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| author | Andreas Rheinhardt <andreas.rheinhardt@outlook.com> | 2021-04-14 04:33:24 +0200 |
|---|---|---|
| committer | James Almer <jamrial@gmail.com> | 2021-04-27 10:43:13 -0300 |
| commit | 420cedd49745b284c35d97b936b71ff79b43bdf7 (patch) | |
| tree | c77f82a84afc7b4d206e28ef42b25fa0f4cdbd75 /libavresample | |
| parent | d40bb518b50561db60ef71ab0e37eb7f3fb9043b (diff) | |
| download | ffmpeg-420cedd49745b284c35d97b936b71ff79b43bdf7.tar.gz | |
libavresample: Remove deprecated library
Deprecated in c29038f3041a4080342b2e333c1967d136749c0f.
The resample filter based upon this library has been removed as well.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Diffstat (limited to 'libavresample')
45 files changed, 0 insertions, 9694 deletions
diff --git a/libavresample/Makefile b/libavresample/Makefile deleted file mode 100644 index 90f025a9f9..0000000000 --- a/libavresample/Makefile +++ /dev/null @@ -1,19 +0,0 @@ -NAME = avresample -DESC = Libav audio resampling library - -HEADERS = avresample.h \ - version.h \ - -OBJS = audio_convert.o \ - audio_data.o \ - audio_mix.o \ - audio_mix_matrix.o \ - dither.o \ - options.o \ - resample.o \ - utils.o \ - -# Windows resource file -SLIBOBJS-$(HAVE_GNU_WINDRES) += avresampleres.o - -TESTPROGS = avresample diff --git a/libavresample/aarch64/Makefile b/libavresample/aarch64/Makefile deleted file mode 100644 index f92699ef1a..0000000000 --- a/libavresample/aarch64/Makefile +++ /dev/null @@ -1,7 +0,0 @@ -OBJS += aarch64/audio_convert_init.o \ - aarch64/resample_init.o \ - -OBJS-$(CONFIG_NEON_CLOBBER_TEST) += aarch64/neontest.o - -NEON-OBJS += aarch64/audio_convert_neon.o \ - aarch64/resample_neon.o \ diff --git a/libavresample/aarch64/asm-offsets.h b/libavresample/aarch64/asm-offsets.h deleted file mode 100644 index 0b582446f6..0000000000 --- a/libavresample/aarch64/asm-offsets.h +++ /dev/null @@ -1,28 +0,0 @@ -/* - * 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 - */ - -#ifndef AVRESAMPLE_AARCH64_ASM_OFFSETS_H -#define AVRESAMPLE_AARCH64_ASM_OFFSETS_H - -/* struct ResampleContext */ -#define FILTER_BANK 0x10 -#define FILTER_LENGTH 0x18 -#define PHASE_SHIFT 0x34 -#define PHASE_MASK (PHASE_SHIFT + 0x04) // loaded as pair - -#endif /* AVRESAMPLE_AARCH64_ASM_OFFSETS_H */ diff --git a/libavresample/aarch64/audio_convert_init.c b/libavresample/aarch64/audio_convert_init.c deleted file mode 100644 index b5b0d1eee0..0000000000 --- a/libavresample/aarch64/audio_convert_init.c +++ /dev/null @@ -1,49 +0,0 @@ -/* - * 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 <stdint.h> - -#include "config.h" -#include "libavutil/attributes.h" -#include "libavutil/cpu.h" -#include "libavutil/aarch64/cpu.h" -#include "libavutil/samplefmt.h" -#include "libavresample/audio_convert.h" - -void ff_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len); -void ff_conv_fltp_to_s16_neon(int16_t *dst, float *const *src, - int len, int channels); -void ff_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src, - int len, int channels); - -av_cold void ff_audio_convert_init_aarch64(AudioConvert *ac) -{ - int cpu_flags = av_get_cpu_flags(); - - if (have_neon(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT, - 0, 16, 8, "NEON", - ff_conv_flt_to_s16_neon); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 2, 16, 8, "NEON", - ff_conv_fltp_to_s16_2ch_neon); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 0, 16, 8, "NEON", - ff_conv_fltp_to_s16_neon); - } -} diff --git a/libavresample/aarch64/audio_convert_neon.S b/libavresample/aarch64/audio_convert_neon.S deleted file mode 100644 index e13e277e61..0000000000 --- a/libavresample/aarch64/audio_convert_neon.S +++ /dev/null @@ -1,363 +0,0 @@ -/* - * Copyright (c) 2008 Mans Rullgard <mans@mansr.com> - * Copyright (c) 2014 Janne Grunau <janne-libav@jannau.net> - * - * 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/aarch64/asm.S" - -function ff_conv_flt_to_s16_neon, export=1 - subs x2, x2, #8 - ld1 {v0.4s}, [x1], #16 - fcvtzs v4.4s, v0.4s, #31 - ld1 {v1.4s}, [x1], #16 - fcvtzs v5.4s, v1.4s, #31 - b.eq 3f - ands x12, x2, #~15 - b.eq 2f -1: subs x12, x12, #16 - sqrshrn v4.4h, v4.4s, #16 - ld1 {v2.4s}, [x1], #16 - fcvtzs v6.4s, v2.4s, #31 - sqrshrn2 v4.8h, v5.4s, #16 - ld1 {v3.4s}, [x1], #16 - fcvtzs v7.4s, v3.4s, #31 - sqrshrn v6.4h, v6.4s, #16 - st1 {v4.8h}, [x0], #16 - sqrshrn2 v6.8h, v7.4s, #16 - ld1 {v0.4s}, [x1], #16 - fcvtzs v4.4s, v0.4s, #31 - ld1 {v1.4s}, [x1], #16 - fcvtzs v5.4s, v1.4s, #31 - st1 {v6.8h}, [x0], #16 - b.ne 1b - ands x2, x2, #15 - b.eq 3f -2: ld1 {v2.4s}, [x1], #16 - sqrshrn v4.4h, v4.4s, #16 - fcvtzs v6.4s, v2.4s, #31 - ld1 {v3.4s}, [x1], #16 - sqrshrn2 v4.8h, v5.4s, #16 - fcvtzs v7.4s, v3.4s, #31 - sqrshrn v6.4h, v6.4s, #16 - st1 {v4.8h}, [x0], #16 - sqrshrn2 v6.8h, v7.4s, #16 - st1 {v6.8h}, [x0] - ret -3: sqrshrn v4.4h, v4.4s, #16 - sqrshrn2 v4.8h, v5.4s, #16 - st1 {v4.8h}, [x0] - ret -endfunc - -function ff_conv_fltp_to_s16_2ch_neon, export=1 - ldp x4, x5, [x1] - subs x2, x2, #8 - ld1 {v0.4s}, [x4], #16 - fcvtzs v4.4s, v0.4s, #31 - ld1 {v1.4s}, [x4], #16 - fcvtzs v5.4s, v1.4s, #31 - ld1 {v2.4s}, [x5], #16 - fcvtzs v6.4s, v2.4s, #31 - ld1 {v3.4s}, [x5], #16 - fcvtzs v7.4s, v3.4s, #31 - b.eq 3f - ands x12, x2, #~15 - b.eq 2f -1: subs x12, x12, #16 - ld1 {v16.4s}, [x4], #16 - fcvtzs v20.4s, v16.4s, #31 - sri v6.4s, v4.4s, #16 - ld1 {v17.4s}, [x4], #16 - fcvtzs v21.4s, v17.4s, #31 - ld1 {v18.4s}, [x5], #16 - fcvtzs v22.4s, v18.4s, #31 - ld1 {v19.4s}, [x5], #16 - sri v7.4s, v5.4s, #16 - st1 {v6.4s}, [x0], #16 - fcvtzs v23.4s, v19.4s, #31 - st1 {v7.4s}, [x0], #16 - sri v22.4s, v20.4s, #16 - ld1 {v0.4s}, [x4], #16 - sri v23.4s, v21.4s, #16 - st1 {v22.4s}, [x0], #16 - fcvtzs v4.4s, v0.4s, #31 - ld1 {v1.4s}, [x4], #16 - fcvtzs v5.4s, v1.4s, #31 - ld1 {v2.4s}, [x5], #16 - fcvtzs v6.4s, v2.4s, #31 - ld1 {v3.4s}, [x5], #16 - fcvtzs v7.4s, v3.4s, #31 - st1 {v23.4s}, [x0], #16 - b.ne 1b - ands x2, x2, #15 - b.eq 3f -2: sri v6.4s, v4.4s, #16 - ld1 {v0.4s}, [x4], #16 - fcvtzs v0.4s, v0.4s, #31 - ld1 {v1.4s}, [x4], #16 - fcvtzs v1.4s, v1.4s, #31 - ld1 {v2.4s}, [x5], #16 - fcvtzs v2.4s, v2.4s, #31 - sri v7.4s, v5.4s, #16 - ld1 {v3.4s}, [x5], #16 - fcvtzs v3.4s, v3.4s, #31 - sri v2.4s, v0.4s, #16 - st1 {v6.4s,v7.4s}, [x0], #32 - sri v3.4s, v1.4s, #16 - st1 {v2.4s,v3.4s}, [x0], #32 - ret -3: sri v6.4s, v4.4s, #16 - sri v7.4s, v5.4s, #16 - st1 {v6.4s,v7.4s}, [x0] - ret -endfunc - -function ff_conv_fltp_to_s16_neon, export=1 - cmp w3, #2 - b.eq X(ff_conv_fltp_to_s16_2ch_neon) - b.gt 1f - ldr x1, [x1] - b X(ff_conv_flt_to_s16_neon) -1: - cmp w3, #4 - lsl x12, x3, #1 - b.lt 4f - -5: // 4 channels - ldp x4, x5, [x1], #16 - ldp x6, x7, [x1], #16 - mov w9, w2 - mov x8, x0 - ld1 {v4.4s}, [x4], #16 - fcvtzs v4.4s, v4.4s, #31 - ld1 {v5.4s}, [x5], #16 - fcvtzs v5.4s, v5.4s, #31 - ld1 {v6.4s}, [x6], #16 - fcvtzs v6.4s, v6.4s, #31 - ld1 {v7.4s}, [x7], #16 - fcvtzs v7.4s, v7.4s, #31 -6: - subs w9, w9, #8 - ld1 {v0.4s}, [x4], #16 - fcvtzs v0.4s, v0.4s, #31 - sri v5.4s, v4.4s, #16 - ld1 {v1.4s}, [x5], #16 - fcvtzs v1.4s, v1.4s, #31 - sri v7.4s, v6.4s, #16 - ld1 {v2.4s}, [x6], #16 - fcvtzs v2.4s, v2.4s, #31 - zip1 v16.4s, v5.4s, v7.4s - ld1 {v3.4s}, [x7], #16 - fcvtzs v3.4s, v3.4s, #31 - zip2 v17.4s, v5.4s, v7.4s - st1 {v16.d}[0], [x8], x12 - sri v1.4s, v0.4s, #16 - st1 {v16.d}[1], [x8], x12 - sri v3.4s, v2.4s, #16 - st1 {v17.d}[0], [x8], x12 - zip1 v18.4s, v1.4s, v3.4s - st1 {v17.d}[1], [x8], x12 - zip2 v19.4s, v1.4s, v3.4s - b.eq 7f - ld1 {v4.4s}, [x4], #16 - fcvtzs v4.4s, v4.4s, #31 - st1 {v18.d}[0], [x8], x12 - ld1 {v5.4s}, [x5], #16 - fcvtzs v5.4s, v5.4s, #31 - st1 {v18.d}[1], [x8], x12 - ld1 {v6.4s}, [x6], #16 - fcvtzs v6.4s, v6.4s, #31 - st1 {v19.d}[0], [x8], x12 - ld1 {v7.4s}, [x7], #16 - fcvtzs v7.4s, v7.4s, #31 - st1 {v19.d}[1], [x8], x12 - b 6b -7: - st1 {v18.d}[0], [x8], x12 - st1 {v18.d}[1], [x8], x12 - st1 {v19.d}[0], [x8], x12 - st1 {v19.d}[1], [x8], x12 - subs w3, w3, #4 - b.eq end - cmp w3, #4 - add x0, x0, #8 - b.ge 5b - -4: // 2 channels - cmp w3, #2 - b.lt 4f - ldp x4, x5, [x1], #16 - mov w9, w2 - mov x8, x0 - tst w9, #8 - ld1 {v4.4s}, [x4], #16 - fcvtzs v4.4s, v4.4s, #31 - ld1 {v5.4s}, [x5], #16 - fcvtzs v5.4s, v5.4s, #31 - ld1 {v6.4s}, [x4], #16 - fcvtzs v6.4s, v6.4s, #31 - ld1 {v7.4s}, [x5], #16 - fcvtzs v7.4s, v7.4s, #31 - b.eq 6f - subs w9, w9, #8 - b.eq 7f - sri v5.4s, v4.4s, #16 - ld1 {v4.4s}, [x4], #16 - fcvtzs v4.4s, v4.4s, #31 - st1 {v5.s}[0], [x8], x12 - sri v7.4s, v6.4s, #16 - st1 {v5.s}[1], [x8], x12 - ld1 {v6.4s}, [x4], #16 - fcvtzs v6.4s, v6.4s, #31 - st1 {v5.s}[2], [x8], x12 - st1 {v5.s}[3], [x8], x12 - st1 {v7.s}[0], [x8], x12 - st1 {v7.s}[1], [x8], x12 - ld1 {v5.4s}, [x5], #16 - fcvtzs v5.4s, v5.4s, #31 - st1 {v7.s}[2], [x8], x12 - st1 {v7.s}[3], [x8], x12 - ld1 {v7.4s}, [x5], #16 - fcvtzs v7.4s, v7.4s, #31 -6: - subs w9, w9, #16 - ld1 {v0.4s}, [x4], #16 - sri v5.4s, v4.4s, #16 - fcvtzs v0.4s, v0.4s, #31 - ld1 {v1.4s}, [x5], #16 - sri v7.4s, v6.4s, #16 - st1 {v5.s}[0], [x8], x12 - st1 {v5.s}[1], [x8], x12 - fcvtzs v1.4s, v1.4s, #31 - st1 {v5.s}[2], [x8], x12 - st1 {v5.s}[3], [x8], x12 - ld1 {v2.4s}, [x4], #16 - st1 {v7.s}[0], [x8], x12 - fcvtzs v2.4s, v2.4s, #31 - st1 {v7.s}[1], [x8], x12 - ld1 {v3.4s}, [x5], #16 - st1 {v7.s}[2], [x8], x12 - fcvtzs v3.4s, v3.4s, #31 - st1 {v7.s}[3], [x8], x12 - sri v1.4s, v0.4s, #16 - sri v3.4s, v2.4s, #16 - b.eq 6f - ld1 {v4.4s}, [x4], #16 - st1 {v1.s}[0], [x8], x12 - fcvtzs v4.4s, v4.4s, #31 - st1 {v1.s}[1], [x8], x12 - ld1 {v5.4s}, [x5], #16 - st1 {v1.s}[2], [x8], x12 - fcvtzs v5.4s, v5.4s, #31 - st1 {v1.s}[3], [x8], x12 - ld1 {v6.4s}, [x4], #16 - st1 {v3.s}[0], [x8], x12 - fcvtzs v6.4s, v6.4s, #31 - st1 {v3.s}[1], [x8], x12 - ld1 {v7.4s}, [x5], #16 - st1 {v3.s}[2], [x8], x12 - fcvtzs v7.4s, v7.4s, #31 - st1 {v3.s}[3], [x8], x12 - b.gt 6b -6: - st1 {v1.s}[0], [x8], x12 - st1 {v1.s}[1], [x8], x12 - st1 {v1.s}[2], [x8], x12 - st1 {v1.s}[3], [x8], x12 - st1 {v3.s}[0], [x8], x12 - st1 {v3.s}[1], [x8], x12 - st1 {v3.s}[2], [x8], x12 - st1 {v3.s}[3], [x8], x12 - b 8f -7: - sri v5.4s, v4.4s, #16 - sri v7.4s, v6.4s, #16 - st1 {v5.s}[0], [x8], x12 - st1 {v5.s}[1], [x8], x12 - st1 {v5.s}[2], [x8], x12 - st1 {v5.s}[3], [x8], x12 - st1 {v7.s}[0], [x8], x12 - st1 {v7.s}[1], [x8], x12 - st1 {v7.s}[2], [x8], x12 - st1 {v7.s}[3], [x8], x12 -8: - subs w3, w3, #2 - add x0, x0, #4 - b.eq end - -4: // 1 channel - ldr x4, [x1] - tst w2, #8 - mov w9, w2 - mov x5, x0 - ld1 {v0.4s}, [x4], #16 - fcvtzs v0.4s, v0.4s, #31 - ld1 {v1.4s}, [x4], #16 - fcvtzs v1.4s, v1.4s, #31 - b.ne 8f -6: - subs w9, w9, #16 - ld1 {v2.4s}, [x4], #16 - fcvtzs v2.4s, v2.4s, #31 - ld1 {v3.4s}, [x4], #16 - fcvtzs v3.4s, v3.4s, #31 - st1 {v0.h}[1], [x5], x12 - st1 {v0.h}[3], [x5], x12 - st1 {v0.h}[5], [x5], x12 - st1 {v0.h}[7], [x5], x12 - st1 {v1.h}[1], [x5], x12 - st1 {v1.h}[3], [x5], x12 - st1 {v1.h}[5], [x5], x12 - st1 {v1.h}[7], [x5], x12 - b.eq 7f - ld1 {v0.4s}, [x4], #16 - fcvtzs v0.4s, v0.4s, #31 - ld1 {v1.4s}, [x4], #16 - fcvtzs v1.4s, v1.4s, #31 -7: - st1 {v2.h}[1], [x5], x12 - st1 {v2.h}[3], [x5], x12 - st1 {v2.h}[5], [x5], x12 - st1 {v2.h}[7], [x5], x12 - st1 {v3.h}[1], [x5], x12 - st1 {v3.h}[3], [x5], x12 - st1 {v3.h}[5], [x5], x12 - st1 {v3.h}[7], [x5], x12 - b.gt 6b - ret -8: - subs w9, w9, #8 - st1 {v0.h}[1], [x5], x12 - st1 {v0.h}[3], [x5], x12 - st1 {v0.h}[5], [x5], x12 - st1 {v0.h}[7], [x5], x12 - st1 {v1.h}[1], [x5], x12 - st1 {v1.h}[3], [x5], x12 - st1 {v1.h}[5], [x5], x12 - st1 {v1.h}[7], [x5], x12 - b.eq end - ld1 {v0.4s}, [x4], #16 - fcvtzs v0.4s, v0.4s, #31 - ld1 {v1.4s}, [x4], #16 - fcvtzs v1.4s, v1.4s, #31 - b 6b -end: - ret -endfunc diff --git a/libavresample/aarch64/neontest.c b/libavresample/aarch64/neontest.c deleted file mode 100644 index e956ee6b0d..0000000000 --- a/libavresample/aarch64/neontest.c +++ /dev/null @@ -1,31 +0,0 @@ -/* - * check NEON registers for clobbers - * Copyright (c) 2013 Martin Storsjo - * - * 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 "libavresample/avresample.h" -#include "libavutil/aarch64/neontest.h" - -wrap(avresample_convert(AVAudioResampleContext *avr, uint8_t **output, - int out_plane_size, int out_samples, uint8_t **input, - int in_plane_size, int in_samples)) -{ - testneonclobbers(avresample_convert, avr, output, out_plane_size, - out_samples, input, in_plane_size, in_samples); -} diff --git a/libavresample/aarch64/resample_init.c b/libavresample/aarch64/resample_init.c deleted file mode 100644 index e21c600286..0000000000 --- a/libavresample/aarch64/resample_init.c +++ /dev/null @@ -1,71 +0,0 @@ -/* - * 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 <stdint.h> - -#include "config.h" -#include "libavutil/cpu.h" -#include "libavutil/aarch64/cpu.h" -#include "libavutil/internal.h" -#include "libavutil/samplefmt.h" -#include "libavresample/resample.h" - -#include "asm-offsets.h" - -AV_CHECK_OFFSET(struct ResampleContext, filter_bank, FILTER_BANK); -AV_CHECK_OFFSET(struct ResampleContext, filter_length, FILTER_LENGTH); -AV_CHECK_OFFSET(struct ResampleContext, phase_shift, PHASE_SHIFT); -AV_CHECK_OFFSET(struct ResampleContext, phase_mask, PHASE_MASK); - -void ff_resample_one_dbl_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); -void ff_resample_one_flt_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); -void ff_resample_one_s16_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); -void ff_resample_one_s32_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); - -av_cold void ff_audio_resample_init_aarch64(ResampleContext *c, - enum AVSampleFormat sample_fmt) -{ - int cpu_flags = av_get_cpu_flags(); - - if (have_neon(cpu_flags)) { - if (!c->linear) { - switch (sample_fmt) { - case AV_SAMPLE_FMT_DBLP: - c->resample_one = ff_resample_one_dbl_neon; - break; - case AV_SAMPLE_FMT_FLTP: - c->resample_one = ff_resample_one_flt_neon; - break; - case AV_SAMPLE_FMT_S16P: - c->resample_one = ff_resample_one_s16_neon; - break; - case AV_SAMPLE_FMT_S32P: - c->resample_one = ff_resample_one_s32_neon; - break; - } - } - } -} diff --git a/libavresample/aarch64/resample_neon.S b/libavresample/aarch64/resample_neon.S deleted file mode 100644 index d3c2cbf561..0000000000 --- a/libavresample/aarch64/resample_neon.S +++ /dev/null @@ -1,233 +0,0 @@ -/* - * Copyright (c) 2014 Janne Grunau <janne-libav@jannau.net> - * - * 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 "libavutil/aarch64/asm.S" -#include "asm-offsets.h" - -.macro resample_one fmt, es=2 -.ifnc \fmt, dbl - .macro M_MUL2 x:vararg - .endm - .macro M_MLA2 x:vararg - .endm -.endif -function ff_resample_one_\fmt\()_neon, export=1 - sxtw x2, w2 - ldr x9, [x0, #FILTER_BANK] - ldr w6, [x0, #FILTER_LENGTH] - ldp w7, w8, [x0, #PHASE_SHIFT] // and phase_mask - lsr x10, x4, x7 // sample_index - and x4, x4, x8 - lsl x11, x6, #\es // filter_length * elem_size - add x3, x3, x10, lsl #\es // src[sample_index] - madd x9, x11, x4, x9 // filter - cmp w6, #16 - b.lt 5f -8: // remaining filter_length at least 16 - subs w6, w6, #16 - LOAD8 v4, v5, v6, v7, x3 - LOAD8 v16, v17, v18, v19, x9 - M_MUL v0, v4, v16, v1 - M_MUL2 v1, v6, v18 -7: - LOAD8 v20, v21, v22, v23, x3 - M_MLA v0, v5, v17, v1 - M_MLA2 v1, v7, v19 - LOAD8 v24, v25, v26, v27, x9 - M_MLA v0, v20, v24, v1 - M_MLA2 v1, v22, v26 - b.eq 6f - cmp w6, #16 - M_MLA v0, v21, v25, v1 - M_MLA2 v1, v23, v27 - b.lt 4f - subs w6, w6, #16 - LOAD8 v4, v5, v6, v7, x3 - LOAD8 v16, v17, v18, v19, x9 - M_MLA v0, v4, v16, v1 - M_MLA2 v1, v6, v18 - b 7b -6: - M_MLA v0, v21, v25, v1 - M_MLA2 v1, v23, v27 - STORE_ONE 0, x1, x2, v1 - ret -5: - movi v0.16b, #0 - movi v1.16b, #0 -4: // remaining filter_length 1-15 - cmp w6, #4 - b.lt 2f - subs w6, w6, #4 - LOAD4 v4, v5, x3 - LOAD4 v6, v7, x9 - M_MLA v0, v4, v6, v1 - M_MLA2 v1, v5, v7 - b.eq 0f - b 4b -2: // remaining filter_length 1-3 - cmp w6, #2 - b.lt 1f - LOAD2 2, x3 - LOAD2 3, x9 - subs w6, w6, #2 - M_MLA v0, v2, v3 - b.eq 0f -1: // remaining filter_length 1 - LOAD1 6, x3 - LOAD1 7, x9 - M_MLA v0, v6, v7 -0: - STORE_ONE 0, x1, x2, v1 - ret -endfunc - -.purgem LOAD1 -.purgem LOAD2 -.purgem LOAD4 -.purgem LOAD8 -.purgem M_MLA -.purgem M_MLA2 -.purgem M_MUL -.purgem M_MUL2 -.purgem STORE_ONE -.endm - - -.macro LOAD1 d1, addr - ldr d\d1, [\addr], #8 -.endm -.macro LOAD2 d1, addr - ld1 {v\d1\().2d}, [\addr], #16 -.endm -.macro LOAD4 d1, d2, addr - ld1 {\d1\().2d,\d2\().2d}, [\addr], #32 -.endm -.macro LOAD8 d1, d2, d3, d4, addr - ld1 {\d1\().2d,\d2\().2d,\d3\().2d,\d4\().2d}, [\addr], #64 -.endm -.macro M_MLA d, r0, r1, d2:vararg - fmla \d\().2d, \r0\().2d, \r1\().2d -.endm -.macro M_MLA2 second:vararg - M_MLA \second -.endm -.macro M_MUL d, r0, r1, d2:vararg - fmul \d\().2d, \r0\().2d, \r1\().2d -.endm -.macro M_MUL2 second:vararg - M_MUL \second -.endm -.macro STORE_ONE rn, addr, idx, d2 - fadd v\rn\().2d, v\rn\().2d, \d2\().2d - faddp d\rn\(), v\rn\().2d - str d\rn\(), [\addr, \idx, lsl #3] -.endm - -resample_one dbl, 3 - - -.macro LOAD1 d1, addr - ldr s\d1, [\addr], #4 -.endm -.macro LOAD2 d1, addr - ld1 {v\d1\().2s}, [\addr], #8 -.endm -.macro LOAD4 d1, d2, addr - ld1 {\d1\().4s}, [\addr], #16 -.endm -.macro LOAD8 d1, d2, d3, d4, addr - ld1 {\d1\().4s,\d2\().4s}, [\addr], #32 -.endm -.macro M_MLA d, r0, r1, d2:vararg - fmla \d\().4s, \r0\().4s, \r1\().4s -.endm -.macro M_MUL d, r0, r1, d2:vararg - fmul \d\().4s, \r0\().4s, \r1\().4s -.endm -.macro STORE_ONE rn, addr, idx, d2 - faddp v\rn\().4s, v\rn\().4s, v\rn\().4s - faddp s\rn\(), v\rn\().2s - str s\rn\(), [\addr, \idx, lsl #2] -.endm - -resample_one flt - - -.macro LOAD1 d1, addr - ldr h\d1, [\addr], #2 -.endm -.macro LOAD2 d1, addr - ldr s\d1, [\addr], #4 -.endm -.macro LOAD4 d1, d2, addr - ld1 {\d1\().4h}, [\addr], #8 -.endm -.macro LOAD8 d1, d2, d3, d4, addr - ld1 {\d1\().4h,\d2\().4h}, [\addr], #16 -.endm -.macro M_MLA d, r0, r1, d2:vararg - smlal \d\().4s, \r0\().4h, \r1\().4h -.endm -.macro M_MUL d, r0, r1, d2:vararg - smull \d\().4s, \r0\().4h, \r1\().4h -.endm -.macro STORE_ONE rn, addr, idx, d2 - addp v\rn\().4s, v\rn\().4s, v\rn\().4s - addp v\rn\().4s, v\rn\().4s, v\rn\().4s - sqrshrn v\rn\().4h, v\rn\().4s, #15 - str h\rn\(), [\addr, \idx, lsl #1] -.endm - -resample_one s16, 1 - - -.macro LOAD1 d1, addr - ldr s\d1, [\addr], #4 -.endm -.macro LOAD2 d1, addr - ld1 {v\d1\().2s}, [\addr], #8 -.endm -.macro LOAD4 d1, d2, addr - ld1 {\d1\().4s}, [\addr], #16 -.endm -.macro LOAD8 d1, d2, d3, d4, addr - ld1 {\d1\().4s,\d2\().4s}, [\addr], #32 -.endm -.macro M_MLA d1, r0, r1, d2:vararg - smlal \d1\().2d, \r0\().2s, \r1\().2s -.ifnb \d2 - smlal2 \d2\().2d, \r0\().4s, \r1\().4s -.endif -.endm -.macro M_MUL d1, r0, r1, d2:vararg - smull \d1\().2d, \r0\().2s, \r1\().2s -.ifnb \d2 - smull2 \d2\().2d, \r0\().4s, \r1\().4s -.endif -.endm -.macro STORE_ONE rn, addr, idx, d2 - add v\rn\().2d, v\rn\().2d, \d2\().2d - addp d\rn\(), v\rn\().2d - sqrshrn v\rn\().2s, v\rn\().2d, #30 - str s\rn\(), [\addr, \idx, lsl #2] -.endm - -resample_one s32 diff --git a/libavresample/arm/Makefile b/libavresample/arm/Makefile deleted file mode 100644 index 352d1a8c13..0000000000 --- a/libavresample/arm/Makefile +++ /dev/null @@ -1,7 +0,0 @@ -OBJS += arm/audio_convert_init.o \ - arm/resample_init.o - -OBJS-$(CONFIG_NEON_CLOBBER_TEST) += arm/neontest.o - -NEON-OBJS += arm/audio_convert_neon.o \ - arm/resample_neon.o diff --git a/libavresample/arm/asm-offsets.h b/libavresample/arm/asm-offsets.h deleted file mode 100644 index 4d3d116dc0..0000000000 --- a/libavresample/arm/asm-offsets.h +++ /dev/null @@ -1,29 +0,0 @@ -/* - * 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 - */ - -#ifndef AVRESAMPLE_ARM_ASM_OFFSETS_H -#define AVRESAMPLE_ARM_ASM_OFFSETS_H - -/* struct ResampleContext */ -#define FILTER_BANK 0x08 -#define FILTER_LENGTH 0x0c -#define SRC_INCR 0x20 -#define PHASE_SHIFT 0x28 -#define PHASE_MASK (PHASE_SHIFT + 0x04) - -#endif /* AVRESAMPLE_ARM_ASM_OFFSETS_H */ diff --git a/libavresample/arm/audio_convert_init.c b/libavresample/arm/audio_convert_init.c deleted file mode 100644 index 3d19a0e0e5..0000000000 --- a/libavresample/arm/audio_convert_init.c +++ /dev/null @@ -1,49 +0,0 @@ -/* - * 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 <stdint.h> - -#include "config.h" -#include "libavutil/attributes.h" -#include "libavutil/cpu.h" -#include "libavutil/arm/cpu.h" -#include "libavutil/samplefmt.h" -#include "libavresample/audio_convert.h" - -void ff_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len); -void ff_conv_fltp_to_s16_neon(int16_t *dst, float *const *src, - int len, int channels); -void ff_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src, - int len, int channels); - -av_cold void ff_audio_convert_init_arm(AudioConvert *ac) -{ - int cpu_flags = av_get_cpu_flags(); - - if (have_neon(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT, - 0, 16, 8, "NEON", - ff_conv_flt_to_s16_neon); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 0, 16, 8, "NEON", - ff_conv_fltp_to_s16_neon); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 2, 16, 8, "NEON", - ff_conv_fltp_to_s16_2ch_neon); - } -} diff --git a/libavresample/arm/audio_convert_neon.S b/libavresample/arm/audio_convert_neon.S deleted file mode 100644 index a120e8793b..0000000000 --- a/libavresample/arm/audio_convert_neon.S +++ /dev/null @@ -1,363 +0,0 @@ -/* - * Copyright (c) 2008 Mans Rullgard <mans@mansr.com> - * - * 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/arm/asm.S" - -function ff_conv_flt_to_s16_neon, export=1 - subs r2, r2, #8 - vld1.32 {q0}, [r1,:128]! - vcvt.s32.f32 q8, q0, #31 - vld1.32 {q1}, [r1,:128]! - vcvt.s32.f32 q9, q1, #31 - beq 3f - bics r12, r2, #15 - beq 2f -1: subs r12, r12, #16 - vqrshrn.s32 d4, q8, #16 - vld1.32 {q0}, [r1,:128]! - vcvt.s32.f32 q0, q0, #31 - vqrshrn.s32 d5, q9, #16 - vld1.32 {q1}, [r1,:128]! - vcvt.s32.f32 q1, q1, #31 - vqrshrn.s32 d6, q0, #16 - vst1.16 {q2}, [r0,:128]! - vqrshrn.s32 d7, q1, #16 - vld1.32 {q8}, [r1,:128]! - vcvt.s32.f32 q8, q8, #31 - vld1.32 {q9}, [r1,:128]! - vcvt.s32.f32 q9, q9, #31 - vst1.16 {q3}, [r0,:128]! - bne 1b - ands r2, r2, #15 - beq 3f -2: vld1.32 {q0}, [r1,:128]! - vqrshrn.s32 d4, q8, #16 - vcvt.s32.f32 q0, q0, #31 - vld1.32 {q1}, [r1,:128]! - vqrshrn.s32 d5, q9, #16 - vcvt.s32.f32 q1, q1, #31 - vqrshrn.s32 d6, q0, #16 - vst1.16 {q2}, [r0,:128]! - vqrshrn.s32 d7, q1, #16 - vst1.16 {q3}, [r0,:128]! - bx lr -3: vqrshrn.s32 d4, q8, #16 - vqrshrn.s32 d5, q9, #16 - vst1.16 {q2}, [r0,:128]! - bx lr -endfunc - -function ff_conv_fltp_to_s16_2ch_neon, export=1 - ldm r1, {r1, r3} - subs r2, r2, #8 - vld1.32 {q0}, [r1,:128]! - vcvt.s32.f32 q8, q0, #31 - vld1.32 {q1}, [r1,:128]! - vcvt.s32.f32 q9, q1, #31 - vld1.32 {q10}, [r3,:128]! - vcvt.s32.f32 q10, q10, #31 - vld1.32 {q11}, [r3,:128]! - vcvt.s32.f32 q11, q11, #31 - beq 3f - bics r12, r2, #15 - beq 2f -1: subs r12, r12, #16 - vld1.32 {q0}, [r1,:128]! - vcvt.s32.f32 q0, q0, #31 - vsri.32 q10, q8, #16 - vld1.32 {q1}, [r1,:128]! - vcvt.s32.f32 q1, q1, #31 - vld1.32 {q12}, [r3,:128]! - vcvt.s32.f32 q12, q12, #31 - vld1.32 {q13}, [r3,:128]! - vsri.32 q11, q9, #16 - vst1.16 {q10}, [r0,:128]! - vcvt.s32.f32 q13, q13, #31 - vst1.16 {q11}, [r0,:128]! - vsri.32 q12, q0, #16 - vld1.32 {q8}, [r1,:128]! - vsri.32 q13, q1, #16 - vst1.16 {q12}, [r0,:128]! - vcvt.s32.f32 q8, q8, #31 - vld1.32 {q9}, [r1,:128]! - vcvt.s32.f32 q9, q9, #31 - vld1.32 {q10}, [r3,:128]! - vcvt.s32.f32 q10, q10, #31 - vld1.32 {q11}, [r3,:128]! - vcvt.s32.f32 q11, q11, #31 - vst1.16 {q13}, [r0,:128]! - bne 1b - ands r2, r2, #15 - beq 3f -2: vsri.32 q10, q8, #16 - vld1.32 {q0}, [r1,:128]! - vcvt.s32.f32 q0, q0, #31 - vld1.32 {q1}, [r1,:128]! - vcvt.s32.f32 q1, q1, #31 - vld1.32 {q12}, [r3,:128]! - vcvt.s32.f32 q12, q12, #31 - vsri.32 q11, q9, #16 - vld1.32 {q13}, [r3,:128]! - vcvt.s32.f32 q13, q13, #31 - vst1.16 {q10}, [r0,:128]! - vsri.32 q12, q0, #16 - vst1.16 {q11}, [r0,:128]! - vsri.32 q13, q1, #16 - vst1.16 {q12-q13},[r0,:128]! - bx lr -3: vsri.32 q10, q8, #16 - vsri.32 q11, q9, #16 - vst1.16 {q10-q11},[r0,:128]! - bx lr -endfunc - -function ff_conv_fltp_to_s16_neon, export=1 - cmp r3, #2 - itt lt - ldrlt r1, [r1] - blt X(ff_conv_flt_to_s16_neon) - beq X(ff_conv_fltp_to_s16_2ch_neon) - - push {r4-r8, lr} - cmp r3, #4 - lsl r12, r3, #1 - blt 4f - - @ 4 channels -5: ldm r1!, {r4-r7} - mov lr, r2 - mov r8, r0 - vld1.32 {q8}, [r4,:128]! - vcvt.s32.f32 q8, q8, #31 - vld1.32 {q9}, [r5,:128]! - vcvt.s32.f32 q9, q9, #31 - vld1.32 {q10}, [r6,:128]! - vcvt.s32.f32 q10, q10, #31 - vld1.32 {q11}, [r7,:128]! - vcvt.s32.f32 q11, q11, #31 -6: subs lr, lr, #8 - vld1.32 {q0}, [r4,:128]! - vcvt.s32.f32 q0, q0, #31 - vsri.32 q9, q8, #16 - vld1.32 {q1}, [r5,:128]! - vcvt.s32.f32 q1, q1, #31 - vsri.32 q11, q10, #16 - vld1.32 {q2}, [r6,:128]! - vcvt.s32.f32 q2, q2, #31 - vzip.32 d18, d22 - vld1.32 {q3}, [r7,:128]! - vcvt.s32.f32 q3, q3, #31 - vzip.32 d19, d23 - vst1.16 {d18}, [r8], r12 - vsri.32 q1, q0, #16 - vst1.16 {d22}, [r8], r12 - vsri.32 q3, q2, #16 - vst1.16 {d19}, [r8], r12 - vzip.32 d2, d6 - vst1.16 {d23}, [r8], r12 - vzip.32 d3, d7 - beq 7f - vld1.32 {q8}, [r4,:128]! - vcvt.s32.f32 q8, q8, #31 - vst1.16 {d2}, [r8], r12 - vld1.32 {q9}, [r5,:128]! - vcvt.s32.f32 q9, q9, #31 - vst1.16 {d6}, [r8], r12 - vld1.32 {q10}, [r6,:128]! - vcvt.s32.f32 q10, q10, #31 - vst1.16 {d3}, [r8], r12 - vld1.32 {q11}, [r7,:128]! - vcvt.s32.f32 q11, q11, #31 - vst1.16 {d7}, [r8], r12 - b 6b -7: vst1.16 {d2}, [r8], r12 - vst1.16 {d6}, [r8], r12 - vst1.16 {d3}, [r8], r12 - vst1.16 {d7}, [r8], r12 - subs r3, r3, #4 - it eq - popeq {r4-r8, pc} - cmp r3, #4 - add r0, r0, #8 - bge 5b - - @ 2 channels -4: cmp r3, #2 - blt 4f - ldm r1!, {r4-r5} - mov lr, r2 - mov r8, r0 - tst lr, #8 - vld1.32 {q8}, [r4,:128]! - vcvt.s32.f32 q8, q8, #31 - vld1.32 {q9}, [r5,:128]! - vcvt.s32.f32 q9, q9, #31 - vld1.32 {q10}, [r4,:128]! - vcvt.s32.f32 q10, q10, #31 - vld1.32 {q11}, [r5,:128]! - vcvt.s32.f32 q11, q11, #31 - beq 6f - subs lr, lr, #8 - beq 7f - vsri.32 d18, d16, #16 - vsri.32 d19, d17, #16 - vld1.32 {q8}, [r4,:128]! - vcvt.s32.f32 q8, q8, #31 - vst1.32 {d18[0]}, [r8], r12 - vsri.32 d22, d20, #16 - vst1.32 {d18[1]}, [r8], r12 - vsri.32 d23, d21, #16 - vst1.32 {d19[0]}, [r8], r12 - vst1.32 {d19[1]}, [r8], r12 - vld1.32 {q9}, [r5,:128]! - vcvt.s32.f32 q9, q9, #31 - vst1.32 {d22[0]}, [r8], r12 - vst1.32 {d22[1]}, [r8], r12 - vld1.32 {q10}, [r4,:128]! - vcvt.s32.f32 q10, q10, #31 - vst1.32 {d23[0]}, [r8], r12 - vst1.32 {d23[1]}, [r8], r12 - vld1.32 {q11}, [r5,:128]! - vcvt.s32.f32 q11, q11, #31 -6: subs lr, lr, #16 - vld1.32 {q0}, [r4,:128]! - vcvt.s32.f32 q0, q0, #31 - vsri.32 d18, d16, #16 - vld1.32 {q1}, [r5,:128]! - vcvt.s32.f32 q1, q1, #31 - vsri.32 d19, d17, #16 - vld1.32 {q2}, [r4,:128]! - vcvt.s32.f32 q2, q2, #31 - vld1.32 {q3}, [r5,:128]! - vcvt.s32.f32 q3, q3, #31 - vst1.32 {d18[0]}, [r8], r12 - vsri.32 d22, d20, #16 - vst1.32 {d18[1]}, [r8], r12 - vsri.32 d23, d21, #16 - vst1.32 {d19[0]}, [r8], r12 - vsri.32 d2, d0, #16 - vst1.32 {d19[1]}, [r8], r12 - vsri.32 d3, d1, #16 - vst1.32 {d22[0]}, [r8], r12 - vsri.32 d6, d4, #16 - vst1.32 {d22[1]}, [r8], r12 - vsri.32 d7, d5, #16 - vst1.32 {d23[0]}, [r8], r12 - vst1.32 {d23[1]}, [r8], r12 - beq 6f - vld1.32 {q8}, [r4,:128]! - vcvt.s32.f32 q8, q8, #31 - vst1.32 {d2[0]}, [r8], r12 - vst1.32 {d2[1]}, [r8], r12 - vld1.32 {q9}, [r5,:128]! - vcvt.s32.f32 q9, q9, #31 - vst1.32 {d3[0]}, [r8], r12 - vst1.32 {d3[1]}, [r8], r12 - vld1.32 {q10}, [r4,:128]! - vcvt.s32.f32 q10, q10, #31 - vst1.32 {d6[0]}, [r8], r12 - vst1.32 {d6[1]}, [r8], r12 - vld1.32 {q11}, [r5,:128]! - vcvt.s32.f32 q11, q11, #31 - vst1.32 {d7[0]}, [r8], r12 - vst1.32 {d7[1]}, [r8], r12 - bgt 6b -6: vst1.32 {d2[0]}, [r8], r12 - vst1.32 {d2[1]}, [r8], r12 - vst1.32 {d3[0]}, [r8], r12 - vst1.32 {d3[1]}, [r8], r12 - vst1.32 {d6[0]}, [r8], r12 - vst1.32 {d6[1]}, [r8], r12 - vst1.32 {d7[0]}, [r8], r12 - vst1.32 {d7[1]}, [r8], r12 - b 8f -7: vsri.32 d18, d16, #16 - vsri.32 d19, d17, #16 - vst1.32 {d18[0]}, [r8], r12 - vsri.32 d22, d20, #16 - vst1.32 {d18[1]}, [r8], r12 - vsri.32 d23, d21, #16 - vst1.32 {d19[0]}, [r8], r12 - vst1.32 {d19[1]}, [r8], r12 - vst1.32 {d22[0]}, [r8], r12 - vst1.32 {d22[1]}, [r8], r12 - vst1.32 {d23[0]}, [r8], r12 - vst1.32 {d23[1]}, [r8], r12 -8: subs r3, r3, #2 - add r0, r0, #4 - it eq - popeq {r4-r8, pc} - - @ 1 channel -4: ldr r4, [r1] - tst r2, #8 - mov lr, r2 - mov r5, r0 - vld1.32 {q0}, [r4,:128]! - vcvt.s32.f32 q0, q0, #31 - vld1.32 {q1}, [r4,:128]! - vcvt.s32.f32 q1, q1, #31 - bne 8f -6: subs lr, lr, #16 - vld1.32 {q2}, [r4,:128]! - vcvt.s32.f32 q2, q2, #31 - vld1.32 {q3}, [r4,:128]! - vcvt.s32.f32 q3, q3, #31 - vst1.16 {d0[1]}, [r5,:16], r12 - vst1.16 {d0[3]}, [r5,:16], r12 - vst1.16 {d1[1]}, [r5,:16], r12 - vst1.16 {d1[3]}, [r5,:16], r12 - vst1.16 {d2[1]}, [r5,:16], r12 - vst1.16 {d2[3]}, [r5,:16], r12 - vst1.16 {d3[1]}, [r5,:16], r12 - vst1.16 {d3[3]}, [r5,:16], r12 - beq 7f - vld1.32 {q0}, [r4,:128]! - vcvt.s32.f32 q0, q0, #31 - vld1.32 {q1}, [r4,:128]! - vcvt.s32.f32 q1, q1, #31 -7: vst1.16 {d4[1]}, [r5,:16], r12 - vst1.16 {d4[3]}, [r5,:16], r12 - vst1.16 {d5[1]}, [r5,:16], r12 - vst1.16 {d5[3]}, [r5,:16], r12 - vst1.16 {d6[1]}, [r5,:16], r12 - vst1.16 {d6[3]}, [r5,:16], r12 - vst1.16 {d7[1]}, [r5,:16], r12 - vst1.16 {d7[3]}, [r5,:16], r12 - bgt 6b - pop {r4-r8, pc} -8: subs lr, lr, #8 - vst1.16 {d0[1]}, [r5,:16], r12 - vst1.16 {d0[3]}, [r5,:16], r12 - vst1.16 {d1[1]}, [r5,:16], r12 - vst1.16 {d1[3]}, [r5,:16], r12 - vst1.16 {d2[1]}, [r5,:16], r12 - vst1.16 {d2[3]}, [r5,:16], r12 - vst1.16 {d3[1]}, [r5,:16], r12 - vst1.16 {d3[3]}, [r5,:16], r12 - it eq - popeq {r4-r8, pc} - vld1.32 {q0}, [r4,:128]! - vcvt.s32.f32 q0, q0, #31 - vld1.32 {q1}, [r4,:128]! - vcvt.s32.f32 q1, q1, #31 - b 6b -endfunc diff --git a/libavresample/arm/neontest.c b/libavresample/arm/neontest.c deleted file mode 100644 index 22afedbc60..0000000000 --- a/libavresample/arm/neontest.c +++ /dev/null @@ -1,31 +0,0 @@ -/* - * check NEON registers for clobbers - * Copyright (c) 2013 Martin Storsjo - * - * 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 "libavresample/avresample.h" -#include "libavutil/arm/neontest.h" - -wrap(avresample_convert(AVAudioResampleContext *avr, uint8_t **output, - int out_plane_size, int out_samples, uint8_t **input, - int in_plane_size, int in_samples)) -{ - testneonclobbers(avresample_convert, avr, output, out_plane_size, - out_samples, input, in_plane_size, in_samples); -} diff --git a/libavresample/arm/resample_init.c b/libavresample/arm/resample_init.c deleted file mode 100644 index 10af09cb91..0000000000 --- a/libavresample/arm/resample_init.c +++ /dev/null @@ -1,74 +0,0 @@ -/* - * Copyright (c) 2014 Peter Meerwald <pmeerw@pmeerw.net> - * - * 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/cpu.h" -#include "libavutil/arm/cpu.h" -#include "libavutil/internal.h" -#include "libavutil/samplefmt.h" - -#include "libavresample/resample.h" - -#include "asm-offsets.h" - -AV_CHECK_OFFSET(struct ResampleContext, filter_bank, FILTER_BANK); -AV_CHECK_OFFSET(struct ResampleContext, filter_length, FILTER_LENGTH); -AV_CHECK_OFFSET(struct ResampleContext, src_incr, SRC_INCR); -AV_CHECK_OFFSET(struct ResampleContext, phase_shift, PHASE_SHIFT); -AV_CHECK_OFFSET(struct ResampleContext, phase_mask, PHASE_MASK); - -void ff_resample_one_flt_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); -void ff_resample_one_s16_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); -void ff_resample_one_s32_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); - -void ff_resample_linear_flt_neon(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); - -av_cold void ff_audio_resample_init_arm(ResampleContext *c, - enum AVSampleFormat sample_fmt) -{ - int cpu_flags = av_get_cpu_flags(); - if (have_neon(cpu_flags)) { - switch (sample_fmt) { - case AV_SAMPLE_FMT_FLTP: - if (c->linear) - c->resample_one = ff_resample_linear_flt_neon; - else - c->resample_one = ff_resample_one_flt_neon; - break; - case AV_SAMPLE_FMT_S16P: - if (!c->linear) - c->resample_one = ff_resample_one_s16_neon; - break; - case AV_SAMPLE_FMT_S32P: - if (!c->linear) - c->resample_one = ff_resample_one_s32_neon; - break; - } - } -} diff --git a/libavresample/arm/resample_neon.S b/libavresample/arm/resample_neon.S deleted file mode 100644 index 7ee8497a5c..0000000000 --- a/libavresample/arm/resample_neon.S +++ /dev/null @@ -1,358 +0,0 @@ -/* - * Copyright (c) 2014 Peter Meerwald <pmeerw@pmeerw.net> - * - * 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 "libavutil/arm/asm.S" - -#include "asm-offsets.h" - -.macro resample_one fmt, es=2 -function ff_resample_one_\fmt\()_neon, export=1 - push {r4, r5} - add r1, r1, r2, lsl #\es - - ldr r2, [r0, #PHASE_SHIFT+4] /* phase_mask */ - ldr ip, [sp, #8] /* index */ - ldr r5, [r0, #FILTER_LENGTH] - and r2, ip, r2 /* (index & phase_mask) */ - ldr r4, [r0, #PHASE_SHIFT] - lsr r4, ip, r4 /* compute sample_index */ - mul r2, r2, r5 - - ldr ip, [r0, #FILTER_BANK] - add r3, r3, r4, lsl #\es /* &src[sample_index] */ - - cmp r5, #8 - add r0, ip, r2, lsl #\es /* filter = &filter_bank[...] */ - - blt 5f -8: - subs r5, r5, #8 - LOAD4 - MUL4 -7: - LOAD4 - beq 6f - cmp r5, #8 - MLA4 - blt 4f - subs r5, r5, #8 - LOAD4 - MLA4 - b 7b -6: - MLA4 - STORE - pop {r4, r5} - bx lr -5: - INIT4 -4: /* remaining filter_length 1 to 7 */ - cmp r5, #4 - blt 2f - subs r5, r5, #4 - LOAD4 - MLA4 - beq 0f -2: /* remaining filter_length 1 to 3 */ - cmp r5, #2 - blt 1f - subs r5, r5, #2 - LOAD2 - MLA2 - beq 0f -1: /* remaining filter_length 1 */ - LOAD1 - MLA1 -0: - STORE - pop {r4, r5} - bx lr -endfunc - -.purgem LOAD1 -.purgem LOAD2 -.purgem LOAD4 -.purgem MLA1 -.purgem MLA2 -.purgem MLA4 -.purgem MUL4 -.purgem INIT4 -.purgem STORE -.endm - - -/* float32 */ -.macro LOAD1 - veor.32 d0, d0 - vld1.32 {d0[0]}, [r0]! /* load filter */ - vld1.32 {d4[0]}, [r3]! /* load src */ -.endm -.macro LOAD2 - vld1.32 {d0}, [r0]! /* load filter */ - vld1.32 {d4}, [r3]! /* load src */ -.endm -.macro LOAD4 - vld1.32 {d0,d1}, [r0]! /* load filter */ - vld1.32 {d4,d5}, [r3]! /* load src */ -.endm -.macro MLA1 - vmla.f32 d16, d0, d4[0] -.endm -.macro MLA2 - vmla.f32 d16, d0, d4 -.endm -.macro MLA4 - vmla.f32 d16, d0, d4 - vmla.f32 d17, d1, d5 -.endm -.macro MUL4 - vmul.f32 d16, d0, d4 - vmul.f32 d17, d1, d5 -.endm -.macro INIT4 - veor.f32 q8, q8 -.endm -.macro STORE - vpadd.f32 d16, d16, d17 - vpadd.f32 d16, d16, d16 - vst1.32 d16[0], [r1] -.endm - -resample_one flt, 2 - - -/* s32 */ -.macro LOAD1 - veor.32 d0, d0 - vld1.32 {d0[0]}, [r0]! /* load filter */ - vld1.32 {d4[0]}, [r3]! /* load