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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
* floating-point AC-3 encoder.
*/
#define CONFIG_AC3ENC_FLOAT 1
#include "ac3enc.c"
/**
* Finalize MDCT and free allocated memory.
*/
static av_cold void mdct_end(AC3MDCTContext *mdct)
{
ff_mdct_end(&mdct->fft);
av_freep(&mdct->window);
}
/**
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*/
static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
int nbits)
{
float *window;
int i, n, n2;
n = 1 << nbits;
n2 = n >> 1;
window = av_malloc(n * sizeof(*window));
if (!window) {
av_log(avctx, AV_LOG_ERROR, "Cannot allocate memory.\n");
return AVERROR(ENOMEM);
}
ff_kbd_window_init(window, 5.0, n2);
for (i = 0; i < n2; i++)
window[n-1-i] = window[i];
mdct->window = window;
return ff_mdct_init(&mdct->fft, nbits, 0, -2.0 / n);
}
/**
* Calculate a 512-point MDCT
* @param out 256 output frequency coefficients
* @param in 512 windowed input audio samples
*/
static void mdct512(AC3MDCTContext *mdct, float *out, float *in)
{
ff_mdct_calc(&mdct->fft, out, in);
}
/**
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, float *output, const float *input,
const float *window, int n)
{
dsp->vector_fmul(output, input, window, n);
}
/**
* Normalize the input samples to use the maximum available precision.
*/
static int normalize_samples(AC3EncodeContext *s)
{
/* Normalization is not needed for floating-point samples, so just return 0 */
return 0;
}
/**
* Scale MDCT coefficients from float to 24-bit fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
{
int i;
for (i = 0; i < AC3_MAX_COEFS * AC3_MAX_BLOCKS * s->channels; i++)
s->fixed_coef_buffer[i] = SCALE_FLOAT(s->mdct_coef_buffer[i], 24);
}
AVCodec ac3_encoder = {
"ac3",
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_FLT,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.channel_layouts = ac3_channel_layouts,
};
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