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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /media/ffvpx/libavcodec/mpegaudiodec_fixed.c | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | media/ffvpx/libavcodec/mpegaudiodec_fixed.c | 148 |
1 files changed, 148 insertions, 0 deletions
diff --git a/media/ffvpx/libavcodec/mpegaudiodec_fixed.c b/media/ffvpx/libavcodec/mpegaudiodec_fixed.c new file mode 100644 index 0000000000..b5b6822a19 --- /dev/null +++ b/media/ffvpx/libavcodec/mpegaudiodec_fixed.c @@ -0,0 +1,148 @@ +/* + * 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 "config_components.h" +#include "libavutil/samplefmt.h" + +#define USE_FLOATS 0 + +#include "codec_internal.h" +#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 +const FFCodec ff_mp1_decoder = { + .p.name = "mp1", + CODEC_LONG_NAME("MP1 (MPEG audio layer 1)"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_MP1, + .priv_data_size = sizeof(MPADecodeContext), + .init = decode_init, + FF_CODEC_DECODE_CB(decode_frame), + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | + AV_CODEC_CAP_DR1, + .flush = flush, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#endif +#if CONFIG_MP2_DECODER +const FFCodec ff_mp2_decoder = { + .p.name = "mp2", + CODEC_LONG_NAME("MP2 (MPEG audio layer 2)"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_MP2, + .priv_data_size = sizeof(MPADecodeContext), + .init = decode_init, + FF_CODEC_DECODE_CB(decode_frame), + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | + AV_CODEC_CAP_DR1, + .flush = flush, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#endif +#if CONFIG_MP3_DECODER +const FFCodec ff_mp3_decoder = { + .p.name = "mp3", + CODEC_LONG_NAME("MP3 (MPEG audio layer 3)"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_MP3, + .priv_data_size = sizeof(MPADecodeContext), + .init = decode_init, + FF_CODEC_DECODE_CB(decode_frame), + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | + AV_CODEC_CAP_DR1, + .flush = flush, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#endif +#if CONFIG_MP3ADU_DECODER +const FFCodec ff_mp3adu_decoder = { + .p.name = "mp3adu", + CODEC_LONG_NAME("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_MP3ADU, + .priv_data_size = sizeof(MPADecodeContext), + .init = decode_init, + FF_CODEC_DECODE_CB(decode_frame_adu), + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | + AV_CODEC_CAP_DR1, + .flush = flush, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#endif +#if CONFIG_MP3ON4_DECODER +const FFCodec ff_mp3on4_decoder = { + .p.name = "mp3on4", + CODEC_LONG_NAME("MP3onMP4"), + .p.type = AVMEDIA_TYPE_AUDIO, + .p.id = AV_CODEC_ID_MP3ON4, + .priv_data_size = sizeof(MP3On4DecodeContext), + .init = decode_init_mp3on4, + .close = decode_close_mp3on4, + FF_CODEC_DECODE_CB(decode_frame_mp3on4), + .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | + AV_CODEC_CAP_DR1, + .flush = flush_mp3on4, + .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_NONE }, + .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, +}; +#endif |