src */ -.endm -.macro LOAD2 - vld1.32 {d0}, [r0]! /* load filter */ - vld1.32 {d4}, [r3]! /* load src */ -.endm -.macro LOAD4 - vld1.32 {d0,d1}, [r0]! /* load filter */ - vld1.32 {d4,d5}, [r3]! /* load src */ -.endm -.macro MLA1 - vmlal.s32 q8, d0, d4[0] -.endm -.macro MLA2 - vmlal.s32 q8, d0, d4 -.endm -.macro MLA4 - vmlal.s32 q8, d0, d4 - vmlal.s32 q9, d1, d5 -.endm -.macro MUL4 - vmull.s32 q8, d0, d4 - vmull.s32 q9, d1, d5 -.endm -.macro INIT4 - veor.s64 q8, q8 - veor.s64 q9, q9 -.endm -.macro STORE - vadd.s64 q8, q8, q9 - vadd.s64 d16, d16, d17 - vqrshrn.s64 d16, q8, #30 - vst1.32 d16[0], [r1] -.endm - -resample_one s32, 2 - - -/* s16 */ -.macro LOAD1 - veor.16 d0, d0 - vld1.16 {d0[0]}, [r0]! /* load filter */ - vld1.16 {d4[0]}, [r3]! /* load src */ -.endm -.macro LOAD2 - veor.16 d0, d0 - vld1.32 {d0[0]}, [r0]! /* load filter */ - veor.16 d4, d4 - vld1.32 {d4[0]}, [r3]! /* load src */ -.endm -.macro LOAD4 - vld1.16 {d0}, [r0]! /* load filter */ - vld1.16 {d4}, [r3]! /* load src */ -.endm -.macro MLA1 - vmlal.s16 q8, d0, d4[0] -.endm -.macro MLA2 - vmlal.s16 q8, d0, d4 -.endm -.macro MLA4 - vmlal.s16 q8, d0, d4 -.endm -.macro MUL4 - vmull.s16 q8, d0, d4 -.endm -.macro INIT4 - veor.s32 q8, q8 -.endm -.macro STORE - vpadd.s32 d16, d16, d17 - vpadd.s32 d16, d16, d16 - vqrshrn.s32 d16, q8, #15 - vst1.16 d16[0], [r1] -.endm - -resample_one s16, 1 - - -.macro resample_linear fmt, es=2 -function ff_resample_linear_\fmt\()_neon, export=1 - push {r4, r5} - add r1, r1, r2, lsl #\es - - ldr r2, [r0, #PHASE_SHIFT+4] /* phase_mask */ - ldr ip, [sp, #8] /* index */ - ldr r5, [r0, #FILTER_LENGTH] - and r2, ip, r2 /* (index & phase_mask) */ - ldr r4, [r0, #PHASE_SHIFT] - lsr r4, ip, r4 /* compute sample_index */ - mul r2, r2, r5 - - ldr ip, [r0, #FILTER_BANK] - add r3, r3, r4, lsl #\es /* &src[sample_index] */ - - cmp r5, #8 - ldr r4, [r0, #SRC_INCR] - add r0, ip, r2, lsl #\es /* filter = &filter_bank[...] */ - add r2, r0, r5, lsl #\es /* filter[... + c->filter_length] */ - - blt 5f -8: - subs r5, r5, #8 - LOAD4 - MUL4 -7: - LOAD4 - beq 6f - cmp r5, #8 - MLA4 - blt 4f - subs r5, r5, #8 - LOAD4 - MLA4 - b 7b -6: - MLA4 - STORE - pop {r4, r5} - bx lr -5: - INIT4 -4: /* remaining filter_length 1 to 7 */ - cmp r5, #4 - blt 2f - subs r5, r5, #4 - LOAD4 - MLA4 - beq 0f -2: /* remaining filter_length 1 to 3 */ - cmp r5, #2 - blt 1f - subs r5, r5, #2 - LOAD2 - MLA2 - beq 0f -1: /* remaining filter_length 1 */ - LOAD1 - MLA1 -0: - STORE - pop {r4, r5} - bx lr -endfunc - -.purgem LOAD1 -.purgem LOAD2 -.purgem LOAD4 -.purgem MLA1 -.purgem MLA2 -.purgem MLA4 -.purgem MUL4 -.purgem INIT4 -.purgem STORE -.endm - - -/* float32 linear */ -.macro LOAD1 - veor.32 d0, d0 - veor.32 d2, d2 - vld1.32 {d0[0]}, [r0]! /* load filter */ - vld1.32 {d2[0]}, [r2]! /* load filter */ - vld1.32 {d4[0]}, [r3]! /* load src */ -.endm -.macro LOAD2 - vld1.32 {d0}, [r0]! /* load filter */ - vld1.32 {d2}, [r2]! /* load filter */ - vld1.32 {d4}, [r3]! /* load src */ -.endm -.macro LOAD4 - vld1.32 {d0,d1}, [r0]! /* load filter */ - vld1.32 {d2,d3}, [r2]! /* load filter */ - vld1.32 {d4,d5}, [r3]! /* load src */ -.endm -.macro MLA1 - vmla.f32 d18, d0, d4[0] - vmla.f32 d16, d2, d4[0] -.endm -.macro MLA2 - vmla.f32 d18, d0, d4 - vmla.f32 d16, d2, d4 -.endm -.macro MLA4 - vmla.f32 q9, q0, q2 - vmla.f32 q8, q1, q2 -.endm -.macro MUL4 - vmul.f32 q9, q0, q2 - vmul.f32 q8, q1, q2 -.endm -.macro INIT4 - veor.f32 q9, q9 - veor.f32 q8, q8 -.endm -.macro STORE - vldr s0, [sp, #12] /* frac */ - vmov s1, r4 - vcvt.f32.s32 d0, d0 - - vsub.f32 q8, q8, q9 /* v2 - val */ - vpadd.f32 d18, d18, d19 - vpadd.f32 d16, d16, d17 - vpadd.f32 d2, d18, d18 - vpadd.f32 d1, d16, d16 - - vmul.f32 s2, s2, s0 /* (v2 - val) * frac */ - vdiv.f32 s2, s2, s1 /* / c->src_incr */ - vadd.f32 s4, s4, s2 - - vstr s4, [r1] -.endm - -resample_linear flt, 2 diff --git a/libavresample/audio_convert.c b/libavresample/audio_convert.c deleted file mode 100644 index f2888cdd17..0000000000 --- a/libavresample/audio_convert.c +++ /dev/null @@ -1,416 +0,0 @@ -/* - * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 <stdint.h> - -#include "config.h" -#include "libavutil/common.h" -#include "libavutil/libm.h" -#include "libavutil/log.h" -#include "libavutil/mem.h" -#include "libavutil/samplefmt.h" -#include "audio_convert.h" -#include "audio_data.h" -#include "dither.h" - -enum ConvFuncType { - CONV_FUNC_TYPE_FLAT, - CONV_FUNC_TYPE_INTERLEAVE, - CONV_FUNC_TYPE_DEINTERLEAVE, -}; - -typedef void (conv_func_flat)(uint8_t *out, const uint8_t *in, int len); - -typedef void (conv_func_interleave)(uint8_t *out, uint8_t *const *in, - int len, int channels); - -typedef void (conv_func_deinterleave)(uint8_t **out, const uint8_t *in, int len, - int channels); - -struct AudioConvert { - AVAudioResampleContext *avr; - DitherContext *dc; - enum AVSampleFormat in_fmt; - enum AVSampleFormat out_fmt; - int apply_map; - int channels; - int planes; - int ptr_align; - int samples_align; - int has_optimized_func; - const char *func_descr; - const char *func_descr_generic; - enum ConvFuncType func_type; - conv_func_flat *conv_flat; - conv_func_flat *conv_flat_generic; - conv_func_interleave *conv_interleave; - conv_func_interleave *conv_interleave_generic; - conv_func_deinterleave *conv_deinterleave; - conv_func_deinterleave *conv_deinterleave_generic; -}; - -void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, int channels, - int ptr_align, int samples_align, - const char *descr, void *conv) -{ - int found = 0; - - switch (ac->func_type) { - case CONV_FUNC_TYPE_FLAT: - if (av_get_packed_sample_fmt(ac->in_fmt) == in_fmt && - av_get_packed_sample_fmt(ac->out_fmt) == out_fmt) { - ac->conv_flat = conv; - ac->func_descr = descr; - ac->ptr_align = ptr_align; - ac->samples_align = samples_align; - if (ptr_align == 1 && samples_align == 1) { - ac->conv_flat_generic = conv; - ac->func_descr_generic = descr; - } else { - ac->has_optimized_func = 1; - } - found = 1; - } - break; - case CONV_FUNC_TYPE_INTERLEAVE: - if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt && - (!channels || ac->channels == channels)) { - ac->conv_interleave = conv; - ac->func_descr = descr; - ac->ptr_align = ptr_align; - ac->samples_align = samples_align; - if (ptr_align == 1 && samples_align == 1) { - ac->conv_interleave_generic = conv; - ac->func_descr_generic = descr; - } else { - ac->has_optimized_func = 1; - } - found = 1; - } - break; - case CONV_FUNC_TYPE_DEINTERLEAVE: - if (ac->in_fmt == in_fmt && ac->out_fmt == out_fmt && - (!channels || ac->channels == channels)) { - ac->conv_deinterleave = conv; - ac->func_descr = descr; - ac->ptr_align = ptr_align; - ac->samples_align = samples_align; - if (ptr_align == 1 && samples_align == 1) { - ac->conv_deinterleave_generic = conv; - ac->func_descr_generic = descr; - } else { - ac->has_optimized_func = 1; - } - found = 1; - } - break; - } - if (found) { - av_log(ac->avr, AV_LOG_DEBUG, "audio_convert: found function: %-4s " - "to %-4s (%s)\n", av_get_sample_fmt_name(ac->in_fmt), - av_get_sample_fmt_name(ac->out_fmt), descr); - } -} - -#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt - -#define CONV_LOOP(otype, expr) \ - do { \ - *(otype *)po = expr; \ - pi += is; \ - po += os; \ - } while (po < end); \ - -#define CONV_FUNC_FLAT(ofmt, otype, ifmt, itype, expr) \ -static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t *in, \ - int len) \ -{ \ - int is = sizeof(itype); \ - int os = sizeof(otype); \ - const uint8_t *pi = in; \ - uint8_t *po = out; \ - uint8_t *end = out + os * len; \ - CONV_LOOP(otype, expr) \ -} - -#define CONV_FUNC_INTERLEAVE(ofmt, otype, ifmt, itype, expr) \ -static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *out, const uint8_t **in, \ - int len, int channels) \ -{ \ - int ch; \ - int out_bps = sizeof(otype); \ - int is = sizeof(itype); \ - int os = channels * out_bps; \ - for (ch = 0; ch < channels; ch++) { \ - const uint8_t *pi = in[ch]; \ - uint8_t *po = out + ch * out_bps; \ - uint8_t *end = po + os * len; \ - CONV_LOOP(otype, expr) \ - } \ -} - -#define CONV_FUNC_DEINTERLEAVE(ofmt, otype, ifmt, itype, expr) \ -static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t **out, const uint8_t *in, \ - int len, int channels) \ -{ \ - int ch; \ - int in_bps = sizeof(itype); \ - int is = channels * in_bps; \ - int os = sizeof(otype); \ - for (ch = 0; ch < channels; ch++) { \ - const uint8_t *pi = in + ch * in_bps; \ - uint8_t *po = out[ch]; \ - uint8_t *end = po + os * len; \ - CONV_LOOP(otype, expr) \ - } \ -} - -#define CONV_FUNC_GROUP(ofmt, otype, ifmt, itype, expr) \ -CONV_FUNC_FLAT( ofmt, otype, ifmt, itype, expr) \ -CONV_FUNC_INTERLEAVE( ofmt, otype, ifmt ## P, itype, expr) \ -CONV_FUNC_DEINTERLEAVE(ofmt ## P, otype, ifmt, itype, expr) - -CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_U8, uint8_t, *(const uint8_t *)pi) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 8) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) << 24) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0f / (1 << 7))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t, (*(const uint8_t *)pi - 0x80) * (1.0 / (1 << 7))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t, (*(const int16_t *)pi >> 8) + 0x80) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi << 16) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0f / (1 << 15))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *)pi * (1.0 / (1 << 15))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t, (*(const int32_t *)pi >> 24) + 0x80) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi >> 16) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0f / (1U << 31))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *)pi * (1.0 / (1U << 31))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8( lrintf(*(const float *)pi * (1 << 7)) + 0x80)) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16( lrintf(*(const float *)pi * (1 << 15)))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *)pi * (1U << 31)))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_FLT, float, *(const float *)pi) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_FLT, float, *(const float *)pi) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8( lrint(*(const double *)pi * (1 << 7)) + 0x80)) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16( lrint(*(const double *)pi * (1 << 15)))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *)pi * (1U << 31)))) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_DBL, double, *(const double *)pi) -CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_DBL, double, *(const double *)pi) - -#define SET_CONV_FUNC_GROUP(ofmt, ifmt) \ -ff_audio_convert_set_func(ac, ofmt, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt)); \ -ff_audio_convert_set_func(ac, ofmt ## P, ifmt, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt ## P, ifmt)); \ -ff_audio_convert_set_func(ac, ofmt, ifmt ## P, 0, 1, 1, "C", CONV_FUNC_NAME(ofmt, ifmt ## P)); - -static void set_generic_function(AudioConvert *ac) -{ - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL) - SET_CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL) -} - -void ff_audio_convert_free(AudioConvert **ac) -{ - if (!*ac) - return; - ff_dither_free(&(*ac)->dc); - av_freep(ac); -} - -AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels, int sample_rate, - int apply_map) -{ - AudioConvert *ac; - int in_planar, out_planar; - - ac = av_mallocz(sizeof(*ac)); - if (!ac) - return NULL; - - ac->avr = avr; - ac->out_fmt = out_fmt; - ac->in_fmt = in_fmt; - ac->channels = channels; - ac->apply_map = apply_map; - - if (avr->dither_method != AV_RESAMPLE_DITHER_NONE && - av_get_packed_sample_fmt(out_fmt) == AV_SAMPLE_FMT_S16 && - av_get_bytes_per_sample(in_fmt) > 2) { - ac->dc = ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, - apply_map); - if (!ac->dc) { - av_free(ac); - return NULL; - } - return ac; - } - - in_planar = ff_sample_fmt_is_planar(in_fmt, channels); - out_planar = ff_sample_fmt_is_planar(out_fmt, channels); - - if (in_planar == out_planar) { - ac->func_type = CONV_FUNC_TYPE_FLAT; - ac->planes = in_planar ? ac->channels : 1; - } else if (in_planar) - ac->func_type = CONV_FUNC_TYPE_INTERLEAVE; - else - ac->func_type = CONV_FUNC_TYPE_DEINTERLEAVE; - - set_generic_function(ac); - - if (ARCH_AARCH64) - ff_audio_convert_init_aarch64(ac); - if (ARCH_ARM) - ff_audio_convert_init_arm(ac); - if (ARCH_X86) - ff_audio_convert_init_x86(ac); - - return ac; -} - -int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) -{ - int use_generic = 1; - int len = in->nb_samples; - int p; - - if (ac->dc) { - /* dithered conversion */ - av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", - len, av_get_sample_fmt_name(ac->in_fmt), - av_get_sample_fmt_name(ac->out_fmt)); - - return ff_convert_dither(ac->dc, out, in); - } - - /* determine whether to use the optimized function based on pointer and - samples alignment in both the input and output */ - if (ac->has_optimized_func) { - int ptr_align = FFMIN(in->ptr_align, out->ptr_align); - int samples_align = FFMIN(in->samples_align, out->samples_align); - int aligned_len = FFALIGN(len, ac->samples_align); - if (!(ptr_align % ac->ptr_align) && samples_align >= aligned_len) { - len = aligned_len; - use_generic = 0; - } - } - av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (%s)\n", len, - av_get_sample_fmt_name(ac->in_fmt), - av_get_sample_fmt_name(ac->out_fmt), - use_generic ? ac->func_descr_generic : ac->func_descr); - - if (ac->apply_map) { - ChannelMapInfo *map = &ac->avr->ch_map_info; - - if (!ff_sample_fmt_is_planar(ac->out_fmt, ac->channels)) { - av_log(ac->avr, AV_LOG_ERROR, "cannot remap packed format during conversion\n"); - return AVERROR(EINVAL); - } - - if (map->do_remap) { - if (ff_sample_fmt_is_planar(ac->in_fmt, ac->channels)) { - conv_func_flat *convert = use_generic ? ac->conv_flat_generic : - ac->conv_flat; - - for (p = 0; p < ac->planes; p++) - if (map->channel_map[p] >= 0) - convert(out->data[p], in->data[map->channel_map[p]], len); - } else { - uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; - conv_func_deinterleave *convert = use_generic ? - ac->conv_deinterleave_generic : - ac->conv_deinterleave; - - for (p = 0; p < ac->channels; p++) - data[map->input_map[p]] = out->data[p]; - - convert(data, in->data[0], len, ac->channels); - } - } - if (map->do_copy || map->do_zero) { - for (p = 0; p < ac->planes; p++) { - if (map->channel_copy[p]) - memcpy(out->data[p], out->data[map->channel_copy[p]], - len * out->stride); - else if (map->channel_zero[p]) - av_samples_set_silence(&out->data[p], 0, len, 1, ac->out_fmt); - } - } - } else { - switch (ac->func_type) { - case CONV_FUNC_TYPE_FLAT: { - if (!in->is_planar) - len *= in->channels; - if (use_generic) { - for (p = 0; p < ac->planes; p++) - ac->conv_flat_generic(out->data[p], in->data[p], len); - } else { - for (p = 0; p < ac->planes; p++) - ac->conv_flat(out->data[p], in->data[p], len); - } - break; - } - case CONV_FUNC_TYPE_INTERLEAVE: - if (use_generic) - ac->conv_interleave_generic(out->data[0], in->data, len, - ac->channels); - else - ac->conv_interleave(out->data[0], in->data, len, ac->channels); - break; - case CONV_FUNC_TYPE_DEINTERLEAVE: - if (use_generic) - ac->conv_deinterleave_generic(out->data, in->data[0], len, - ac->channels); - else - ac->conv_deinterleave(out->data, in->data[0], len, - ac->channels); - break; - } - } - - out->nb_samples = in->nb_samples; - return 0; -} diff --git a/libavresample/audio_convert.h b/libavresample/audio_convert.h deleted file mode 100644 index df15442c18..0000000000 --- a/libavresample/audio_convert.h +++ /dev/null @@ -1,103 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#ifndef AVRESAMPLE_AUDIO_CONVERT_H -#define AVRESAMPLE_AUDIO_CONVERT_H - -#include "libavutil/samplefmt.h" -#include "avresample.h" -#include "internal.h" -#include "audio_data.h" - -/** - * Set conversion function if the parameters match. - * - * This compares the parameters of the conversion function to the parameters - * in the AudioConvert context. If the parameters do not match, no changes are - * made to the active functions. If the parameters do match and the alignment - * is not constrained, the function is set as the generic conversion function. - * If the parameters match and the alignment is constrained, the function is - * set as the optimized conversion function. - * - * @param ac AudioConvert context - * @param out_fmt output sample format - * @param in_fmt input sample format - * @param channels number of channels, or 0 for any number of channels - * @param ptr_align buffer pointer alignment, in bytes - * @param samples_align buffer size alignment, in samples - * @param descr function type description (e.g. "C" or "SSE") - * @param conv conversion function pointer - */ -void ff_audio_convert_set_func(AudioConvert *ac, enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, int channels, - int ptr_align, int samples_align, - const char *descr, void *conv); - -/** - * Allocate and initialize AudioConvert context for sample format conversion. - * - * @param avr AVAudioResampleContext - * @param out_fmt output sample format - * @param in_fmt input sample format - * @param channels number of channels - * @param sample_rate sample rate (used for dithering) - * @param apply_map apply channel map during conversion - * @return newly-allocated AudioConvert context - */ -AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels, int sample_rate, - int apply_map); - -/** - * Free AudioConvert. - * - * The AudioConvert must have been previously allocated with ff_audio_convert_alloc(). - * - * @param ac AudioConvert struct - */ -void ff_audio_convert_free(AudioConvert **ac); - -/** - * Convert audio data from one sample format to another. - * - * For each call, the alignment of the input and output AudioData buffers are - * examined to determine whether to use the generic or optimized conversion - * function (when available). - * - * The number of samples to convert is determined by in->nb_samples. The output - * buffer must be large enough to handle this many samples. out->nb_samples is - * set by this function before a successful return. - * - * @param ac AudioConvert context - * @param out output audio data - * @param in input audio data - * @return 0 on success, negative AVERROR code on failure - */ -int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in); - -/* arch-specific initialization functions */ - -void ff_audio_convert_init_aarch64(AudioConvert *ac); -void ff_audio_convert_init_arm(AudioConvert *ac); -void ff_audio_convert_init_x86(AudioConvert *ac); - -#endif /* AVRESAMPLE_AUDIO_CONVERT_H */ diff --git a/libavresample/audio_data.c b/libavresample/audio_data.c deleted file mode 100644 index b54ead841a..0000000000 --- a/libavresample/audio_data.c +++ /dev/null @@ -1,381 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 <stdint.h> -#include <string.h> - -#include "libavutil/mem.h" -#include "audio_data.h" - -static const AVClass audio_data_class = { - .class_name = "AudioData", - .item_name = av_default_item_name, - .version = LIBAVUTIL_VERSION_INT, -}; - -/* - * Calculate alignment for data pointers. - */ -static void calc_ptr_alignment(AudioData *a) -{ - int p; - int min_align = 128; - - for (p = 0; p < a->planes; p++) { - int cur_align = 128; - while ((intptr_t)a->data[p] % cur_align) - cur_align >>= 1; - if (cur_align < min_align) - min_align = cur_align; - } - a->ptr_align = min_align; -} - -int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels) -{ - if (channels == 1) - return 1; - else - return av_sample_fmt_is_planar(sample_fmt); -} - -int ff_audio_data_set_channels(AudioData *a, int channels) -{ - if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS || - channels > a->allocated_channels) - return AVERROR(EINVAL); - - a->channels = channels; - a->planes = a->is_planar ? channels : 1; - - calc_ptr_alignment(a); - - return 0; -} - -int ff_audio_data_init(AudioData *a, uint8_t * const *src, int plane_size, - int channels, int nb_samples, - enum AVSampleFormat sample_fmt, int read_only, - const char *name) -{ - int p; - - memset(a, 0, sizeof(*a)); - a->class = &audio_data_class; - - if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels); - return AVERROR(EINVAL); - } - - a->sample_size = av_get_bytes_per_sample(sample_fmt); - if (!a->sample_size) { - av_log(a, AV_LOG_ERROR, "invalid sample format\n"); - return AVERROR(EINVAL); - } - a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels); - a->planes = a->is_planar ? channels : 1; - a->stride = a->sample_size * (a->is_planar ? 1 : channels); - - for (p = 0; p < (a->is_planar ? channels : 1); p++) { - if (!src[p]) { - av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p); - return AVERROR(EINVAL); - } - a->data[p] = src[p]; - } - a->allocated_samples = nb_samples * !read_only; - a->nb_samples = nb_samples; - a->sample_fmt = sample_fmt; - a->channels = channels; - a->allocated_channels = channels; - a->read_only = read_only; - a->allow_realloc = 0; - a->name = name ? name : "{no name}"; - - calc_ptr_alignment(a); - a->samples_align = plane_size / a->stride; - - return 0; -} - -AudioData *ff_audio_data_alloc(int channels, int nb_samples, - enum AVSampleFormat sample_fmt, const char *name) -{ - AudioData *a; - int ret; - - if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) - return NULL; - - a = av_mallocz(sizeof(*a)); - if (!a) - return NULL; - - a->sample_size = av_get_bytes_per_sample(sample_fmt); - if (!a->sample_size) { - av_free(a); - return NULL; - } - a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels); - a->planes = a->is_planar ? channels : 1; - a->stride = a->sample_size * (a->is_planar ? 1 : channels); - - a->class = &audio_data_class; - a->sample_fmt = sample_fmt; - a->channels = channels; - a->allocated_channels = channels; - a->read_only = 0; - a->allow_realloc = 1; - a->name = name ? name : "{no name}"; - - if (nb_samples > 0) { - ret = ff_audio_data_realloc(a, nb_samples); - if (ret < 0) { - av_free(a); - return NULL; - } - return a; - } else { - calc_ptr_alignment(a); - return a; - } -} - -int ff_audio_data_realloc(AudioData *a, int nb_samples) -{ - int ret, new_buf_size, plane_size, p; - - /* check if buffer is already large enough */ - if (a->allocated_samples >= nb_samples) - return 0; - - /* validate that the output is not read-only and realloc is allowed */ - if (a->read_only || !a->allow_realloc) - return AVERROR(EINVAL); - - new_buf_size = av_samples_get_buffer_size(&plane_size, - a->allocated_channels, nb_samples, - a->sample_fmt, 0); - if (new_buf_size < 0) - return new_buf_size; - - /* if there is already data in the buffer and the sample format is planar, - allocate a new buffer and copy the data, otherwise just realloc the - internal buffer and set new data pointers */ - if (a->nb_samples > 0 && a->is_planar) { - uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL }; - - ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels, - nb_samples, a->sample_fmt, 0); - if (ret < 0) - return ret; - - for (p = 0; p < a->planes; p++) - memcpy(new_data[p], a->data[p], a->nb_samples * a->stride); - - av_freep(&a->buffer); - memcpy(a->data, new_data, sizeof(new_data)); - a->buffer = a->data[0]; - } else { - av_freep(&a->buffer); - a->buffer = av_malloc(new_buf_size); - if (!a->buffer) - return AVERROR(ENOMEM); - ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer, - a->allocated_channels, nb_samples, - a->sample_fmt, 0); - if (ret < 0) - return ret; - } - a->buffer_size = new_buf_size; - a->allocated_samples = nb_samples; - - calc_ptr_alignment(a); - a->samples_align = plane_size / a->stride; - - return 0; -} - -void ff_audio_data_free(AudioData **a) -{ - if (!*a) - return; - av_free((*a)->buffer); - av_freep(a); -} - -int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map) -{ - int ret, p; - - /* validate input/output compatibility */ - if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels) - return AVERROR(EINVAL); - - if (map && !src->is_planar) { - av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n"); - return AVERROR(EINVAL); - } - - /* if the input is empty, just empty the output */ - if (!src->nb_samples) { - dst->nb_samples = 0; - return 0; - } - - /* reallocate output if necessary */ - ret = ff_audio_data_realloc(dst, src->nb_samples); - if (ret < 0) - return ret; - - /* copy data */ - if (map) { - if (map->do_remap) { - for (p = 0; p < src->planes; p++) { - if (map->channel_map[p] >= 0) - memcpy(dst->data[p], src->data[map->channel_map[p]], - src->nb_samples * src->stride); - } - } - if (map->do_copy || map->do_zero) { - for (p = 0; p < src->planes; p++) { - if (map->channel_copy[p]) - memcpy(dst->data[p], dst->data[map->channel_copy[p]], - src->nb_samples * src->stride); - else if (map->channel_zero[p]) - av_samples_set_silence(&dst->data[p], 0, src->nb_samples, - 1, dst->sample_fmt); - } - } - } else { - for (p = 0; p < src->planes; p++) - memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride); - } - - dst->nb_samples = src->nb_samples; - - return 0; -} - -int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, - int src_offset, int nb_samples) -{ - int ret, p, dst_offset2, dst_move_size; - - /* validate input/output compatibility */ - if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) { - av_log(src, AV_LOG_ERROR, "sample format mismatch\n"); - return AVERROR(EINVAL); - } - - /* validate offsets are within the buffer bounds */ - if (dst_offset < 0 || dst_offset > dst->nb_samples || - src_offset < 0 || src_offset > src->nb_samples) { - av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n", - src_offset, dst_offset); - return AVERROR(EINVAL); - } - - /* check offsets and sizes to see if we can just do nothing and return */ - if (nb_samples > src->nb_samples - src_offset) - nb_samples = src->nb_samples - src_offset; - if (nb_samples <= 0) - return 0; - - /* validate that the output is not read-only */ - if (dst->read_only) { - av_log(dst, AV_LOG_ERROR, "dst is read-only\n"); - return AVERROR(EINVAL); - } - - /* reallocate output if necessary */ - ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples); - if (ret < 0) { - av_log(dst, AV_LOG_ERROR, "error reallocating dst\n"); - return ret; - } - - dst_offset2 = dst_offset + nb_samples; - dst_move_size = dst->nb_samples - dst_offset; - - for (p = 0; p < src->planes; p++) { - if (dst_move_size > 0) { - memmove(dst->data[p] + dst_offset2 * dst->stride, - dst->data[p] + dst_offset * dst->stride, - dst_move_size * dst->stride); - } - memcpy(dst->data[p] + dst_offset * dst->stride, - src->data[p] + src_offset * src->stride, - nb_samples * src->stride); - } - dst->nb_samples += nb_samples; - - return 0; -} - -void ff_audio_data_drain(AudioData *a, int nb_samples) -{ - if (a->nb_samples <= nb_samples) { - /* drain the whole buffer */ - a->nb_samples = 0; - } else { - int p; - int move_offset = a->stride * nb_samples; - int move_size = a->stride * (a->nb_samples - nb_samples); - - for (p = 0; p < a->planes; p++) - memmove(a->data[p], a->data[p] + move_offset, move_size); - - a->nb_samples -= nb_samples; - } -} - -int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, - int nb_samples) -{ - uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS]; - int offset_size, p; - - if (offset >= a->nb_samples) - return 0; - offset_size = offset * a->stride; - for (p = 0; p < a->planes; p++) - offset_data[p] = a->data[p] + offset_size; - - return av_audio_fifo_write(af, (void **)offset_data, nb_samples); -} - -int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples) -{ - int ret; - - if (a->read_only) - return AVERROR(EINVAL); - - ret = ff_audio_data_realloc(a, nb_samples); - if (ret < 0) - return ret; - - ret = av_audio_fifo_read(af, (void **)a->data, nb_samples); - if (ret >= 0) - a->nb_samples = ret; - return ret; -} diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h deleted file mode 100644 index 1280307a95..0000000000 --- a/libavresample/audio_data.h +++ /dev/null @@ -1,178 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#ifndef AVRESAMPLE_AUDIO_DATA_H -#define AVRESAMPLE_AUDIO_DATA_H - -#include <stdint.h> - -#include "libavutil/audio_fifo.h" -#include "libavutil/log.h" -#include "libavutil/samplefmt.h" -#include "avresample.h" -#include "internal.h" - -int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels); - -/** - * Audio buffer used for intermediate storage between conversion phases. - */ -struct AudioData { - const AVClass *class; /**< AVClass for logging */ - uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */ - uint8_t *buffer; /**< data buffer */ - unsigned int buffer_size; /**< allocated buffer size */ - int allocated_samples; /**< number of samples the buffer can hold */ - int nb_samples; /**< current number of samples */ - enum AVSampleFormat sample_fmt; /**< sample format */ - int channels; /**< channel count */ - int allocated_channels; /**< allocated channel count */ - int is_planar; /**< sample format is planar */ - int planes; /**< number of data planes */ - int sample_size; /**< bytes per sample */ - int stride; /**< sample byte offset within a plane */ - int read_only; /**< data is read-only */ - int allow_realloc; /**< realloc is allowed */ - int ptr_align; /**< minimum data pointer alignment */ - int samples_align; /**< allocated samples alignment */ - const char *name; /**< name for debug logging */ -}; - -int ff_audio_data_set_channels(AudioData *a, int channels); - -/** - * Initialize AudioData using a given source. - * - * This does not allocate an internal buffer. It only sets the data pointers - * and audio parameters. - * - * @param a AudioData struct - * @param src source data pointers - * @param plane_size plane size, in bytes. - * This can be 0 if unknown, but that will lead to - * optimized functions not being used in many cases, - * which could slow down some conversions. - * @param channels channel count - * @param nb_samples number of samples in the source data - * @param sample_fmt sample format - * @param read_only indicates if buffer is read only or read/write - * @param name name for debug logging (can be NULL) - * @return 0 on success, negative AVERROR value on error - */ -int ff_audio_data_init(AudioData *a, uint8_t * const *src, int plane_size, - int channels, int nb_samples, - enum AVSampleFormat sample_fmt, int read_only, - const char *name); - -/** - * Allocate AudioData. - * - * This allocates an internal buffer and sets audio parameters. - * - * @param channels channel count - * @param nb_samples number of samples to allocate space for - * @param sample_fmt sample format - * @param name name for debug logging (can be NULL) - * @return newly allocated AudioData struct, or NULL on error - */ -AudioData *ff_audio_data_alloc(int channels, int nb_samples, - enum AVSampleFormat sample_fmt, - const char *name); - -/** - * Reallocate AudioData. - * - * The AudioData must have been previously allocated with ff_audio_data_alloc(). - * - * @param a AudioData struct - * @param nb_samples number of samples to allocate space for - * @return 0 on success, negative AVERROR value on error - */ -int ff_audio_data_realloc(AudioData *a, int nb_samples); - -/** - * Free AudioData. - * - * The AudioData must have been previously allocated with ff_audio_data_alloc(). - * - * @param a AudioData struct - */ -void ff_audio_data_free(AudioData **a); - -/** - * Copy data from one AudioData to another. - * - * @param out output AudioData - * @param in input AudioData - * @param map channel map, NULL if not remapping - * @return 0 on success, negative AVERROR value on error - */ -int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map); - -/** - * Append data from one AudioData to the end of another. - * - * @param dst destination AudioData - * @param dst_offset offset, in samples, to start writing, relative to the - * start of dst - * @param src source AudioData - * @param src_offset offset, in samples, to start copying, relative to the - * start of the src - * @param nb_samples number of samples to copy - * @return 0 on success, negative AVERROR value on error - */ -int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, - int src_offset, int nb_samples); - -/** - * Drain samples from the start of the AudioData. - * - * Remaining samples are shifted to the start of the AudioData. - * - * @param a AudioData struct - * @param nb_samples number of samples to drain - */ -void ff_audio_data_drain(AudioData *a, int nb_samples); - -/** - * Add samples in AudioData to an AVAudioFifo. - * - * @param af Audio FIFO Buffer - * @param a AudioData struct - * @param offset number of samples to skip from the start of the data - * @param nb_samples number of samples to add to the FIFO - * @return number of samples actually added to the FIFO, or - * negative AVERROR code on error - */ -int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, - int nb_samples); - -/** - * Read samples from an AVAudioFifo to AudioData. - * - * @param af Audio FIFO Buffer - * @param a AudioData struct - * @param nb_samples number of samples to read from the FIFO - * @return number of samples actually read from the FIFO, or - * negative AVERROR code on error - */ -int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples); - -#endif /* AVRESAMPLE_AUDIO_DATA_H */ diff --git a/libavresample/audio_mix.c b/libavresample/audio_mix.c deleted file mode 100644 index 7ae0aeb74d..0000000000 --- a/libavresample/audio_mix.c +++ /dev/null @@ -1,742 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 <stdint.h> - -#include "libavutil/common.h" -#include "libavutil/libm.h" -#include "libavutil/samplefmt.h" -#include "avresample.h" -#include "internal.h" -#include "audio_data.h" -#include "audio_mix.h" - -static const char * const coeff_type_names[] = { "q8", "q15", "flt" }; - -struct AudioMix { - AVAudioResampleContext *avr; - enum AVSampleFormat fmt; - enum AVMixCoeffType coeff_type; - uint64_t in_layout; - uint64_t out_layout; - int in_channels; - int out_channels; - - int ptr_align; - int samples_align; - int has_optimized_func; - const char *func_descr; - const char *func_descr_generic; - mix_func *mix; - mix_func *mix_generic; - - int in_matrix_channels; - int out_matrix_channels; - int output_zero[AVRESAMPLE_MAX_CHANNELS]; - int input_skip[AVRESAMPLE_MAX_CHANNELS]; - int output_skip[AVRESAMPLE_MAX_CHANNELS]; - int16_t *matrix_q8[AVRESAMPLE_MAX_CHANNELS]; - int32_t *matrix_q15[AVRESAMPLE_MAX_CHANNELS]; - float *matrix_flt[AVRESAMPLE_MAX_CHANNELS]; - void **matrix; -}; - -void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt, - enum AVMixCoeffType coeff_type, int in_channels, - int out_channels, int ptr_align, int samples_align, - const char *descr, void *mix_func) -{ - if (fmt == am->fmt && coeff_type == am->coeff_type && - ( in_channels == am->in_matrix_channels || in_channels == 0) && - (out_channels == am->out_matrix_channels || out_channels == 0)) { - char chan_str[16]; - am->mix = mix_func; - am->func_descr = descr; - am->ptr_align = ptr_align; - am->samples_align = samples_align; - if (ptr_align == 1 && samples_align == 1) { - am->mix_generic = mix_func; - am->func_descr_generic = descr; - } else { - am->has_optimized_func = 1; - } - if (in_channels) { - if (out_channels) - snprintf(chan_str, sizeof(chan_str), "[%d to %d] ", - in_channels, out_channels); - else - snprintf(chan_str, sizeof(chan_str), "[%d to any] ", - in_channels); - } else if (out_channels) { - snprintf(chan_str, sizeof(chan_str), "[any to %d] ", - out_channels); - } else { - snprintf(chan_str, sizeof(chan_str), "[any to any] "); - } - av_log(am->avr, AV_LOG_DEBUG, "audio_mix: found function: [fmt=%s] " - "[c=%s] %s(%s)\n", av_get_sample_fmt_name(fmt), - coeff_type_names[coeff_type], chan_str, descr); - } -} - -#define MIX_FUNC_NAME(fmt, cfmt) mix_any_ ## fmt ##_## cfmt ##_c - -#define MIX_FUNC_GENERIC(fmt, cfmt, stype, ctype, sumtype, expr) \ -static void MIX_FUNC_NAME(fmt, cfmt)(stype **samples, ctype **matrix, \ - int len, int out_ch, int in_ch) \ -{ \ - int i, in, out; \ - stype temp[AVRESAMPLE_MAX_CHANNELS]; \ - for (i = 0; i < len; i++) { \ - for (out = 0; out < out_ch; out++) { \ - sumtype sum = 0; \ - for (in = 0; in < in_ch; in++) \ - sum += samples[in][i] * matrix[out][in]; \ - temp[out] = expr; \ - } \ - for (out = 0; out < out_ch; out++) \ - samples[out][i] = temp[out]; \ - } \ -} - -MIX_FUNC_GENERIC(FLTP, FLT, float, float, float, sum) -MIX_FUNC_GENERIC(S16P, FLT, int16_t, float, float, av_clip_int16(lrintf(sum))) -MIX_FUNC_GENERIC(S16P, Q15, int16_t, int32_t, int64_t, av_clip_int16(sum >> 15)) -MIX_FUNC_GENERIC(S16P, Q8, int16_t, int16_t, int32_t, av_clip_int16(sum >> 8)) - -/* TODO: templatize the channel-specific C functions */ - -static void mix_2_to_1_fltp_flt_c(float **samples, float **matrix, int len, - int out_ch, int in_ch) -{ - float *src0 = samples[0]; - float *src1 = samples[1]; - float *dst = src0; - float m0 = matrix[0][0]; - float m1 = matrix[0][1]; - - while (len > 4) { - *dst++ = *src0++ * m0 + *src1++ * m1; - *dst++ = *src0++ * m0 + *src1++ * m1; - *dst++ = *src0++ * m0 + *src1++ * m1; - *dst++ = *src0++ * m0 + *src1++ * m1; - len -= 4; - } - while (len > 0) { - *dst++ = *src0++ * m0 + *src1++ * m1; - len--; - } -} - -static void mix_2_to_1_s16p_flt_c(int16_t **samples, float **matrix, int len, - int out_ch, int in_ch) -{ - int16_t *src0 = samples[0]; - int16_t *src1 = samples[1]; - int16_t *dst = src0; - float m0 = matrix[0][0]; - float m1 = matrix[0][1]; - - while (len > 4) { - *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); - *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); - *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); - *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); - len -= 4; - } - while (len > 0) { - *dst++ = av_clip_int16(lrintf(*src0++ * m0 + *src1++ * m1)); - len--; - } -} - -static void mix_2_to_1_s16p_q8_c(int16_t **samples, int16_t **matrix, int len, - int out_ch, int in_ch) -{ - int16_t *src0 = samples[0]; - int16_t *src1 = samples[1]; - int16_t *dst = src0; - int16_t m0 = matrix[0][0]; - int16_t m1 = matrix[0][1]; - - while (len > 4) { - *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8; - *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8; - *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8; - *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8; - len -= 4; - } - while (len > 0) { - *dst++ = (*src0++ * m0 + *src1++ * m1) >> 8; - len--; - } -} - -static void mix_1_to_2_fltp_flt_c(float **samples, float **matrix, int len, - int out_ch, int in_ch) -{ - float v; - float *dst0 = samples[0]; - float *dst1 = samples[1]; - float *src = dst0; - float m0 = matrix[0][0]; - float m1 = matrix[1][0]; - - while (len > 4) { - v = *src++; - *dst0++ = v * m0; - *dst1++ = v * m1; - v = *src++; - *dst0++ = v * m0; - *dst1++ = v * m1; - v = *src++; - *dst0++ = v * m0; - *dst1++ = v * m1; - v = *src++; - *dst0++ = v * m0; - *dst1++ = v * m1; - len -= 4; - } - while (len > 0) { - v = *src++; - *dst0++ = v * m0; - *dst1++ = v * m1; - len--; - } -} - -static void mix_6_to_2_fltp_flt_c(float **samples, float **matrix, int len, - int out_ch, int in_ch) -{ - float v0, v1; - float *src0 = samples[0]; - float *src1 = samples[1]; - float *src2 = samples[2]; - float *src3 = samples[3]; - float *src4 = samples[4]; - float *src5 = samples[5]; - float *dst0 = src0; - float *dst1 = src1; - float *m0 = matrix[0]; - float *m1 = matrix[1]; - - while (len > 0) { - v0 = *src0++; - v1 = *src1++; - *dst0++ = v0 * m0[0] + - v1 * m0[1] + - *src2 * m0[2] + - *src3 * m0[3] + - *src4 * m0[4] + - *src5 * m0[5]; - *dst1++ = v0 * m1[0] + - v1 * m1[1] + - *src2++ * m1[2] + - *src3++ * m1[3] + - *src4++ * m1[4] + - *src5++ * m1[5]; - len--; - } -} - -static void mix_2_to_6_fltp_flt_c(float **samples, float **matrix, int len, - int out_ch, int in_ch) -{ - float v0, v1; - float *dst0 = samples[0]; - float *dst1 = samples[1]; - float *dst2 = samples[2]; - float *dst3 = samples[3]; - float *dst4 = samples[4]; - float *dst5 = samples[5]; - float *src0 = dst0; - float *src1 = dst1; - - while (len > 0) { - v0 = *src0++; - v1 = *src1++; - *dst0++ = v0 * matrix[0][0] + v1 * matrix[0][1]; - *dst1++ = v0 * matrix[1][0] + v1 * matrix[1][1]; - *dst2++ = v0 * matrix[2][0] + v1 * matrix[2][1]; - *dst3++ = v0 * matrix[3][0] + v1 * matrix[3][1]; - *dst4++ = v0 * matrix[4][0] + v1 * matrix[4][1]; - *dst5++ = v0 * matrix[5][0] + v1 * matrix[5][1]; - len--; - } -} - -static av_cold int mix_function_init(AudioMix *am) -{ - am->func_descr = am->func_descr_generic = "n/a"; - am->mix = am->mix_generic = NULL; - - /* no need to set a mix function when we're skipping mixing */ - if (!am->in_matrix_channels || !am->out_matrix_channels) - return 0; - - /* any-to-any C versions */ - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 0, 0, 1, 1, "C", MIX_FUNC_NAME(FLTP, FLT)); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, FLT)); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q15, - 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q15)); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8, - 0, 0, 1, 1, "C", MIX_FUNC_NAME(S16P, Q8)); - - /* channel-specific C versions */ - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 2, 1, 1, 1, "C", mix_2_to_1_fltp_flt_c); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 2, 1, 1, 1, "C", mix_2_to_1_s16p_flt_c); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8, - 2, 1, 1, 1, "C", mix_2_to_1_s16p_q8_c); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 1, 2, 1, 1, "C", mix_1_to_2_fltp_flt_c); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 6, 2, 1, 1, "C", mix_6_to_2_fltp_flt_c); - - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 2, 6, 1, 1, "C", mix_2_to_6_fltp_flt_c); - - if (ARCH_X86) - ff_audio_mix_init_x86(am); - - if (!am->mix) { - av_log(am->avr, AV_LOG_ERROR, "audio_mix: NO FUNCTION FOUND: [fmt=%s] " - "[c=%s] [%d to %d]\n", av_get_sample_fmt_name(am->fmt), - coeff_type_names[am->coeff_type], am->in_channels, - am->out_channels); - return AVERROR_PATCHWELCOME; - } - return 0; -} - -AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr) -{ - AudioMix *am; - int ret; - - am = av_mallocz(sizeof(*am)); - if (!am) - return NULL; - am->avr = avr; - - if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P && - avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) { - av_log(avr, AV_LOG_ERROR, "Unsupported internal format for " - "mixing: %s\n", - av_get_sample_fmt_name(avr->internal_sample_fmt)); - goto error; - } - - am->fmt = avr->internal_sample_fmt; - am->coeff_type = avr->mix_coeff_type; - am->in_layout = avr->in_channel_layout; - am->out_layout = avr->out_channel_layout; - am->in_channels = avr->in_channels; - am->out_channels = avr->out_channels; - - /* build matrix if the user did not already set one */ - if (avr->mix_matrix) { - ret = ff_audio_mix_set_matrix(am, avr->mix_matrix, avr->in_channels); - if (ret < 0) - goto error; - av_freep(&avr->mix_matrix); - } else { - double *matrix_dbl = av_mallocz(avr->out_channels * avr->in_channels * - sizeof(*matrix_dbl)); - if (!matrix_dbl) - goto error; - - ret = avresample_build_matrix(avr->in_channel_layout, - avr->out_channel_layout, - avr->center_mix_level, - avr->surround_mix_level, - avr->lfe_mix_level, - avr->normalize_mix_level, - matrix_dbl, - avr->in_channels, - avr->matrix_encoding); - if (ret < 0) { - av_free(matrix_dbl); - goto error; - } - - ret = ff_audio_mix_set_matrix(am, matrix_dbl, avr->in_channels); - if (ret < 0) { - av_log(avr, AV_LOG_ERROR, "error setting mix matrix\n"); - av_free(matrix_dbl); - goto error; - } - - av_free(matrix_dbl); - } - - return am; - -error: - av_free(am); - return NULL; -} - -void ff_audio_mix_free(AudioMix **am_p) -{ - AudioMix *am; - - if (!*am_p) - return; - am = *am_p; - - if (am->matrix) { - av_free(am->matrix[0]); - am->matrix = NULL; - } - memset(am->matrix_q8, 0, sizeof(am->matrix_q8 )); - memset(am->matrix_q15, 0, sizeof(am->matrix_q15)); - memset(am->matrix_flt, 0, sizeof(am->matrix_flt)); - - av_freep(am_p); -} - -int ff_audio_mix(AudioMix *am, AudioData *src) -{ - int use_generic = 1; - int len = src->nb_samples; - int i, j; - - /* determine whether to use the optimized function based on pointer and - samples alignment in both the input and output */ - if (am->has_optimized_func) { - int aligned_len = FFALIGN(len, am->samples_align); - if (!(src->ptr_align % am->ptr_align) && - src->samples_align >= aligned_len) { - len = aligned_len; - use_generic = 0; - } - } - av_log(am->avr, AV_LOG_TRACE, "audio_mix: %d samples - %d to %d channels (%s)\n", - src->nb_samples, am->in_channels, am->out_channels, - use_generic ? am->func_descr_generic : am->func_descr); - - if (am->in_matrix_channels && am->out_matrix_channels) { - uint8_t **data; - uint8_t *data0[AVRESAMPLE_MAX_CHANNELS] = { NULL }; - - if (am->out_matrix_channels < am->out_channels || - am->in_matrix_channels < am->in_channels) { - for (i = 0, j = 0; i < FFMAX(am->in_channels, am->out_channels); i++) { - if (am->input_skip[i] || am->output_skip[i] || am->output_zero[i]) - continue; - data0[j++] = src->data[i]; - } - data = data0; - } else { - data = src->data; - } - - if (use_generic) - am->mix_generic(data, am->matrix, len, am->out_matrix_channels, - am->in_matrix_channels); - else - am->mix(data, am->matrix, len, am->out_matrix_channels, - am->in_matrix_channels); - } - - if (am->out_matrix_channels < am->out_channels) { - for (i = 0; i < am->out_channels; i++) - if (am->output_zero[i]) - av_samples_set_silence(&src->data[i], 0, len, 1, am->fmt); - } - - ff_audio_data_set_channels(src, am->out_channels); - - return 0; -} - -int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride) -{ - int i, o, i0, o0; - - if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS || - am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n"); - return AVERROR(EINVAL); - } - -#define GET_MATRIX_CONVERT(suffix, scale) \ - if (!am->matrix_ ## suffix[0]) { \ - av_log(am->avr, AV_LOG_ERROR, "matrix is not set\n"); \ - return AVERROR(EINVAL); \ - } \ - for (o = 0, o0 = 0; o < am->out_channels; o++) { \ - for (i = 0, i0 = 0; i < am->in_channels; i++) { \ - if (am->input_skip[i] || am->output_zero[o]) \ - matrix[o * stride + i] = 0.0; \ - else \ - matrix[o * stride + i] = am->matrix_ ## suffix[o0][i0] * \ - (scale); \ - if (!am->input_skip[i]) \ - i0++; \ - } \ - if (!am->output_zero[o]) \ - o0++; \ - } - - switch (am->coeff_type) { - case AV_MIX_COEFF_TYPE_Q8: - GET_MATRIX_CONVERT(q8, 1.0 / 256.0); - break; - case AV_MIX_COEFF_TYPE_Q15: - GET_MATRIX_CONVERT(q15, 1.0 / 32768.0); - break; - case AV_MIX_COEFF_TYPE_FLT: - GET_MATRIX_CONVERT(flt, 1.0); - break; - default: - av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); - return AVERROR(EINVAL); - } - - return 0; -} - -static void reduce_matrix(AudioMix *am, const double *matrix, int stride) -{ - int i, o; - - memset(am->output_zero, 0, sizeof(am->output_zero)); - memset(am->input_skip, 0, sizeof(am->input_skip)); - memset(am->output_skip, 0, sizeof(am->output_skip)); - - /* exclude output channels if they can be zeroed instead of mixed */ - for (o = 0; o < am->out_channels; o++) { - int zero = 1; - - /* check if the output is always silent */ - for (i = 0; i < am->in_channels; i++) { - if (matrix[o * stride + i] != 0.0) { - zero = 0; - break; - } - } - /* check if the corresponding input channel makes a contribution to - any output channel */ - if (o < am->in_channels) { - for (i = 0; i < am->out_channels; i++) { - if (matrix[i * stride + o] != 0.0) { - zero = 0; - break; - } - } - } - if (zero) { - am->output_zero[o] = 1; - am->out_matrix_channels--; - if (o < am->in_channels) - am->in_matrix_channels--; - } - } - if (am->out_matrix_channels == 0 || am->in_matrix_channels == 0) { - am->out_matrix_channels = 0; - am->in_matrix_channels = 0; - return; - } - - /* skip input channels that contribute fully only to the corresponding - output channel */ - for (i = 0; i < FFMIN(am->in_channels, am->out_channels); i++) { - int skip = 1; - - for (o = 0; o < am->out_channels; o++) { - int i0; - if ((o != i && matrix[o * stride + i] != 0.0) || - (o == i && matrix[o * stride + i] != 1.0)) { - skip = 0; - break; - } - /* if the input contributes fully to the output, also check that no - other inputs contribute to this output */ - if (o == i) { - for (i0 = 0; i0 < am->in_channels; i0++) { - if (i0 != i && matrix[o * stride + i0] != 0.0) { - skip = 0; - break; - } - } - } - } - if (skip) { - am->input_skip[i] = 1; - am->in_matrix_channels--; - } - } - /* skip input channels that do not contribute to any output channel */ - for (; i < am->in_channels; i++) { - int contrib = 0; - - for (o = 0; o < am->out_channels; o++) { - if (matrix[o * stride + i] != 0.0) { - contrib = 1; - break; - } - } - if (!contrib) { - am->input_skip[i] = 1; - am->in_matrix_channels--; - } - } - if (am->in_matrix_channels == 0) { - am->out_matrix_channels = 0; - return; - } - - /* skip output channels that only get full contribution from the - corresponding input channel */ - for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) { - int skip = 1; - int o0; - - for (i = 0; i < am->in_channels; i++) { - if ((o != i && matrix[o * stride + i] != 0.0) || - (o == i && matrix[o * stride + i] != 1.0)) { - skip = 0; - break; - } - } - /* check if the corresponding input channel makes a contribution to - any other output channel */ - i = o; - for (o0 = 0; o0 < am->out_channels; o0++) { - if (o0 != i && matrix[o0 * stride + i] != 0.0) { - skip = 0; - break; - } - } - if (skip) { - am->output_skip[o] = 1; - am->out_matrix_channels--; - } - } - if (am->out_matrix_channels == 0) { - am->in_matrix_channels = 0; - return; - } -} - -int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride) -{ - int i, o, i0, o0, ret; - char in_layout_name[128]; - char out_layout_name[128]; - - if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS || - am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n"); - return AVERROR(EINVAL); - } - - if (am->matrix) { - av_free(am->matrix[0]); - am->matrix = NULL; - } - - am->in_matrix_channels = am->in_channels; - am->out_matrix_channels = am->out_channels; - - reduce_matrix(am, matrix, stride); - -#define CONVERT_MATRIX(type, expr) \ - am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \ - am->in_matrix_channels * \ - sizeof(*am->matrix_## type[0])); \ - if (!am->matrix_## type[0]) \ - return AVERROR(ENOMEM); \ - for (o = 0, o0 = 0; o < am->out_channels; o++) { \ - if (am->output_zero[o] || am->output_skip[o]) \ - continue; \ - if (o0 > 0) \ - am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \ - am->in_matrix_channels; \ - for (i = 0, i0 = 0; i < am->in_channels; i++) { \ - double v; \ - if (am->input_skip[i] || am->output_zero[i]) \ - continue; \ - v = matrix[o * stride + i]; \ - am->matrix_## type[o0][i0] = expr; \ - i0++; \ - } \ - o0++; \ - } \ - am->matrix = (void **)am->matrix_## type; - - if (am->in_matrix_channels && am->out_matrix_channels) { - switch (am->coeff_type) { - case AV_MIX_COEFF_TYPE_Q8: - CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) - break; - case AV_MIX_COEFF_TYPE_Q15: - CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) - break; - case AV_MIX_COEFF_TYPE_FLT: - CONVERT_MATRIX(flt, v) - break; - default: - av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); - return AVERROR(EINVAL); - } - } - - ret = mix_function_init(am); - if (ret < 0) - return