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https://github.com/FFmpeg/FFmpeg.git
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211619ad7f
Since the removal of the 16-bit FFT said define is unnecessary as FFT_FIXED_32 is always !FFT_FLOAT. But one wouldn't believe it when looking at the code. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
140 lines
4.1 KiB
C
140 lines
4.1 KiB
C
/*
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* The simplest AC-3 encoder
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* Copyright (c) 2000 Fabrice Bellard
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* Copyright (c) 2006-2010 Justin Ruggles <justin.ruggles@gmail.com>
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* Copyright (c) 2006-2010 Prakash Punnoor <prakash@punnoor.de>
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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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* fixed-point AC-3 encoder.
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*/
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#define AC3ENC_FLOAT 0
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#define FFT_FLOAT 0
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#include "internal.h"
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#include "audiodsp.h"
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#include "ac3enc.h"
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#include "eac3enc.h"
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#include "kbdwin.h"
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static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4],
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const int32_t *coef0, const int32_t *coef1,
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int len)
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{
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s->ac3dsp.sum_square_butterfly_int32(sum, coef0, coef1, len);
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}
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/*
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* Clip MDCT coefficients to allowable range.
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*/
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static void clip_coefficients(AudioDSPContext *adsp, int32_t *coef,
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unsigned int len)
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{
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adsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len);
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}
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/*
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* Calculate a single coupling coordinate.
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*/
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static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
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{
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if (energy_cpl <= COEF_MAX) {
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return 1048576;
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} else {
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uint64_t coord = energy_ch / (energy_cpl >> 24);
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uint32_t coord32 = FFMIN(coord, 1073741824);
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coord32 = ff_sqrt(coord32) << 9;
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return FFMIN(coord32, COEF_MAX);
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}
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}
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#include "ac3enc_template.c"
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/**
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* Finalize MDCT and free allocated memory.
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*
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* @param s AC-3 encoder private context
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*/
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static av_cold void ac3_fixed_mdct_end(AC3EncodeContext *s)
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{
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ff_mdct_end(&s->mdct);
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}
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/**
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* Initialize MDCT tables.
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*
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* @param s AC-3 encoder private context
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* @return 0 on success, negative error code on failure
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*/
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static av_cold int ac3_fixed_mdct_init(AC3EncodeContext *s)
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{
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float fwin[AC3_BLOCK_SIZE];
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int32_t *iwin = av_malloc_array(AC3_BLOCK_SIZE, sizeof(*iwin));
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if (!iwin)
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return AVERROR(ENOMEM);
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ff_kbd_window_init(fwin, 5.0, AC3_BLOCK_SIZE);
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for (int i = 0; i < AC3_BLOCK_SIZE; i++)
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iwin[i] = lrintf(fwin[i] * (1 << 22));
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s->mdct_window = iwin;
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s->fdsp = avpriv_alloc_fixed_dsp(s->avctx->flags & AV_CODEC_FLAG_BITEXACT);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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return ff_mdct_init(&s->mdct, 9, 0, -1.0);
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}
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static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
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{
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AC3EncodeContext *s = avctx->priv_data;
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s->fixed_point = 1;
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s->mdct_end = ac3_fixed_mdct_end;
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s->mdct_init = ac3_fixed_mdct_init;
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s->allocate_sample_buffers = allocate_sample_buffers;
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return ff_ac3_encode_init(avctx);
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}
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const AVCodec ff_ac3_fixed_encoder = {
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.name = "ac3_fixed",
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_AC3,
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.capabilities = AV_CODEC_CAP_DR1,
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.priv_data_size = sizeof(AC3EncodeContext),
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.init = ac3_fixed_encode_init,
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.encode2 = ff_ac3_fixed_encode_frame,
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.close = ff_ac3_encode_close,
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.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S32P,
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AV_SAMPLE_FMT_NONE },
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.priv_class = &ff_ac3enc_class,
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
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.supported_samplerates = ff_ac3_sample_rate_tab,
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.channel_layouts = ff_ac3_channel_layouts,
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.defaults = ff_ac3_enc_defaults,
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};
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