ffmpeg / libavcodec / ac3enc_float.c @ d36beb3f
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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 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 021101301 USA

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*/

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/**

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* @file

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* floatingpoint AC3 encoder.

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*/

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#define CONFIG_AC3ENC_FLOAT 1 
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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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av_freep(&mdct>window); 
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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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float *window;

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int i, n, n2;

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n = 1 << nbits;

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n2 = n >> 1;

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window = av_malloc(n * sizeof(*window));

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if (!window) {

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av_log(avctx, AV_LOG_ERROR, "Cannot allocate memory.\n");

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return AVERROR(ENOMEM);

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} 
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ff_kbd_window_init(window, 5.0, n2); 
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for (i = 0; i < n2; i++) 
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window[n1i] = window[i];

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mdct>window = window; 
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return ff_mdct_init(&mdct>fft, nbits, 0, 2.0 / n); 
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} 
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/**

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* Calculate a 512point MDCT

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* @param out 256 output frequency coefficients

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* @param in 512 windowed input audio samples

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*/

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static void mdct512(AC3MDCTContext *mdct, float *out, float *in) 
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{ 
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ff_mdct_calc(&mdct>fft, out, in); 
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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, float *output, const float *input, 
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const float *window, int n) 
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{ 
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dsp>vector_fmul(output, input, window, n); 
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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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*/

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static int normalize_samples(AC3EncodeContext *s) 
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{ 
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/* Normalization is not needed for floatingpoint samples, so just return 0 */

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return 0; 
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} 
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/**

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* Scale MDCT coefficients from float to 24bit fixedpoint.

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*/

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static void scale_coefficients(AC3EncodeContext *s) 
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{ 
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int i;

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for (i = 0; i < AC3_MAX_COEFS * AC3_MAX_BLOCKS * s>channels; i++) 
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s>fixed_coef_buffer[i] = SCALE_FLOAT(s>mdct_coef_buffer[i], 24);

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} 
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AVCodec ff_ac3_encoder = { 
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"ac3",

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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_FLT,AV_SAMPLE_FMT_NONE}, 
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.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC3)"),

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.channel_layouts = ac3_channel_layouts, 
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}; 