mirror of https://git.ffmpeg.org/ffmpeg.git
126 lines
3.3 KiB
C
126 lines
3.3 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 CONFIG_FFT_FLOAT 0
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#undef CONFIG_AC3ENC_FLOAT
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#include "ac3enc.c"
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/**
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* Finalize MDCT and free allocated memory.
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*/
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static av_cold void mdct_end(AC3MDCTContext *mdct)
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{
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ff_mdct_end(&mdct->fft);
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}
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/**
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* Initialize MDCT tables.
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* @param nbits log2(MDCT size)
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*/
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static av_cold int mdct_init(AVCodecContext *avctx, AC3MDCTContext *mdct,
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int nbits)
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{
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int ret = ff_mdct_init(&mdct->fft, nbits, 0, -1.0);
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mdct->window = ff_ac3_window;
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return ret;
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}
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/**
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* Apply KBD window to input samples prior to MDCT.
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*/
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static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
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const int16_t *window, unsigned int len)
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{
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dsp->apply_window_int16(output, input, window, len);
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}
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/**
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* Calculate the log2() of the maximum absolute value in an array.
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* @param tab input array
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* @param n number of values in the array
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* @return log2(max(abs(tab[])))
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*/
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static int log2_tab(AC3EncodeContext *s, int16_t *src, int len)
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{
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int v = s->ac3dsp.ac3_max_msb_abs_int16(src, len);
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return av_log2(v);
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}
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/**
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* Normalize the input samples to use the maximum available precision.
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* This assumes signed 16-bit input samples.
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*
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* @return exponent shift
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*/
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static int normalize_samples(AC3EncodeContext *s)
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{
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int v = 14 - log2_tab(s, s->windowed_samples, AC3_WINDOW_SIZE);
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if (v > 0)
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s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v);
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/* +6 to right-shift from 31-bit to 25-bit */
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return v + 6;
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}
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/**
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* Scale MDCT coefficients to 25-bit signed fixed-point.
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*/
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static void scale_coefficients(AC3EncodeContext *s)
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{
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int blk, ch;
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for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
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AC3Block *block = &s->blocks[blk];
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for (ch = 1; ch <= s->channels; ch++) {
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s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS,
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block->coeff_shift[ch]);
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}
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}
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}
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AVCodec ff_ac3_fixed_encoder = {
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"ac3_fixed",
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AVMEDIA_TYPE_AUDIO,
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CODEC_ID_AC3,
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sizeof(AC3EncodeContext),
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ac3_encode_init,
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ac3_encode_frame,
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ac3_encode_close,
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NULL,
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.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
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.priv_class = &ac3enc_class,
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.channel_layouts = ff_ac3_channel_layouts,
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};
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