1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavcodec/adxenc.c
Andreas Rheinhardt 4243da4ff4 avcodec/codec_internal: Use union for FFCodec decode/encode callbacks
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>
2022-04-05 20:02:37 +02:00

205 lines
6.2 KiB
C

/*
* ADX ADPCM codecs
* Copyright (c) 2001,2003 BERO
*
* 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 "adx.h"
#include "bytestream.h"
#include "codec_internal.h"
#include "encode.h"
#include "put_bits.h"
/**
* @file
* SEGA CRI adx codecs.
*
* Reference documents:
* http://ku-www.ss.titech.ac.jp/~yatsushi/adx.html
* adx2wav & wav2adx http://www.geocities.co.jp/Playtown/2004/
*/
static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
ADXChannelState *prev, int channels)
{
PutBitContext pb;
int scale;
int i, j;
int s0, s1, s2, d;
int max = 0;
int min = 0;
s1 = prev->s1;
s2 = prev->s2;
for (i = 0, j = 0; j < 32; i += channels, j++) {
s0 = wav[i];
d = s0 + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
if (max < d)
max = d;
if (min > d)
min = d;
s2 = s1;
s1 = s0;
}
if (max == 0 && min == 0) {
prev->s1 = s1;
prev->s2 = s2;
memset(adx, 0, BLOCK_SIZE);
return;
}
if (max / 7 > -min / 8)
scale = max / 7;
else
scale = -min / 8;
if (scale == 0)
scale = 1;
AV_WB16(adx, scale);
init_put_bits(&pb, adx + 2, 16);
s1 = prev->s1;
s2 = prev->s2;
for (i = 0, j = 0; j < 32; i += channels, j++) {
d = wav[i] + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
put_sbits(&pb, 4, d);
s0 = d * scale + ((c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS);
s2 = s1;
s1 = s0;
}
prev->s1 = s1;
prev->s2 = s2;
flush_put_bits(&pb);
}
#define HEADER_SIZE 36
static int adx_encode_header(AVCodecContext *avctx, uint8_t *buf, int bufsize)
{
ADXContext *c = avctx->priv_data;
bytestream_put_be16(&buf, 0x8000); /* header signature */
bytestream_put_be16(&buf, HEADER_SIZE - 4); /* copyright offset */
bytestream_put_byte(&buf, 3); /* encoding */
bytestream_put_byte(&buf, BLOCK_SIZE); /* block size */
bytestream_put_byte(&buf, 4); /* sample size */
bytestream_put_byte(&buf, avctx->ch_layout.nb_channels); /* channels */
bytestream_put_be32(&buf, avctx->sample_rate); /* sample rate */
bytestream_put_be32(&buf, 0); /* total sample count */
bytestream_put_be16(&buf, c->cutoff); /* cutoff frequency */
bytestream_put_byte(&buf, 3); /* version */
bytestream_put_byte(&buf, 0); /* flags */
bytestream_put_be32(&buf, 0); /* unknown */
bytestream_put_be32(&buf, 0); /* loop enabled */
bytestream_put_be16(&buf, 0); /* padding */
bytestream_put_buffer(&buf, "(c)CRI", 6); /* copyright signature */
return HEADER_SIZE;
}
static av_cold int adx_encode_init(AVCodecContext *avctx)
{
ADXContext *c = avctx->priv_data;
if (avctx->ch_layout.nb_channels > 2) {
av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
return AVERROR(EINVAL);
}
avctx->frame_size = BLOCK_SAMPLES;
/* the cutoff can be adjusted, but this seems to work pretty well */
c->cutoff = 500;
ff_adx_calculate_coeffs(c->cutoff, avctx->sample_rate, COEFF_BITS, c->coeff);
return 0;
}
static int adx_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
ADXContext *c = avctx->priv_data;
const int16_t *samples = frame ? (const int16_t *)frame->data[0] : NULL;
uint8_t *dst;
int channels = avctx->ch_layout.nb_channels;
int ch, out_size, ret;
if (!samples) {
if (c->eof)
return 0;
if ((ret = ff_get_encode_buffer(avctx, avpkt, 18, 0)) < 0)
return ret;
c->eof = 1;
dst = avpkt->data;
bytestream_put_be16(&dst, 0x8001);
bytestream_put_be16(&dst, 0x000E);
bytestream_put_be64(&dst, 0x0);
bytestream_put_be32(&dst, 0x0);
bytestream_put_be16(&dst, 0x0);
*got_packet_ptr = 1;
return 0;
}
out_size = BLOCK_SIZE * channels + !c->header_parsed * HEADER_SIZE;
if ((ret = ff_get_encode_buffer(avctx, avpkt, out_size, 0)) < 0)
return ret;
dst = avpkt->data;
if (!c->header_parsed) {
int hdrsize;
if ((hdrsize = adx_encode_header(avctx, dst, avpkt->size)) < 0) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
dst += hdrsize;
c->header_parsed = 1;
}
for (ch = 0; ch < channels; ch++) {
adx_encode(c, dst, samples + ch, &c->prev[ch], channels);
dst += BLOCK_SIZE;
}
avpkt->pts = frame->pts;
avpkt->duration = frame->nb_samples;
*got_packet_ptr = 1;
return 0;
}
const FFCodec ff_adpcm_adx_encoder = {
.p.name = "adpcm_adx",
.p.long_name = NULL_IF_CONFIG_SMALL("SEGA CRI ADX ADPCM"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_ADPCM_ADX,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY,
.priv_data_size = sizeof(ADXContext),
.init = adx_encode_init,
FF_CODEC_ENCODE_CB(adx_encode_frame),
.p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};