ret; - - av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name), - am->in_channels, am->in_layout); - av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name), - am->out_channels, am->out_layout); - av_log(am->avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n", - in_layout_name, out_layout_name); - av_log(am->avr, AV_LOG_DEBUG, "matrix size: %d x %d\n", - am->in_matrix_channels, am->out_matrix_channels); - for (o = 0; o < am->out_channels; o++) { - for (i = 0; i < am->in_channels; i++) { - if (am->output_zero[o]) - av_log(am->avr, AV_LOG_DEBUG, " (ZERO)"); - else if (am->input_skip[i] || am->output_zero[i] || am->output_skip[o]) - av_log(am->avr, AV_LOG_DEBUG, " (SKIP)"); - else - av_log(am->avr, AV_LOG_DEBUG, " %0.3f ", - matrix[o * am->in_channels + i]); - } - av_log(am->avr, AV_LOG_DEBUG, "\n"); - } - - return 0; -} diff --git a/libavresample/audio_mix.h b/libavresample/audio_mix.h deleted file mode 100644 index 0187d0f9e0..0000000000 --- a/libavresample/audio_mix.h +++ /dev/null @@ -1,94 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#ifndef AVRESAMPLE_AUDIO_MIX_H -#define AVRESAMPLE_AUDIO_MIX_H - -#include <stdint.h> - -#include "libavutil/samplefmt.h" -#include "avresample.h" -#include "internal.h" -#include "audio_data.h" - -typedef void (mix_func)(uint8_t **src, void **matrix, int len, int out_ch, - int in_ch); - -/** - * Set mixing function if the parameters match. - * - * This compares the parameters of the mixing function to the parameters in the - * AudioMix context. If the parameters do not match, no changes are made to the - * active functions. If the parameters do match and the alignment is not - * constrained, the function is set as the generic mixing function. If the - * parameters match and the alignment is constrained, the function is set as - * the optimized mixing function. - * - * @param am AudioMix context - * @param fmt input/output sample format - * @param coeff_type mixing coefficient type - * @param in_channels number of input channels, or 0 for any number of channels - * @param out_channels number of output channels, or 0 for any number of channels - * @param ptr_align buffer pointer alignment, in bytes - * @param samples_align buffer size alignment, in samples - * @param descr function type description (e.g. "C" or "SSE") - * @param mix_func mixing function pointer - */ -void ff_audio_mix_set_func(AudioMix *am, enum AVSampleFormat fmt, - enum AVMixCoeffType coeff_type, int in_channels, - int out_channels, int ptr_align, int samples_align, - const char *descr, void *mix_func); - -/** - * Allocate and initialize an AudioMix context. - * - * The parameters in the AVAudioResampleContext are used to initialize the - * AudioMix context. - * - * @param avr AVAudioResampleContext - * @return newly-allocated AudioMix context. - */ -AudioMix *ff_audio_mix_alloc(AVAudioResampleContext *avr); - -/** - * Free an AudioMix context. - */ -void ff_audio_mix_free(AudioMix **am); - -/** - * Apply channel mixing to audio data using the current mixing matrix. - */ -int ff_audio_mix(AudioMix *am, AudioData *src); - -/** - * Get the current mixing matrix. - */ -int ff_audio_mix_get_matrix(AudioMix *am, double *matrix, int stride); - -/** - * Set the current mixing matrix. - */ -int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride); - -/* arch-specific initialization functions */ - -void ff_audio_mix_init_x86(AudioMix *am); - -#endif /* AVRESAMPLE_AUDIO_MIX_H */ diff --git a/libavresample/audio_mix_matrix.c b/libavresample/audio_mix_matrix.c deleted file mode 100644 index 5d92351a0e..0000000000 --- a/libavresample/audio_mix_matrix.c +++ /dev/null @@ -1,294 +0,0 @@ -/* - * Copyright (C) 2011 Michael Niedermayer (michaelni@gmx.at) - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 <stdint.h> - -#include "libavutil/common.h" -#include "libavutil/libm.h" -#include "libavutil/samplefmt.h" -#include "avresample.h" -#include "internal.h" -#include "audio_data.h" -#include "audio_mix.h" - -/* channel positions */ -#define FRONT_LEFT 0 -#define FRONT_RIGHT 1 -#define FRONT_CENTER 2 -#define LOW_FREQUENCY 3 -#define BACK_LEFT 4 -#define BACK_RIGHT 5 -#define FRONT_LEFT_OF_CENTER 6 -#define FRONT_RIGHT_OF_CENTER 7 -#define BACK_CENTER 8 -#define SIDE_LEFT 9 -#define SIDE_RIGHT 10 -#define TOP_CENTER 11 -#define TOP_FRONT_LEFT 12 -#define TOP_FRONT_CENTER 13 -#define TOP_FRONT_RIGHT 14 -#define TOP_BACK_LEFT 15 -#define TOP_BACK_CENTER 16 -#define TOP_BACK_RIGHT 17 -#define STEREO_LEFT 29 -#define STEREO_RIGHT 30 -#define WIDE_LEFT 31 -#define WIDE_RIGHT 32 -#define SURROUND_DIRECT_LEFT 33 -#define SURROUND_DIRECT_RIGHT 34 -#define LOW_FREQUENCY_2 35 - -#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */ - -static av_always_inline int even(uint64_t layout) -{ - return (!layout || !!(layout & (layout - 1))); -} - -static int sane_layout(uint64_t layout) -{ - /* check that there is at least 1 front speaker */ - if (!(layout & AV_CH_LAYOUT_SURROUND)) - return 0; - - /* check for left/right symmetry */ - if (!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT)) || - !even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT)) || - !even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT)) || - !even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER)) || - !even(layout & (AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT)) || - !even(layout & (AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT)) || - !even(layout & (AV_CH_STEREO_LEFT | AV_CH_STEREO_RIGHT)) || - !even(layout & (AV_CH_WIDE_LEFT | AV_CH_WIDE_RIGHT)) || - !even(layout & (AV_CH_SURROUND_DIRECT_LEFT | AV_CH_SURROUND_DIRECT_RIGHT))) - return 0; - - return 1; -} - -int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout, - double center_mix_level, double surround_mix_level, - double lfe_mix_level, int normalize, - double *matrix_out, int stride, - enum AVMatrixEncoding matrix_encoding) -{ - int i, j, out_i, out_j; - double matrix[64][64] = {{0}}; - int64_t unaccounted; - double maxcoef = 0; - int in_channels, out_channels; - - if ((out_layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == AV_CH_LAYOUT_STEREO_DOWNMIX) { - out_layout = AV_CH_LAYOUT_STEREO; - } - - unaccounted = in_layout & ~out_layout; - - in_channels = av_get_channel_layout_nb_channels( in_layout); - out_channels = av_get_channel_layout_nb_channels(out_layout); - - memset(matrix_out, 0, out_channels * stride * sizeof(*matrix_out)); - - /* check if layouts are supported */ - if (!in_layout || in_channels > AVRESAMPLE_MAX_CHANNELS) - return AVERROR(EINVAL); - if (!out_layout || out_channels > AVRESAMPLE_MAX_CHANNELS) - return AVERROR(EINVAL); - - /* check if layouts are unbalanced or abnormal */ - if (!sane_layout(in_layout) || !sane_layout(out_layout)) - return AVERROR_PATCHWELCOME; - - /* route matching input/output channels */ - for (i = 0; i < 64; i++) { - if (in_layout & out_layout & (1ULL << i)) - matrix[i][i] = 1.0; - } - - /* mix front center to front left/right */ - if (unaccounted & AV_CH_FRONT_CENTER) { - if ((out_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO) { - if ((in_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO) { - matrix[FRONT_LEFT ][FRONT_CENTER] += center_mix_level; - matrix[FRONT_RIGHT][FRONT_CENTER] += center_mix_level; - } else { - matrix[FRONT_LEFT ][FRONT_CENTER] += M_SQRT1_2; - matrix[FRONT_RIGHT][FRONT_CENTER] += M_SQRT1_2; - } - } else - return AVERROR_PATCHWELCOME; - } - /* mix front left/right to center */ - if (unaccounted & AV_CH_LAYOUT_STEREO) { - if (out_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][FRONT_LEFT ] += M_SQRT1_2; - matrix[FRONT_CENTER][FRONT_RIGHT] += M_SQRT1_2; - /* mix left/right/center to center */ - if (in_layout & AV_CH_FRONT_CENTER) - matrix[FRONT_CENTER][FRONT_CENTER] = center_mix_level * M_SQRT2; - } else - return AVERROR_PATCHWELCOME; - } - /* mix back center to back, side, or front */ - if (unaccounted & AV_CH_BACK_CENTER) { - if (out_layout & AV_CH_BACK_LEFT) { - matrix[BACK_LEFT ][BACK_CENTER] += M_SQRT1_2; - matrix[BACK_RIGHT][BACK_CENTER] += M_SQRT1_2; - } else if (out_layout & AV_CH_SIDE_LEFT) { - matrix[SIDE_LEFT ][BACK_CENTER] += M_SQRT1_2; - matrix[SIDE_RIGHT][BACK_CENTER] += M_SQRT1_2; - } else if (out_layout & AV_CH_FRONT_LEFT) { - if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY || - matrix_encoding == AV_MATRIX_ENCODING_DPLII) { - if (unaccounted & (AV_CH_BACK_LEFT | AV_CH_SIDE_LEFT)) { - matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2; - } else { - matrix[FRONT_LEFT ][BACK_CENTER] -= surround_mix_level; - matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level; - } - } else { - matrix[FRONT_LEFT ][BACK_CENTER] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_CENTER] += surround_mix_level * M_SQRT1_2; - } - } else if (out_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][BACK_CENTER] += surround_mix_level * M_SQRT1_2; - } else - return AVERROR_PATCHWELCOME; - } - /* mix back left/right to back center, side, or front */ - if (unaccounted & AV_CH_BACK_LEFT) { - if (out_layout & AV_CH_BACK_CENTER) { - matrix[BACK_CENTER][BACK_LEFT ] += M_SQRT1_2; - matrix[BACK_CENTER][BACK_RIGHT] += M_SQRT1_2; - } else if (out_layout & AV_CH_SIDE_LEFT) { - /* if side channels do not exist in the input, just copy back - channels to side channels, otherwise mix back into side */ - if (in_layout & AV_CH_SIDE_LEFT) { - matrix[SIDE_LEFT ][BACK_LEFT ] += M_SQRT1_2; - matrix[SIDE_RIGHT][BACK_RIGHT] += M_SQRT1_2; - } else { - matrix[SIDE_LEFT ][BACK_LEFT ] += 1.0; - matrix[SIDE_RIGHT][BACK_RIGHT] += 1.0; - } - } else if (out_layout & AV_CH_FRONT_LEFT) { - if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) { - matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * M_SQRT1_2; - } else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) { - matrix[FRONT_LEFT ][BACK_LEFT ] -= surround_mix_level * SQRT3_2; - matrix[FRONT_LEFT ][BACK_RIGHT] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_LEFT ] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level * SQRT3_2; - } else { - matrix[FRONT_LEFT ][BACK_LEFT ] += surround_mix_level; - matrix[FRONT_RIGHT][BACK_RIGHT] += surround_mix_level; - } - } else if (out_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][BACK_LEFT ] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_CENTER][BACK_RIGHT] += surround_mix_level * M_SQRT1_2; - } else - return AVERROR_PATCHWELCOME; - } - /* mix side left/right into back or front */ - if (unaccounted & AV_CH_SIDE_LEFT) { - if (out_layout & AV_CH_BACK_LEFT) { - /* if back channels do not exist in the input, just copy side - channels to back channels, otherwise mix side into back */ - if (in_layout & AV_CH_BACK_LEFT) { - matrix[BACK_LEFT ][SIDE_LEFT ] += M_SQRT1_2; - matrix[BACK_RIGHT][SIDE_RIGHT] += M_SQRT1_2; - } else { - matrix[BACK_LEFT ][SIDE_LEFT ] += 1.0; - matrix[BACK_RIGHT][SIDE_RIGHT] += 1.0; - } - } else if (out_layout & AV_CH_BACK_CENTER) { - matrix[BACK_CENTER][SIDE_LEFT ] += M_SQRT1_2; - matrix[BACK_CENTER][SIDE_RIGHT] += M_SQRT1_2; - } else if (out_layout & AV_CH_FRONT_LEFT) { - if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) { - matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2; - } else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) { - matrix[FRONT_LEFT ][SIDE_LEFT ] -= surround_mix_level * SQRT3_2; - matrix[FRONT_LEFT ][SIDE_RIGHT] -= surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level * SQRT3_2; - } else { - matrix[FRONT_LEFT ][SIDE_LEFT ] += surround_mix_level; - matrix[FRONT_RIGHT][SIDE_RIGHT] += surround_mix_level; - } - } else if (out_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][SIDE_LEFT ] += surround_mix_level * M_SQRT1_2; - matrix[FRONT_CENTER][SIDE_RIGHT] += surround_mix_level * M_SQRT1_2; - } else - return AVERROR_PATCHWELCOME; - } - /* mix left-of-center/right-of-center into front left/right or center */ - if (unaccounted & AV_CH_FRONT_LEFT_OF_CENTER) { - if (out_layout & AV_CH_FRONT_LEFT) { - matrix[FRONT_LEFT ][FRONT_LEFT_OF_CENTER ] += 1.0; - matrix[FRONT_RIGHT][FRONT_RIGHT_OF_CENTER] += 1.0; - } else if (out_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][FRONT_LEFT_OF_CENTER ] += M_SQRT1_2; - matrix[FRONT_CENTER][FRONT_RIGHT_OF_CENTER] += M_SQRT1_2; - } else - return AVERROR_PATCHWELCOME; - } - /* mix LFE into front left/right or center */ - if (unaccounted & AV_CH_LOW_FREQUENCY) { - if (out_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][LOW_FREQUENCY] += lfe_mix_level; - } else if (out_layout & AV_CH_FRONT_LEFT) { - matrix[FRONT_LEFT ][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][LOW_FREQUENCY] += lfe_mix_level * M_SQRT1_2; - } else - return AVERROR_PATCHWELCOME; - } - - /* transfer internal matrix to output matrix and calculate maximum - per-channel coefficient sum */ - for (out_i = i = 0; out_i < out_channels && i < 64; i++) { - double sum = 0; - for (out_j = j = 0; out_j < in_channels && j < 64; j++) { - matrix_out[out_i * stride + out_j] = matrix[i][j]; - sum += fabs(matrix[i][j]); - if (in_layout & (1ULL << j)) - out_j++; - } - maxcoef = FFMAX(maxcoef, sum); - if (out_layout & (1ULL << i)) - out_i++; - } - - /* normalize */ - if (normalize && maxcoef > 1.0) { - for (i = 0; i < out_channels; i++) - for (j = 0; j < in_channels; j++) - matrix_out[i * stride + j] /= maxcoef; - } - - return 0; -} diff --git a/libavresample/avresample.h b/libavresample/avresample.h deleted file mode 100644 index 5ac9adb44b..0000000000 --- a/libavresample/avresample.h +++ /dev/null @@ -1,595 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#ifndef AVRESAMPLE_AVRESAMPLE_H -#define AVRESAMPLE_AVRESAMPLE_H - -/** - * @file - * @ingroup lavr - * external API header - */ - -/** - * @defgroup lavr libavresample - * @{ - * - * Libavresample (lavr) is a library that handles audio resampling, sample - * format conversion and mixing. - * - * Interaction with lavr is done through AVAudioResampleContext, which is - * allocated with avresample_alloc_context(). It is opaque, so all parameters - * must be set with the @ref avoptions API. - * - * For example the following code will setup conversion from planar float sample - * format to interleaved signed 16-bit integer, downsampling from 48kHz to - * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing - * matrix): - * @code - * AVAudioResampleContext *avr = avresample_alloc_context(); - * av_opt_set_int(avr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0); - * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0); - * av_opt_set_int(avr, "in_sample_rate", 48000, 0); - * av_opt_set_int(avr, "out_sample_rate", 44100, 0); - * av_opt_set_int(avr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); - * av_opt_set_int(avr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); - * @endcode - * - * Once the context is initialized, it must be opened with avresample_open(). If - * you need to change the conversion parameters, you must close the context with - * avresample_close(), change the parameters as described above, then reopen it - * again. - * - * The conversion itself is done by repeatedly calling avresample_convert(). - * Note that the samples may get buffered in two places in lavr. The first one - * is the output FIFO, where the samples end up if the output buffer is not - * large enough. The data stored in there may be retrieved at any time with - * avresample_read(). The second place is the resampling delay buffer, - * applicable only when resampling is done. The samples in it require more input - * before they can be processed. Their current amount is returned by - * avresample_get_delay(). At the end of conversion the resampling buffer can be - * flushed by calling avresample_convert() with NULL input. - * - * The following code demonstrates the conversion loop assuming the parameters - * from above and caller-defined functions get_input() and handle_output(): - * @code - * uint8_t **input; - * int in_linesize, in_samples; - * - * while (get_input(&input, &in_linesize, &in_samples)) { - * uint8_t *output - * int out_linesize; - * int out_samples = avresample_get_out_samples(avr, in_samples); - * - * av_samples_alloc(&output, &out_linesize, 2, out_samples, - * AV_SAMPLE_FMT_S16, 0); - * out_samples = avresample_convert(avr, &output, out_linesize, out_samples, - * input, in_linesize, in_samples); - * handle_output(output, out_linesize, out_samples); - * av_freep(&output); - * } - * @endcode - * - * When the conversion is finished and the FIFOs are flushed if required, the - * conversion context and everything associated with it must be freed with - * avresample_free(). - */ - -#include "libavutil/avutil.h" -#include "libavutil/channel_layout.h" -#include "libavutil/dict.h" -#include "libavutil/frame.h" -#include "libavutil/log.h" -#include "libavutil/mathematics.h" - -#include "libavresample/version.h" - -#define AVRESAMPLE_MAX_CHANNELS 32 - -typedef struct AVAudioResampleContext AVAudioResampleContext; - -/** - * @deprecated use libswresample - * - * Mixing Coefficient Types */ -enum attribute_deprecated AVMixCoeffType { - AV_MIX_COEFF_TYPE_Q8, /** 16-bit 8.8 fixed-point */ - AV_MIX_COEFF_TYPE_Q15, /** 32-bit 17.15 fixed-point */ - AV_MIX_COEFF_TYPE_FLT, /** floating-point */ - AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */ -}; - -/** - * @deprecated use libswresample - * - * Resampling Filter Types */ -enum attribute_deprecated AVResampleFilterType { - AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */ - AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */ - AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */ -}; - -/** - * @deprecated use libswresample - */ -enum attribute_deprecated AVResampleDitherMethod { - AV_RESAMPLE_DITHER_NONE, /**< Do not use dithering */ - AV_RESAMPLE_DITHER_RECTANGULAR, /**< Rectangular Dither */ - AV_RESAMPLE_DITHER_TRIANGULAR, /**< Triangular Dither*/ - AV_RESAMPLE_DITHER_TRIANGULAR_HP, /**< Triangular Dither with High Pass */ - AV_RESAMPLE_DITHER_TRIANGULAR_NS, /**< Triangular Dither with Noise Shaping */ - AV_RESAMPLE_DITHER_NB, /**< Number of dither types. Not part of ABI. */ -}; - -/** - * - * @deprecated use libswresample - * - * Return the LIBAVRESAMPLE_VERSION_INT constant. - */ -attribute_deprecated -unsigned avresample_version(void); - -/** - * - * @deprecated use libswresample - * - * Return the libavresample build-time configuration. - * @return configure string - */ -attribute_deprecated -const char *avresample_configuration(void); - -/** - * - * @deprecated use libswresample - * - * Return the libavresample license. - */ -attribute_deprecated -const char *avresample_license(void); - -/** - * - * @deprecated use libswresample - * - * Get the AVClass for AVAudioResampleContext. - * - * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options - * without allocating a context. - * - * @see av_opt_find(). - * - * @return AVClass for AVAudioResampleContext - */ -attribute_deprecated -const AVClass *avresample_get_class(void); - -/** - * - * @deprecated use libswresample - * - * Allocate AVAudioResampleContext and set options. - * - * @return allocated audio resample context, or NULL on failure - */ -attribute_deprecated -AVAudioResampleContext *avresample_alloc_context(void); - -/** - * - * @deprecated use libswresample - * - * Initialize AVAudioResampleContext. - * @note The context must be configured using the AVOption API. - * @note The fields "in_channel_layout", "out_channel_layout", - * "in_sample_rate", "out_sample_rate", "in_sample_fmt", - * "out_sample_fmt" must be set. - * - * @see av_opt_set_int() - * @see av_opt_set_dict() - * @see av_get_default_channel_layout() - * - * @param avr audio resample context - * @return 0 on success, negative AVERROR code on failure - */ -attribute_deprecated -int avresample_open(AVAudioResampleContext *avr); - -/** - * - * @deprecated use libswresample - * - * Check whether an AVAudioResampleContext is open or closed. - * - * @param avr AVAudioResampleContext to check - * @return 1 if avr is open, 0 if avr is closed. - */ -attribute_deprecated -int avresample_is_open(AVAudioResampleContext *avr); - -/** - * - * @deprecated use libswresample - * - * Close AVAudioResampleContext. - * - * This closes the context, but it does not change the parameters. The context - * can be reopened with avresample_open(). It does, however, clear the output - * FIFO and any remaining leftover samples in the resampling delay buffer. If - * there was a custom matrix being used, that is also cleared. - * - * @see avresample_convert() - * @see avresample_set_matrix() - * - * @param avr audio resample context - */ -attribute_deprecated -void avresample_close(AVAudioResampleContext *avr); - -/** - * - * @deprecated use libswresample - * - * Free AVAudioResampleContext and associated AVOption values. - * - * This also calls avresample_close() before freeing. - * - * @param avr audio resample context - */ -attribute_deprecated -void avresample_free(AVAudioResampleContext **avr); - -/** - * - * @deprecated use libswresample - * - * Generate a channel mixing matrix. - * - * This function is the one used internally by libavresample for building the - * default mixing matrix. It is made public just as a utility function for - * building custom matrices. - * - * @param in_layout input channel layout - * @param out_layout output channel layout - * @param center_mix_level mix level for the center channel - * @param surround_mix_level mix level for the surround channel(s) - * @param lfe_mix_level mix level for the low-frequency effects channel - * @param normalize if 1, coefficients will be normalized to prevent - * overflow. if 0, coefficients will not be - * normalized. - * @param[out] matrix mixing coefficients; matrix[i + stride * o] is - * the weight of input channel i in output channel o. - * @param stride distance between adjacent input channels in the - * matrix array - * @param matrix_encoding matrixed stereo downmix mode (e.g. dplii) - * @return 0 on success, negative AVERROR code on failure - */ -attribute_deprecated -int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout, - double center_mix_level, double surround_mix_level, - double lfe_mix_level, int normalize, double *matrix, - int stride, enum AVMatrixEncoding matrix_encoding); - -/** - * - * @deprecated use libswresample - * - * Get the current channel mixing matrix. - * - * If no custom matrix has been previously set or the AVAudioResampleContext is - * not open, an error is returned. - * - * @param avr audio resample context - * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of - * input channel i in output channel o. - * @param stride distance between adjacent input channels in the matrix array - * @return 0 on success, negative AVERROR code on failure - */ -attribute_deprecated -int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, - int stride); - -/** - * - * @deprecated use libswresample - * - * Set channel mixing matrix. - * - * Allows for setting a custom mixing matrix, overriding the default matrix - * generated internally during avresample_open(). This function can be called - * anytime on an allocated context, either before or after calling - * avresample_open(), as long as the channel layouts have been set. - * avresample_convert() always uses the current matrix. - * Calling avresample_close() on the context will clear the current matrix. - * - * @see avresample_close() - * - * @param avr audio resample context - * @param matrix mixing coefficients; matrix[i + stride * o] is the weight of - * input channel i in output channel o. - * @param stride distance between adjacent input channels in the matrix array - * @return 0 on success, negative AVERROR code on failure - */ -attribute_deprecated -int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, - int stride); - -/** - * - * @deprecated use libswresample - * - * Set a customized input channel mapping. - * - * This function can only be called when the allocated context is not open. - * Also, the input channel layout must have already been set. - * - * Calling avresample_close() on the context will clear the channel mapping. - * - * The map for each input channel specifies the channel index in the source to - * use for that particular channel, or -1 to mute the channel. Source channels - * can be duplicated by using the same index for multiple input channels. - * - * Examples: - * - * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs): - * { 1, 2, 0, 5, 3, 4 } - * - * Muting the 3rd channel in 4-channel input: - * { 0, 1, -1, 3 } - * - * Duplicating the left channel of stereo input: - * { 0, 0 } - * - * @param avr audio resample context - * @param channel_map customized input channel mapping - * @return 0 on success, negative AVERROR code on failure - */ -attribute_deprecated -int avresample_set_channel_mapping(AVAudioResampleContext *avr, - const int *channel_map); - -/** - * - * @deprecated use libswresample - * - * Set compensation for resampling. - * - * This can be called anytime after avresample_open(). If resampling is not - * automatically enabled because of a sample rate conversion, the - * "force_resampling" option must have been set to 1 when opening the context - * in order to use resampling compensation. - * - * @param avr audio resample context - * @param sample_delta compensation delta, in samples - * @param compensation_distance compensation distance, in samples - * @return 0 on success, negative AVERROR code on failure - */ -attribute_deprecated -int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta, - int compensation_distance); - -/** - * - * @deprecated use libswresample - * - * Provide the upper bound on the number of samples the configured - * conversion would output. - * - * @param avr audio resample context - * @param in_nb_samples number of input samples - * - * @return number of samples or AVERROR(EINVAL) if the value - * would exceed INT_MAX - */ -attribute_deprecated -int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples); - -/** - * - * @deprecated use libswresample - * - * Convert input samples and write them to the output FIFO. - * - * The upper bound on the number of output samples can be obtained through - * avresample_get_out_samples(). - * - * The output data can be NULL or have fewer allocated samples than required. - * In this case, any remaining samples not written to the output will be added - * to an internal FIFO buffer, to be returned at the next call to this function - * or to avresample_read(). - * - * If converting sample rate, there may be data remaining in the internal - * resampling delay buffer. avresample_get_delay() tells the number of remaining - * samples. To get this data as output, call avresample_convert() with NULL - * input. - * - * At the end of the conversion process, there may be data remaining in the - * internal FIFO buffer. avresample_available() tells the number of remaining - * samples. To get this data as output, either call avresample_convert() with - * NULL input or call avresample_read(). - * - * @see avresample_get_out_samples() - * @see avresample_read() - * @see avresample_get_delay() - * - * @param avr audio resample context - * @param output output data pointers - * @param out_plane_size output plane size, in bytes. - * This can be 0 if unknown, but that will lead to - * optimized functions not being used directly on the - * output, which could slow down some conversions. - * @param out_samples maximum number of samples that the output buffer can hold - * @param input input data pointers - * @param in_plane_size input plane size, in bytes - * This can be 0 if unknown, but that will lead to - * optimized functions not being used directly on the - * input, which could slow down some conversions. - * @param in_samples number of input samples to convert - * @return number of samples written to the output buffer, - * not including converted samples added to the internal - * output FIFO - */ -attribute_deprecated -int avresample_convert(AVAudioResampleContext *avr, uint8_t **output, - int out_plane_size, int out_samples, - uint8_t * const *input, int in_plane_size, - int in_samples); - -/** - * - * @deprecated use libswresample - * - * Return the number of samples currently in the resampling delay buffer. - * - * When resampling, there may be a delay between the input and output. Any - * unconverted samples in each call are stored internally in a delay buffer. - * This function allows the user to determine the current number of samples in - * the delay buffer, which can be useful for synchronization. - * - * @see avresample_convert() - * - * @param avr audio resample context - * @return number of samples currently in the resampling delay buffer - */ -attribute_deprecated -int avresample_get_delay(AVAudioResampleContext *avr); - -/** - * - * @deprecated use libswresample - * - * Return the number of available samples in the output FIFO. - * - * During conversion, if the user does not specify an output buffer or - * specifies an output buffer that is smaller than what is needed, remaining - * samples that are not written to the output are stored to an internal FIFO - * buffer. The samples in the FIFO can be read with avresample_read() or - * avresample_convert(). - * - * @see avresample_read() - * @see avresample_convert() - * - * @param avr audio resample context - * @return number of samples available for reading - */ -attribute_deprecated -int avresample_available(AVAudioResampleContext *avr); - -/** - * - * @deprecated use libswresample - * - * Read samples from the output FIFO. - * - * During conversion, if the user does not specify an output buffer or - * specifies an output buffer that is smaller than what is needed, remaining - * samples that are not written to the output are stored to an internal FIFO - * buffer. This function can be used to read samples from that internal FIFO. - * - * @see avresample_available() - * @see avresample_convert() - * - * @param avr audio resample context - * @param output output data pointers. May be NULL, in which case - * nb_samples of data is discarded from output FIFO. - * @param nb_samples number of samples to read from the FIFO - * @return the number of samples written to output - */ -attribute_deprecated -int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples); - -/** - * - * @deprecated use libswresample - * - * Convert the samples in the input AVFrame and write them to the output AVFrame. - * - * Input and output AVFrames must have channel_layout, sample_rate and format set. - * - * The upper bound on the number of output samples is obtained through - * avresample_get_out_samples(). - * - * If the output AVFrame does not have the data pointers allocated the nb_samples - * field will be set using avresample_get_out_samples() and av_frame_get_buffer() - * is called to allocate the frame. - * - * The output AVFrame can be NULL or have fewer allocated samples than required. - * In this case, any remaining samples not written to the output will be added - * to an internal FIFO buffer, to be returned at the next call to this function - * or to avresample_convert() or to avresample_read(). - * - * If converting sample rate, there may be data remaining in the internal - * resampling delay buffer. avresample_get_delay() tells the number of - * remaining samples. To get this data as output, call this function or - * avresample_convert() with NULL input. - * - * At the end of the conversion process, there may be data remaining in the - * internal FIFO buffer. avresample_available() tells the number of remaining - * samples. To get this data as output, either call this function or - * avresample_convert() with NULL input or call avresample_read(). - * - * If the AVAudioResampleContext configuration does not match the output and - * input AVFrame settings the conversion does not take place and depending on - * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED - * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned. - * - * @see avresample_get_out_samples() - * @see avresample_available() - * @see avresample_convert() - * @see avresample_read() - * @see avresample_get_delay() - * - * @param avr audio resample context - * @param output output AVFrame - * @param input input AVFrame - * @return 0 on success, AVERROR on failure or nonmatching - * configuration. - */ -attribute_deprecated -int avresample_convert_frame(AVAudioResampleContext *avr, - AVFrame *output, AVFrame *input); - -/** - * - * @deprecated use libswresample - * - * Configure or reconfigure the AVAudioResampleContext using the information - * provided by the AVFrames. - * - * The original resampling context is reset even on failure. - * The function calls avresample_close() internally if the context is open. - * - * @see avresample_open(); - * @see avresample_close(); - * - * @param avr audio resample context - * @param out output AVFrame - * @param in input AVFrame - * @return 0 on success, AVERROR on failure. - */ -attribute_deprecated -int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in); - -/** - * @} - */ - -#endif /* AVRESAMPLE_AVRESAMPLE_H */ diff --git a/libavresample/avresampleres.rc b/libavresample/avresampleres.rc deleted file mode 100644 index e6d0d151e1..0000000000 --- a/libavresample/avresampleres.rc +++ /dev/null @@ -1,55 +0,0 @@ -/* - * Windows resource file for libavresample - * - * Copyright (C) 2012 James Almer - * Copyright (C) 2013 Tiancheng "Timothy" Gu - * - * 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 <windows.h> -#include "libavresample/version.h" -#include "libavutil/ffversion.h" -#include "config.h" - -1 VERSIONINFO -FILEVERSION LIBAVRESAMPLE_VERSION_MAJOR, LIBAVRESAMPLE_VERSION_MINOR, LIBAVRESAMPLE_VERSION_MICRO, 0 -PRODUCTVERSION LIBAVRESAMPLE_VERSION_MAJOR, LIBAVRESAMPLE_VERSION_MINOR, LIBAVRESAMPLE_VERSION_MICRO, 0 -FILEFLAGSMASK VS_FFI_FILEFLAGSMASK -FILEOS VOS_NT_WINDOWS32 -FILETYPE VFT_DLL -{ - BLOCK "StringFileInfo" - { - BLOCK "040904B0" - { - VALUE "CompanyName", "FFmpeg Project" - VALUE "FileDescription", "Libav audio resampling library" - VALUE "FileVersion", AV_STRINGIFY(LIBAVRESAMPLE_VERSION) - VALUE "InternalName", "libavresample" - VALUE "LegalCopyright", "Copyright (C) 2000-" AV_STRINGIFY(CONFIG_THIS_YEAR) " FFmpeg Project" - VALUE "OriginalFilename", "avresample" BUILDSUF "-" AV_STRINGIFY(LIBAVRESAMPLE_VERSION_MAJOR) SLIBSUF - VALUE "ProductName", "FFmpeg" - VALUE "ProductVersion", FFMPEG_VERSION - } - } - - BLOCK "VarFileInfo" - { - VALUE "Translation", 0x0409, 0x04B0 - } -} diff --git a/libavresample/dither.c