ffmpeg/libavcodec/mpegaudiodec_fixed.c
Andreas Rheinhardt ed33bbe678 avcodec/mpegaudiodec: Hardcode tables to save space
The csa_tables (which always consist of 32 entries of four byte each,
but the type depends upon whether the decoder is fixed or
floating-point) are currently initialized once during decoder
initialization; yet it turns out that this is actually no benefit: The
code used to initialize these tables takes up 153 (fixed point) and 122
(floating point) bytes when compiled with GCC 9.3 with -O3 on x64, so it
is better to just hardcode these tables.

Essentially the same applies to the is_tables: They have a size of 128B
each and the code to initialize them occupies 149 (fixed point) resp.
140 (floating point) bytes. So hardcode them, too.

To make the origin of the tables clear, references to the code used to
create them have been added.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-12-08 17:51:47 +01:00

151 lines
6.0 KiB
C

/*
* Fixed-point MPEG audio decoder
*
* 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
*/
#include "config.h"
#include "libavutil/samplefmt.h"
#define USE_FLOATS 0
#include "mpegaudio.h"
#define SHR(a,b) (((int)(a))>>(b))
/* WARNING: only correct for positive numbers */
#define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
#define MULH3(x, y, s) MULH((s)*(x), y)
#define MULLx(x, y, s) MULL((int)(x),(y),s)
#define RENAME(a) a ## _fixed
#define OUT_FMT AV_SAMPLE_FMT_S16
#define OUT_FMT_P AV_SAMPLE_FMT_S16P
/* Intensity stereo table. See commit b91d46614df189e7905538e7f5c4ed9c7ed0d274
* (float based mp1/mp2/mp3 decoders.) for how they were created. */
static const int32_t is_table[2][16] = {
{ 0x000000, 0x1B0CB1, 0x2ED9EC, 0x400000, 0x512614, 0x64F34F, 0x800000 },
{ 0x800000, 0x64F34F, 0x512614, 0x400000, 0x2ED9EC, 0x1B0CB1, 0x000000 }
};
/* Antialiasing table. See commit ce4a29c066cddfc180979ed86396812f24337985
* (optimize antialias) for how they were created. */
static const int32_t csa_table[8][4] = {
{ 0x36E129F8, 0xDF128056, 0x15F3AA4E, 0xA831565E },
{ 0x386E75F2, 0xE1CF24A5, 0x1A3D9A97, 0xA960AEB3 },
{ 0x3CC6B73A, 0xEBF19FA6, 0x28B856E0, 0xAF2AE86C },
{ 0x3EEEA054, 0xF45B88BC, 0x334A2910, 0xB56CE868 },
{ 0x3FB6905C, 0xF9F27F18, 0x39A90F74, 0xBA3BEEBC },
{ 0x3FF23F20, 0xFD60D1E4, 0x3D531104, 0xBD6E92C4 },
{ 0x3FFE5932, 0xFF175EE4, 0x3F15B816, 0xBF1905B2 },
{ 0x3FFFE34A, 0xFFC3612F, 0x3FC34479, 0xBFC37DE5 }
};
#include "mpegaudiodec_template.c"
#if CONFIG_MP1_DECODER
AVCodec ff_mp1_decoder = {
.name = "mp1",
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP1,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP2_DECODER
AVCodec ff_mp2_decoder = {
.name = "mp2",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP3_DECODER
AVCodec ff_mp3_decoder = {
.name = "mp3",
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP3ADU_DECODER
AVCodec ff_mp3adu_decoder = {
.name = "mp3adu",
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ADU,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame_adu,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
};
#endif
#if CONFIG_MP3ON4_DECODER
AVCodec ff_mp3on4_decoder = {
.name = "mp3on4",
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ON4,
.priv_data_size = sizeof(MP3On4DecodeContext),
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
AV_CODEC_CAP_DR1,
.flush = flush_mp3on4,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
};
#endif