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a247ac640d
Given that the AVCodec.next pointer has now been removed, most of the AVCodecs are not modified at all any more and can therefore be made const (as this patch does); the only exceptions are the very few codecs for external libraries that have a init_static_data callback. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com> Signed-off-by: James Almer <jamrial@gmail.com>
151 lines
6.1 KiB
C
151 lines
6.1 KiB
C
/*
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* Fixed-point MPEG audio decoder
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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 02110-1301 USA
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*/
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#include "config.h"
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#include "libavutil/samplefmt.h"
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#define USE_FLOATS 0
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#include "mpegaudio.h"
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#define SHR(a,b) (((int)(a))>>(b))
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/* WARNING: only correct for positive numbers */
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#define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
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#define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
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#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
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#define MULH3(x, y, s) MULH((s)*(x), y)
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#define MULLx(x, y, s) MULL((int)(x),(y),s)
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#define RENAME(a) a ## _fixed
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#define OUT_FMT AV_SAMPLE_FMT_S16
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#define OUT_FMT_P AV_SAMPLE_FMT_S16P
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/* Intensity stereo table. See commit b91d46614df189e7905538e7f5c4ed9c7ed0d274
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* (float based mp1/mp2/mp3 decoders.) for how they were created. */
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static const int32_t is_table[2][16] = {
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{ 0x000000, 0x1B0CB1, 0x2ED9EC, 0x400000, 0x512614, 0x64F34F, 0x800000 },
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{ 0x800000, 0x64F34F, 0x512614, 0x400000, 0x2ED9EC, 0x1B0CB1, 0x000000 }
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};
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/* Antialiasing table. See commit ce4a29c066cddfc180979ed86396812f24337985
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* (optimize antialias) for how they were created. */
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static const int32_t csa_table[8][4] = {
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{ 0x36E129F8, 0xDF128056, 0x15F3AA4E, 0xA831565E },
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{ 0x386E75F2, 0xE1CF24A5, 0x1A3D9A97, 0xA960AEB3 },
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{ 0x3CC6B73A, 0xEBF19FA6, 0x28B856E0, 0xAF2AE86C },
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{ 0x3EEEA054, 0xF45B88BC, 0x334A2910, 0xB56CE868 },
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{ 0x3FB6905C, 0xF9F27F18, 0x39A90F74, 0xBA3BEEBC },
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{ 0x3FF23F20, 0xFD60D1E4, 0x3D531104, 0xBD6E92C4 },
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{ 0x3FFE5932, 0xFF175EE4, 0x3F15B816, 0xBF1905B2 },
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{ 0x3FFFE34A, 0xFFC3612F, 0x3FC34479, 0xBFC37DE5 }
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};
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#include "mpegaudiodec_template.c"
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#if CONFIG_MP1_DECODER
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const AVCodec ff_mp1_decoder = {
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.name = "mp1",
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.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_MP1,
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.priv_data_size = sizeof(MPADecodeContext),
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.init = decode_init,
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.decode = decode_frame,
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.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
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};
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#endif
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#if CONFIG_MP2_DECODER
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const AVCodec ff_mp2_decoder = {
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.name = "mp2",
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.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_MP2,
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.priv_data_size = sizeof(MPADecodeContext),
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.init = decode_init,
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.decode = decode_frame,
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.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
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};
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#endif
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#if CONFIG_MP3_DECODER
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const AVCodec ff_mp3_decoder = {
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.name = "mp3",
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.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_MP3,
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.priv_data_size = sizeof(MPADecodeContext),
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.init = decode_init,
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.decode = decode_frame,
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.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
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};
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#endif
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#if CONFIG_MP3ADU_DECODER
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const AVCodec ff_mp3adu_decoder = {
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.name = "mp3adu",
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.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_MP3ADU,
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.priv_data_size = sizeof(MPADecodeContext),
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.init = decode_init,
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.decode = decode_frame_adu,
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.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_S16,
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AV_SAMPLE_FMT_NONE },
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
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};
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#endif
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#if CONFIG_MP3ON4_DECODER
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const AVCodec ff_mp3on4_decoder = {
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.name = "mp3on4",
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.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_MP3ON4,
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.priv_data_size = sizeof(MP3On4DecodeContext),
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.init = decode_init_mp3on4,
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.close = decode_close_mp3on4,
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.decode = decode_frame_mp3on4,
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.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush_mp3on4,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
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AV_SAMPLE_FMT_NONE },
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.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
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};
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#endif
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