ffmpeg / libavcodec / ac3enc_fixed.c @ 6ad2bafc
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/*


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* The simplest AC3 encoder

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* Copyright (c) 2000 Fabrice Bellard

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* Copyright (c) 20062010 Justin Ruggles <justin.ruggles@gmail.com>

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* Copyright (c) 20062010 Prakash Punnoor <prakash@punnoor.de>

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*

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* This file is part of Libav.

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*

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* Libav 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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* Libav 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 Libav; if not, write to the Free Software

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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 021101301 USA

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*/

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/**

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* @file

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* fixedpoint AC3 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 16bit 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 rightshift from 31bit to 25bit */

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return v + 6; 
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} 
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/**

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* Scale MDCT coefficients to 25bit signed fixedpoint.

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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 = 0; 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 (AC3)"),

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.priv_class = &ac3enc_class, 
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.channel_layouts = ac3_channel_layouts, 
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}; 