q = nq;
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// adjust strength
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st = (t != state->lt ? 0 : 1023);
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ns = state->s;
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if(ns != st)
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ns += (st != 0 ? 1 : -1);
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if(ns < 8) ns = 8;
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state->s = ns;
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// FILTER: perform antijerk
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ov = (t != state->lt ? (nq+lq+1)>>1 : nq);
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// FILTER: perform LPF
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state->fq += ((fs*(ov-state->fq) + 0x80)>>8);
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ov = state->fq;
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// output sample
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*(outbuf++) = ov + 128;
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state->lt = t;
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}
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}
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}
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static av_cold int dfpwm_dec_init(struct AVCodecContext *ctx)
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{
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DFPWMState *state = ctx->priv_data;
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state->fq = 0;
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state->q = 0;
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state->s = 0;
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state->lt = -128;
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ctx->sample_fmt = AV_SAMPLE_FMT_U8;
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ctx->bits_per_raw_sample = 8;
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return 0;
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}
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static int dfpwm_dec_frame(struct AVCodecContext *ctx, AVFrame *frame,
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int *got_frame, struct AVPacket *packet)
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{
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DFPWMState *state = ctx->priv_data;
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int ret;
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if (packet->size * 8LL % ctx->ch_layout.nb_channels)
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return AVERROR_PATCHWELCOME;
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frame->nb_samples = packet->size * 8LL / ctx->ch_layout.nb_channels;
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if (frame->nb_samples <= 0) {
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av_log(ctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
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return AVERROR_INVALIDDATA;
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}
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if ((ret = ff_get_buffer(ctx, frame, 0)) < 0)
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return ret;
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au_decompress(state, 140, packet->size, frame->data[0], packet->data);
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*got_frame = 1;
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return packet->size;
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}
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const FFCodec ff_dfpwm_decoder = {
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.p.name = "dfpwm",
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CODEC_LONG_NAME("DFPWM1a audio"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_DFPWM,
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.priv_data_size = sizeof(DFPWMState),
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.init = dfpwm_dec_init,
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FF_CODEC_DECODE_CB(dfpwm_dec_frame),
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.p.capabilities = AV_CODEC_CAP_DR1,
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};
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