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This is possible, because every given FFCodec has to implement exactly one of these. Doing so decreases sizeof(FFCodec) and therefore decreases the size of the binary. Notice that in case of position-independent code the decrease is in .data.rel.ro, so that this translates to decreased memory consumption. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
138 lines
3.8 KiB
C
138 lines
3.8 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 "internal.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, uint8_t *outbuf, 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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if (ctx->ch_layout.nb_channels <= 0) {
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av_log(ctx, AV_LOG_ERROR, "Invalid number of channels\n");
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return AVERROR(EINVAL);
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}
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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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.p.long_name = NULL_IF_CONFIG_SMALL("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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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
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};
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