ffmpeg/libavcodec/ac3enc_fixed.c
Michael Niedermayer 79ae084e9b Merge remote-tracking branch 'qatar/master'
* qatar/master: (58 commits)
  amrnbdec: check frame size before decoding.
  cscd: use negative error values to indicate decode_init() failures.
  h264: prevent overreads in intra PCM decoding.
  FATE: do not decode audio in the nuv test.
  dxa: set audio stream time base using the sample rate
  psx-str: do not allow seeking by bytes
  asfdec: Do not set AVCodecContext.frame_size
  vqf: set packet parameters after av_new_packet()
  mpegaudiodec: use DSPUtil.butterflies_float().
  FATE: add mp3 test for sample that exhibited false overreads
  fate: add cdxl test for bit line plane arrangement
  vmnc: return error on decode_init() failure.
  libvorbis: add/update error messages
  libvorbis: use AVFifoBuffer for output packet buffer
  libvorbis: remove unneeded e_o_s check
  libvorbis: check return values for functions that can return errors
  libvorbis: use float input instead of s16
  libvorbis: do not flush libvorbis analysis if dsp state was not initialized
  libvorbis: use VBR by default, with default quality of 3
  libvorbis: fix use of minrate/maxrate AVOptions
  ...

Conflicts:
	Changelog
	doc/APIchanges
	libavcodec/avcodec.h
	libavcodec/dpxenc.c
	libavcodec/libvorbis.c
	libavcodec/vmnc.c
	libavformat/asfdec.c
	libavformat/id3v2enc.c
	libavformat/internal.h
	libavformat/mp3enc.c
	libavformat/utils.c
	libavformat/version.h
	libswscale/utils.c
	tests/fate/video.mak
	tests/ref/fate/nuv
	tests/ref/fate/prores-alpha
	tests/ref/lavf/ffm
	tests/ref/vsynth1/prores
	tests/ref/vsynth2/prores

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-03-01 03:17:11 +01:00

162 lines
4.6 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
* fixed-point AC-3 encoder.
*/
#define CONFIG_FFT_FLOAT 0
#undef CONFIG_AC3ENC_FLOAT
#include "ac3enc.h"
#include "eac3enc.h"
#define AC3ENC_TYPE AC3ENC_TYPE_AC3_FIXED
#include "ac3enc_opts_template.c"
static const AVClass ac3enc_class = { "Fixed-Point AC-3 Encoder", av_default_item_name,
ac3fixed_options, LIBAVUTIL_VERSION_INT };
#include "ac3enc_template.c"
/**
* Finalize MDCT and free allocated memory.
*
* @param s AC-3 encoder private context
*/
av_cold void AC3_NAME(mdct_end)(AC3EncodeContext *s)
{
ff_mdct_end(&s->mdct);
}
/**
* Initialize MDCT tables.
*
* @param s AC-3 encoder private context
* @return 0 on success, negative error code on failure
*/
av_cold int AC3_NAME(mdct_init)(AC3EncodeContext *s)
{
int ret = ff_mdct_init(&s->mdct, 9, 0, -1.0);
s->mdct_window = ff_ac3_window;
return ret;
}
/*
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(DSPContext *dsp, int16_t *output, const int16_t *input,
const int16_t *window, unsigned int len)
{
dsp->apply_window_int16(output, input, window, len);
}
/*
* Normalize the input samples to use the maximum available precision.
* This assumes signed 16-bit input samples.
*/
static int normalize_samples(AC3EncodeContext *s)
{
int v = s->ac3dsp.ac3_max_msb_abs_int16(s->windowed_samples, AC3_WINDOW_SIZE);
v = 14 - av_log2(v);
if (v > 0)
s->ac3dsp.ac3_lshift_int16(s->windowed_samples, AC3_WINDOW_SIZE, v);
/* +6 to right-shift from 31-bit to 25-bit */
return v + 6;
}
/*
* Scale MDCT coefficients to 25-bit signed fixed-point.
*/
static void scale_coefficients(AC3EncodeContext *s)
{
int blk, ch;
for (blk = 0; blk < s->num_blocks; blk++) {
AC3Block *block = &s->blocks[blk];
for (ch = 1; ch <= s->channels; ch++) {
s->ac3dsp.ac3_rshift_int32(block->mdct_coef[ch], AC3_MAX_COEFS,
block->coeff_shift[ch]);
}
}
}
static void sum_square_butterfly(AC3EncodeContext *s, int64_t sum[4],
const int32_t *coef0, const int32_t *coef1,
int len)
{
s->ac3dsp.sum_square_butterfly_int32(sum, coef0, coef1, len);
}
/*
* Clip MDCT coefficients to allowable range.
*/
static void clip_coefficients(DSPContext *dsp, int32_t *coef, unsigned int len)
{
dsp->vector_clip_int32(coef, coef, COEF_MIN, COEF_MAX, len);
}
/*
* Calculate a single coupling coordinate.
*/
static CoefType calc_cpl_coord(CoefSumType energy_ch, CoefSumType energy_cpl)
{
if (energy_cpl <= COEF_MAX) {
return 1048576;
} else {
uint64_t coord = energy_ch / (energy_cpl >> 24);
uint32_t coord32 = FFMIN(coord, 1073741824);
coord32 = ff_sqrt(coord32) << 9;
return FFMIN(coord32, COEF_MAX);
}
}
static av_cold int ac3_fixed_encode_init(AVCodecContext *avctx)
{
AC3EncodeContext *s = avctx->priv_data;
s->fixed_point = 1;
return ff_ac3_encode_init(avctx);
}
AVCodec ff_ac3_fixed_encoder = {
.name = "ac3_fixed",
.type = AVMEDIA_TYPE_AUDIO,
.id = CODEC_ID_AC3,
.priv_data_size = sizeof(AC3EncodeContext),
.init = ac3_fixed_encode_init,
.encode = ff_ac3_fixed_encode_frame,
.close = ff_ac3_encode_close,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,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,
};