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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-07 11:13:41 +02:00
FFmpeg/libavcodec/adxenc.c
Andreas Rheinhardt 21b23ceab3 avcodec: Make init-threadsafety the default
and remove FF_CODEC_CAP_INIT_THREADSAFE
All our native codecs are already init-threadsafe
(only wrappers for external libraries and hwaccels
are typically not marked as init-threadsafe yet),
so it is only natural for this to also be the default state.

Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-07-18 20:04:59 +02:00

204 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 },
};