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/*
* 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
* fixed-point AC-3 encoder.
*/
#define CONFIG_FFT_FLOAT 0
#undef CONFIG_AC3ENC_FLOAT
#include "ac3enc.c"
/**
* Finalize MDCT and free allocated memory.
*/
static av_cold void mdct_end(AC3MDCTContext *mdct)
{
ff_mdct_end(&mdct->fft);
}
/**
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*/
static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits)
{
int ret = ff_mdct_init(&mdct->fft, nbits, 0, -1.0);
mdct->window = ff_ac3_window;
return ret;
}
/**
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
const int16_t *window, unsigned int len)
{
dsp->apply_window_int16(output, input, window, len);
}
/**
* Calculate the log2() of the maximum absolute value in an array.
* @param tab input array
* @param n number of values in the array
* @return log2(max(abs(tab[])))
*/
static int log2_tab(AC3EncodeContext *s, int16_t *src, int len)
{
int v = s->ac3dsp.ac3_max_msb_abs_int16(src, len);
return av_log2(v);
}
/**
* Normalize the input samples to use the maximum available precision.
* This assumes signed 16-bit input samples.
*
* @return exponent shift
*/
static int normalize_samples(AC3EncodeContext *s)
{
int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE);
if (v > 0)
s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v);
/* +6 to right-shift from 31-bit to 25-bit */
return v + 6;
}
/**
* Scale MDCT coefficients to 25-bit signed fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
{
int blk, ch;
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
AC3Block *block = &s->blocks[blk];
for (ch = 0; ch < s->channels; ch++) {
s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS,
block->coeff_shift[ch]);
}
}
}
AVCodec ff_ac3_fixed_encoder = {
"ac3_fixed",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_AC3,
sizeof(AC3EncodeContext),
ac3_encode_init,
ac3_encode_frame,
ac3_encode_close,
NULL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.priv_class = &ac3enc_class,
.channel_layouts = ff_ac3_channel_layouts,
};
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