FFmpeg/libavcodec/ac3enc_float.c
Michael Niedermayer 79d30321a2 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  wmaenc: use float planar sample format
  (e)ac3enc: use planar sample format
  aacenc: use planar sample format
  adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
  adpcmenc: move 'ch' variable to higher scope
  adpcmenc: fix 3 instances of variable shadowing
  adpcm_ima_wav: simplify encoding
  libvorbis: use planar sample format
  libmp3lame: use planar sample formats
  vorbisenc: use float planar sample format
  ffm: do not write or read the audio sample format
  parseutils: fix parsing of invalid alpha values
  doc/RELEASE_NOTES: update for the 9 release.
  smoothstreamingenc: Add a more verbose error message
  smoothstreamingenc: Ignore the return value from mkdir
  smoothstreamingenc: Try writing a manifest when opening the muxer
  smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
  smoothstreamingenc: Properly return errors from ism_flush to the caller
  smoothstreamingenc: Check the output UrlContext before accessing it

Conflicts:
	doc/RELEASE_NOTES
	libavcodec/aacenc.c
	libavcodec/ac3enc_template.c
	libavcodec/wmaenc.c
	tests/ref/lavf/ffm

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-10-07 11:28:38 +02:00

169 lines
4.5 KiB
C

/*
* 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 "internal.h"
#include "ac3enc.h"
#include "eac3enc.h"
#include "kbdwin.h"
#if CONFIG_AC3_ENCODER
#define AC3ENC_TYPE AC3ENC_TYPE_AC3
#include "ac3enc_opts_template.c"
static const AVClass ac3enc_class = {
.class_name = "AC-3 Encoder",
.item_name = av_default_item_name,
.option = ac3_options,
.version = LIBAVUTIL_VERSION_INT,
};
#endif
#include "ac3enc_template.c"
/**
* Finalize MDCT and free allocated memory.
*
* @param s AC-3 encoder private context
*/
av_cold void ff_ac3_float_mdct_end(AC3EncodeContext *s)
{
ff_mdct_end(&s->mdct);
av_freep(&s->mdct_window);
}
/**
* Initialize MDCT tables.
*
* @param s AC-3 encoder private context
* @return 0 on success, negative error code on failure
*/
av_cold int ff_ac3_float_mdct_init(AC3EncodeContext *s)
{
float *window;
int i, n, n2;
n = 1 << 9;
n2 = n >> 1;
window = av_malloc(n * sizeof(*window));
if (!window) {
av_log(s->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];
s->mdct_window = window;
return ff_mdct_init(&s->mdct, 9, 0, -2.0 / n);
}
/*
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(void *dsp, float *output,
const float *input, const float *window,
unsigned int len)
{
AVFloatDSPContext *fdsp = dsp;
fdsp->vector_fmul(output, input, window, len);
}
/*
* Normalize the input samples.
* Not needed for the floating-point encoder.
*/
static int normalize_samples(AC3EncodeContext *s)
{
return 0;
}
/*
* Scale MDCT coefficients from float to 24-bit fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
{
int chan_size = AC3_MAX_COEFS * s->num_blocks;
int cpl = s->cpl_on;
s->ac3dsp.float_to_fixed24(s->fixed_coef_buffer + (chan_size * !cpl),
s->mdct_coef_buffer + (chan_size * !cpl),
chan_size * (s->channels + cpl));
}
static void sum_square_butterfly(AC3EncodeContext *s, float sum[4],
const float *coef0, const float *coef1,
int len)
{
s->ac3dsp.sum_square_butterfly_float(sum, coef0, coef1, len);
}
/*
* Clip MDCT coefficients to allowable range.
*/
static void clip_coefficients(DSPContext *dsp, float *coef, unsigned int len)
{
dsp->vector_clipf(coef, coef, COEF_MIN, COEF_MAX, len);
}
/*
* Calculate a single coupling coordinate.
*/
static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
{
float coord = 0.125;
if (energy_cpl > 0)
coord *= sqrtf(energy_ch / energy_cpl);
return FFMIN(coord, COEF_MAX);
}
#if CONFIG_AC3_ENCODER
AVCodec ff_ac3_encoder = {
.name = "ac3",
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_AC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ff_ac3_encode_init,
.encode2 = ff_ac3_float_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
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,
.defaults = ac3_defaults,
};
#endif