ffmpeg/libavcodec/ac3enc_float.c
Anton Khirnov 8d73f3ce56 lavc: support AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE in all no-delay encoders
Including fake-delay encoders marked with FF_CODEC_CAP_EOF_FLUSH.
2023-01-29 09:22:57 +01:00

133 lines
4.2 KiB
C

/*
* The simplest AC-3 encoder
* Copyright (c) 2000 Fabrice Bellard
* Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
*
* 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
* floating-point AC-3 encoder.
*/
#define AC3ENC_FLOAT 1
#include "audiodsp.h"
#include "ac3enc.h"
#include "codec_internal.h"
#include "eac3enc.h"
#include "kbdwin.h"
/*
* Scale MDCT coefficients from float to 24-bit fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
{
int chan_size = AC3_MAX_COEFS * s->num_blocks;
int cpl = s->cpl_on;
s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer + (chan_size * !cpl),
s->mdct_coef_buffer + (chan_size * !cpl),
chan_size * (s->channels + cpl));
}
/*
* Clip MDCT coefficients to allowable range.
*/
static void clip_coefficients(AudioDSPContext *adsp, float *coef,
unsigned int len)
{
adsp->vector_clipf(coef, coef, len, COEF_MIN, COEF_MAX);
}
/*
* Calculate a single coupling coordinate.
*/
static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
{
float coord = 0.125;
if (energy_cpl > 0)
coord *= sqrtf(energy_ch / energy_cpl);
return FFMIN(coord, COEF_MAX);
}
static void sum_square_butterfly(AC3EncodeContext *s, float sum[4],
const float *coef0, const float *coef1,
int len)
{
s->ac3dsp.sum_square_butterfly_float(sum, coef0, coef1, len);
}
#include "ac3enc_template.c"
/**
* Initialize MDCT tables.
*
* @param s AC-3 encoder private context
* @return 0 on success, negative error code on failure
*/
static av_cold int ac3_float_mdct_init(AC3EncodeContext *s)
{
const float scale = -2.0 / AC3_WINDOW_SIZE;
float *window = av_malloc_array(AC3_BLOCK_SIZE, sizeof(*window));
if (!window) {
av_log(s->avctx, AV_LOG_ERROR, "Cannot allocate memory.\n");
return AVERROR(ENOMEM);
}
ff_kbd_window_init(window, 5.0, AC3_BLOCK_SIZE);
s->mdct_window = window;
return av_tx_init(&s->tx, &s->tx_fn, AV_TX_FLOAT_MDCT, 0,
AC3_BLOCK_SIZE, &scale, 0);
}
av_cold int ff_ac3_float_encode_init(AVCodecContext *avctx)
{
AC3EncodeContext *s = avctx->priv_data;
s->mdct_init = ac3_float_mdct_init;
s->allocate_sample_buffers = allocate_sample_buffers;
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
if (!s->fdsp)
return AVERROR(ENOMEM);
return ff_ac3_encode_init(avctx);
}
const FFCodec ff_ac3_encoder = {
.p.name = "ac3",
CODEC_LONG_NAME("ATSC A/52A (AC-3)"),
.p.type = AVMEDIA_TYPE_AUDIO,
.p.id = AV_CODEC_ID_AC3,
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ff_ac3_float_encode_init,
FF_CODEC_ENCODE_CB(ff_ac3_float_encode_frame),
.close = ff_ac3_encode_close,
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.p.priv_class = &ff_ac3enc_class,
.p.supported_samplerates = ff_ac3_sample_rate_tab,
CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(ff_ac3_channel_layouts)
.p.ch_layouts = ff_ac3_ch_layouts,
.defaults = ff_ac3_enc_defaults,
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
};