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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-28 14:29:10 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-28 14:29:10 +0000 |
commit | 2aa4a82499d4becd2284cdb482213d541b8804dd (patch) | |
tree | b80bf8bf13c3766139fbacc530efd0dd9d54394c /media/ffvpx/libavcodec/mpegaudiodec_fixed.c | |
parent | Initial commit. (diff) | |
download | firefox-upstream.tar.xz firefox-upstream.zip |
Adding upstream version 86.0.1.upstream/86.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | media/ffvpx/libavcodec/mpegaudiodec_fixed.c | 120 |
1 files changed, 120 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..ad7ceb20b6 --- /dev/null +++ b/media/ffvpx/libavcodec/mpegaudiodec_fixed.c @@ -0,0 +1,120 @@ +/* + * 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 + +#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_DR1, + .flush = flush, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#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_DR1, + .flush = flush, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#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_DR1, + .flush = flush, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#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_DR1, + .flush = flush, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_S16, + AV_SAMPLE_FMT_NONE }, +}; +#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_DR1, + .flush = flush_mp3on4, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, + AV_SAMPLE_FMT_NONE }, +}; +#endif |