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https://github.com/FFmpeg/FFmpeg.git
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21b23ceab3
and remove FF_CODEC_CAP_INIT_THREADSAFE All our native codecs are already init-threadsafe (only wrappers for external libraries and hwaccels are typically not marked as init-threadsafe yet), so it is only natural for this to also be the default state. Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
149 lines
5.9 KiB
C
149 lines
5.9 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 "config_components.h"
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#include "libavutil/samplefmt.h"
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#define USE_FLOATS 0
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#include "codec_internal.h"
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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 FFCodec ff_mp1_decoder = {
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.p.name = "mp1",
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.p.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.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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FF_CODEC_DECODE_CB(decode_frame),
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.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.p.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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};
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#endif
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#if CONFIG_MP2_DECODER
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const FFCodec ff_mp2_decoder = {
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.p.name = "mp2",
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.p.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.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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FF_CODEC_DECODE_CB(decode_frame),
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.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.p.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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};
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#endif
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#if CONFIG_MP3_DECODER
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const FFCodec ff_mp3_decoder = {
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.p.name = "mp3",
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.p.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.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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FF_CODEC_DECODE_CB(decode_frame),
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.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.p.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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};
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#endif
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#if CONFIG_MP3ADU_DECODER
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const FFCodec ff_mp3adu_decoder = {
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.p.name = "mp3adu",
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.p.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.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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FF_CODEC_DECODE_CB(decode_frame_adu),
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.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF |
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AV_CODEC_CAP_DR1,
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.flush = flush,
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.p.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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};
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#endif
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#if CONFIG_MP3ON4_DECODER
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const FFCodec ff_mp3on4_decoder = {
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.p.name = "mp3on4",
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.p.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.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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FF_CODEC_DECODE_CB(decode_frame_mp3on4),
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.p.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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.p.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_CLEANUP,
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};
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#endif
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