b/libavresample/dither.c deleted file mode 100644 index 2ae8d338be..0000000000 --- a/libavresample/dither.c +++ /dev/null @@ -1,440 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * Triangular with Noise Shaping is based on opusfile. - * Copyright (c) 1994-2012 by the Xiph.Org Foundation and contributors - * - * 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 - */ - -/** - * @file - * Dithered Audio Sample Quantization - * - * Converts from dbl, flt, or s32 to s16 using dithering. - */ - -#include <math.h> -#include <stdint.h> - -#include "libavutil/attributes.h" -#include "libavutil/common.h" -#include "libavutil/lfg.h" -#include "libavutil/mem.h" -#include "libavutil/samplefmt.h" -#include "audio_convert.h" -#include "dither.h" -#include "internal.h" - -typedef struct DitherState { - int mute; - unsigned int seed; - AVLFG lfg; - float *noise_buf; - int noise_buf_size; - int noise_buf_ptr; - float dither_a[4]; - float dither_b[4]; -} DitherState; - -struct DitherContext { - DitherDSPContext ddsp; - enum AVResampleDitherMethod method; - int apply_map; - ChannelMapInfo *ch_map_info; - - int mute_dither_threshold; // threshold for disabling dither - int mute_reset_threshold; // threshold for resetting noise shaping - const float *ns_coef_b; // noise shaping coeffs - const float *ns_coef_a; // noise shaping coeffs - - int channels; - DitherState *state; // dither states for each channel - - AudioData *flt_data; // input data in fltp - AudioData *s16_data; // dithered output in s16p - AudioConvert *ac_in; // converter for input to fltp - AudioConvert *ac_out; // converter for s16p to s16 (if needed) - - void (*quantize)(int16_t *dst, const float *src, float *dither, int len); - int samples_align; -}; - -/* mute threshold, in seconds */ -#define MUTE_THRESHOLD_SEC 0.000333 - -/* scale factor for 16-bit output. - The signal is attenuated slightly to avoid clipping */ -#define S16_SCALE 32753.0f - -/* scale to convert lfg from INT_MIN/INT_MAX to -0.5/0.5 */ -#define LFG_SCALE (1.0f / (2.0f * INT32_MAX)) - -/* noise shaping coefficients */ - -static const float ns_48_coef_b[4] = { - 2.2374f, -0.7339f, -0.1251f, -0.6033f -}; - -static const float ns_48_coef_a[4] = { - 0.9030f, 0.0116f, -0.5853f, -0.2571f -}; - -static const float ns_44_coef_b[4] = { - 2.2061f, -0.4707f, -0.2534f, -0.6213f -}; - -static const float ns_44_coef_a[4] = { - 1.0587f, 0.0676f, -0.6054f, -0.2738f -}; - -static void dither_int_to_float_rectangular_c(float *dst, int *src, int len) -{ - int i; - for (i = 0; i < len; i++) - dst[i] = src[i] * LFG_SCALE; -} - -static void dither_int_to_float_triangular_c(float *dst, int *src0, int len) -{ - int i; - int *src1 = src0 + len; - - for (i = 0; i < len; i++) { - float r = src0[i] * LFG_SCALE; - r += src1[i] * LFG_SCALE; - dst[i] = r; - } -} - -static void quantize_c(int16_t *dst, const float *src, float *dither, int len) -{ - int i; - for (i = 0; i < len; i++) - dst[i] = av_clip_int16(lrintf(src[i] * S16_SCALE + dither[i])); -} - -#define SQRT_1_6 0.40824829046386301723f - -static void dither_highpass_filter(float *src, int len) -{ - int i; - - /* filter is from libswresample in FFmpeg */ - for (i = 0; i < len - 2; i++) - src[i] = (-src[i] + 2 * src[i + 1] - src[i + 2]) * SQRT_1_6; -} - -static int generate_dither_noise(DitherContext *c, DitherState *state, - int min_samples) -{ - int i; - int nb_samples = FFALIGN(min_samples, 16) + 16; - int buf_samples = nb_samples * - (c->method == AV_RESAMPLE_DITHER_RECTANGULAR ? 1 : 2); - unsigned int *noise_buf_ui; - - av_freep(&state->noise_buf); - state->noise_buf_size = state->noise_buf_ptr = 0; - - state->noise_buf = av_malloc(buf_samples * sizeof(*state->noise_buf)); - if (!state->noise_buf) - return AVERROR(ENOMEM); - state->noise_buf_size = FFALIGN(min_samples, 16); - noise_buf_ui = (unsigned int *)state->noise_buf; - - av_lfg_init(&state->lfg, state->seed); - for (i = 0; i < buf_samples; i++) - noise_buf_ui[i] = av_lfg_get(&state->lfg); - - c->ddsp.dither_int_to_float(state->noise_buf, noise_buf_ui, nb_samples); - - if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_HP) - dither_highpass_filter(state->noise_buf, nb_samples); - - return 0; -} - -static void quantize_triangular_ns(DitherContext *c, DitherState *state, - int16_t *dst, const float *src, - int nb_samples) -{ - int i, j; - float *dither = &state->noise_buf[state->noise_buf_ptr]; - - if (state->mute > c->mute_reset_threshold) - memset(state->dither_a, 0, sizeof(state->dither_a)); - - for (i = 0; i < nb_samples; i++) { - float err = 0; - float sample = src[i] * S16_SCALE; - - for (j = 0; j < 4; j++) { - err += c->ns_coef_b[j] * state->dither_b[j] - - c->ns_coef_a[j] * state->dither_a[j]; - } - for (j = 3; j > 0; j--) { - state->dither_a[j] = state->dither_a[j - 1]; - state->dither_b[j] = state->dither_b[j - 1]; - } - state->dither_a[0] = err; - sample -= err; - - if (state->mute > c->mute_dither_threshold) { - dst[i] = av_clip_int16(lrintf(sample)); - state->dither_b[0] = 0; - } else { - dst[i] = av_clip_int16(lrintf(sample + dither[i])); - state->dither_b[0] = av_clipf(dst[i] - sample, -1.5f, 1.5f); - } - - state->mute++; - if (src[i]) - state->mute = 0; - } -} - -static int convert_samples(DitherContext *c, int16_t **dst, float * const *src, - int channels, int nb_samples) -{ - int ch, ret; - int aligned_samples = FFALIGN(nb_samples, 16); - - for (ch = 0; ch < channels; ch++) { - DitherState *state = &c->state[ch]; - - if (state->noise_buf_size < aligned_samples) { - ret = generate_dither_noise(c, state, nb_samples); - if (ret < 0) - return ret; - } else if (state->noise_buf_size - state->noise_buf_ptr < aligned_samples) { - state->noise_buf_ptr = 0; - } - - if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) { - quantize_triangular_ns(c, state, dst[ch], src[ch], nb_samples); - } else { - c->quantize(dst[ch], src[ch], - &state->noise_buf[state->noise_buf_ptr], - FFALIGN(nb_samples, c->samples_align)); - } - - state->noise_buf_ptr += aligned_samples; - } - - return 0; -} - -int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src) -{ - int ret; - AudioData *flt_data; - - /* output directly to dst if it is planar */ - if (dst->sample_fmt == AV_SAMPLE_FMT_S16P) - c->s16_data = dst; - else { - /* make sure s16_data is large enough for the output */ - ret = ff_audio_data_realloc(c->s16_data, src->nb_samples); - if (ret < 0) - return ret; - } - - if (src->sample_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) { - /* make sure flt_data is large enough for the input */ - ret = ff_audio_data_realloc(c->flt_data, src->nb_samples); - if (ret < 0) - return ret; - flt_data = c->flt_data; - } - - if (src->sample_fmt != AV_SAMPLE_FMT_FLTP) { - /* convert input samples to fltp and scale to s16 range */ - ret = ff_audio_convert(c->ac_in, flt_data, src); - if (ret < 0) - return ret; - } else if (c->apply_map) { - ret = ff_audio_data_copy(flt_data, src, c->ch_map_info); - if (ret < 0) - return ret; - } else { - flt_data = src; - } - - /* check alignment and padding constraints */ - if (c->method != AV_RESAMPLE_DITHER_TRIANGULAR_NS) { - int ptr_align = FFMIN(flt_data->ptr_align, c->s16_data->ptr_align); - int samples_align = FFMIN(flt_data->samples_align, c->s16_data->samples_align); - int aligned_len = FFALIGN(src->nb_samples, c->ddsp.samples_align); - - if (!(ptr_align % c->ddsp.ptr_align) && samples_align >= aligned_len) { - c->quantize = c->ddsp.quantize; - c->samples_align = c->ddsp.samples_align; - } else { - c->quantize = quantize_c; - c->samples_align = 1; - } - } - - ret = convert_samples(c, (int16_t **)c->s16_data->data, - (float * const *)flt_data->data, src->channels, - src->nb_samples); - if (ret < 0) - return ret; - - c->s16_data->nb_samples = src->nb_samples; - - /* interleave output to dst if needed */ - if (dst->sample_fmt == AV_SAMPLE_FMT_S16) { - ret = ff_audio_convert(c->ac_out, dst, c->s16_data); - if (ret < 0) - return ret; - } else - c->s16_data = NULL; - - return 0; -} - -void ff_dither_free(DitherContext **cp) -{ - DitherContext *c = *cp; - int ch; - - if (!c) - return; - ff_audio_data_free(&c->flt_data); - ff_audio_data_free(&c->s16_data); - ff_audio_convert_free(&c->ac_in); - ff_audio_convert_free(&c->ac_out); - for (ch = 0; ch < c->channels; ch++) - av_free(c->state[ch].noise_buf); - av_free(c->state); - av_freep(cp); -} - -static av_cold void dither_init(DitherDSPContext *ddsp, - enum AVResampleDitherMethod method) -{ - ddsp->quantize = quantize_c; - ddsp->ptr_align = 1; - ddsp->samples_align = 1; - - if (method == AV_RESAMPLE_DITHER_RECTANGULAR) - ddsp->dither_int_to_float = dither_int_to_float_rectangular_c; - else - ddsp->dither_int_to_float = dither_int_to_float_triangular_c; - - if (ARCH_X86) - ff_dither_init_x86(ddsp, method); -} - -DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels, int sample_rate, int apply_map) -{ - AVLFG seed_gen; - DitherContext *c; - int ch; - - if (av_get_packed_sample_fmt(out_fmt) != AV_SAMPLE_FMT_S16 || - av_get_bytes_per_sample(in_fmt) <= 2) { - av_log(avr, AV_LOG_ERROR, "dithering %s to %s is not supported\n", - av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt)); - return NULL; - } - - c = av_mallocz(sizeof(*c)); - if (!c) - return NULL; - - c->apply_map = apply_map; - if (apply_map) - c->ch_map_info = &avr->ch_map_info; - - if (avr->dither_method == AV_RESAMPLE_DITHER_TRIANGULAR_NS && - sample_rate != 48000 && sample_rate != 44100) { - av_log(avr, AV_LOG_WARNING, "sample rate must be 48000 or 44100 Hz " - "for triangular_ns dither. using triangular_hp instead.\n"); - avr->dither_method = AV_RESAMPLE_DITHER_TRIANGULAR_HP; - } - c->method = avr->dither_method; - dither_init(&c->ddsp, c->method); - - if (c->method == AV_RESAMPLE_DITHER_TRIANGULAR_NS) { - if (sample_rate == 48000) { - c->ns_coef_b = ns_48_coef_b; - c->ns_coef_a = ns_48_coef_a; - } else { - c->ns_coef_b = ns_44_coef_b; - c->ns_coef_a = ns_44_coef_a; - } - } - - /* Either s16 or s16p output format is allowed, but s16p is used - internally, so we need to use a temp buffer and interleave if the output - format is s16 */ - if (out_fmt != AV_SAMPLE_FMT_S16P) { - c->s16_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_S16P, - "dither s16 buffer"); - if (!c->s16_data) - goto fail; - - c->ac_out = ff_audio_convert_alloc(avr, out_fmt, AV_SAMPLE_FMT_S16P, - channels, sample_rate, 0); - if (!c->ac_out) - goto fail; - } - - if (in_fmt != AV_SAMPLE_FMT_FLTP || c->apply_map) { - c->flt_data = ff_audio_data_alloc(channels, 1024, AV_SAMPLE_FMT_FLTP, - "dither flt buffer"); - if (!c->flt_data) - goto fail; - } - if (in_fmt != AV_SAMPLE_FMT_FLTP) { - c->ac_in = ff_audio_convert_alloc(avr, AV_SAMPLE_FMT_FLTP, in_fmt, - channels, sample_rate, c->apply_map); - if (!c->ac_in) - goto fail; - } - - c->state = av_mallocz(channels * sizeof(*c->state)); - if (!c->state) - goto fail; - c->channels = channels; - - /* calculate thresholds for turning off dithering during periods of - silence to avoid replacing digital silence with quiet dither noise */ - c->mute_dither_threshold = lrintf(sample_rate * MUTE_THRESHOLD_SEC); - c->mute_reset_threshold = c->mute_dither_threshold * 4; - - /* initialize dither states */ - av_lfg_init(&seed_gen, 0xC0FFEE); - for (ch = 0; ch < channels; ch++) { - DitherState *state = &c->state[ch]; - state->mute = c->mute_reset_threshold + 1; - state->seed = av_lfg_get(&seed_gen); - generate_dither_noise(c, state, FFMAX(32768, sample_rate / 2)); - } - - return c; - -fail: - ff_dither_free(&c); - return NULL; -} diff --git a/libavresample/dither.h b/libavresample/dither.h deleted file mode 100644 index 72f09cbdde..0000000000 --- a/libavresample/dither.h +++ /dev/null @@ -1,93 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#ifndef AVRESAMPLE_DITHER_H -#define AVRESAMPLE_DITHER_H - -#include "avresample.h" -#include "audio_data.h" - -typedef struct DitherContext DitherContext; - -typedef struct DitherDSPContext { - /** - * Convert samples from flt to s16 with added dither noise. - * - * @param dst destination float array, range -0.5 to 0.5 - * @param src source int array, range INT_MIN to INT_MAX. - * @param dither float dither noise array - * @param len number of samples - */ - void (*quantize)(int16_t *dst, const float *src, float *dither, int len); - - int ptr_align; ///< src and dst constraints for quantize() - int samples_align; ///< len constraints for quantize() - - /** - * Convert dither noise from int to float with triangular distribution. - * - * @param dst destination float array, range -0.5 to 0.5 - * constraints: 32-byte aligned - * @param src0 source int array, range INT_MIN to INT_MAX. - * the array size is len * 2 - * constraints: 32-byte aligned - * @param len number of output noise samples - * constraints: multiple of 16 - */ - void (*dither_int_to_float)(float *dst, int *src0, int len); -} DitherDSPContext; - -/** - * Allocate and initialize a DitherContext. - * - * The parameters in the AVAudioResampleContext are used to initialize the - * DitherContext. - * - * @param avr AVAudioResampleContext - * @return newly-allocated DitherContext - */ -DitherContext *ff_dither_alloc(AVAudioResampleContext *avr, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels, int sample_rate, int apply_map); - -/** - * Free a DitherContext. - * - * @param c DitherContext - */ -void ff_dither_free(DitherContext **c); - -/** - * Convert audio sample format with dithering. - * - * @param c DitherContext - * @param dst destination audio data - * @param src source audio data - * @return 0 if ok, negative AVERROR code on failure - */ -int ff_convert_dither(DitherContext *c, AudioData *dst, AudioData *src); - -/* arch-specific initialization functions */ - -void ff_dither_init_x86(DitherDSPContext *ddsp, - enum AVResampleDitherMethod method); - -#endif /* AVRESAMPLE_DITHER_H */ diff --git a/libavresample/internal.h b/libavresample/internal.h deleted file mode 100644 index 2fc3f6da67..0000000000 --- a/libavresample/internal.h +++ /dev/null @@ -1,116 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 - */ - -#ifndef AVRESAMPLE_INTERNAL_H -#define AVRESAMPLE_INTERNAL_H - -#include "libavutil/audio_fifo.h" -#include "libavutil/log.h" -#include "libavutil/opt.h" -#include "libavutil/samplefmt.h" -#include "avresample.h" - -typedef struct AudioData AudioData; -typedef struct AudioConvert AudioConvert; -typedef struct AudioMix AudioMix; -typedef struct ResampleContext ResampleContext; - -enum RemapPoint { - REMAP_NONE, - REMAP_IN_COPY, - REMAP_IN_CONVERT, - REMAP_OUT_COPY, - REMAP_OUT_CONVERT, -}; - -typedef struct ChannelMapInfo { - int channel_map[AVRESAMPLE_MAX_CHANNELS]; /**< source index of each output channel, -1 if not remapped */ - int do_remap; /**< remap needed */ - int channel_copy[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to copy from */ - int do_copy; /**< copy needed */ - int channel_zero[AVRESAMPLE_MAX_CHANNELS]; /**< dest index to zero */ - int do_zero; /**< zeroing needed */ - int input_map[AVRESAMPLE_MAX_CHANNELS]; /**< dest index of each input channel */ -} ChannelMapInfo; - -struct AVAudioResampleContext { - const AVClass *av_class; /**< AVClass for logging and AVOptions */ - - uint64_t in_channel_layout; /**< input channel layout */ - enum AVSampleFormat in_sample_fmt; /**< input sample format */ - int in_sample_rate; /**< input sample rate */ - uint64_t out_channel_layout; /**< output channel layout */ - enum AVSampleFormat out_sample_fmt; /**< output sample format */ - int out_sample_rate; /**< output sample rate */ - enum AVSampleFormat internal_sample_fmt; /**< internal sample format */ - enum AVMixCoeffType mix_coeff_type; /**< mixing coefficient type */ - double center_mix_level; /**< center mix level */ - double surround_mix_level; /**< surround mix level */ - double lfe_mix_level; /**< lfe mix level */ - int normalize_mix_level; /**< enable mix level normalization */ - int force_resampling; /**< force resampling */ - int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */ - int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */ - int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */ - double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */ - enum AVResampleFilterType filter_type; /**< resampling filter type */ - int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ - enum AVResampleDitherMethod dither_method; /**< dither method */ - - int in_channels; /**< number of input channels */ - int out_channels; /**< number of output channels */ - int resample_channels; /**< number of channels used for resampling */ - int downmix_needed; /**< downmixing is needed */ - int upmix_needed; /**< upmixing is needed */ - int mixing_needed; /**< either upmixing or downmixing is needed */ - int resample_needed; /**< resampling is needed */ - int in_convert_needed; /**< input sample format conversion is needed */ - int out_convert_needed; /**< output sample format conversion is needed */ - int in_copy_needed; /**< input data copy is needed */ - - AudioData *in_buffer; /**< buffer for converted input */ - AudioData *resample_out_buffer; /**< buffer for output from resampler */ - AudioData *out_buffer; /**< buffer for converted output */ - AVAudioFifo *out_fifo; /**< FIFO for output samples */ - - AudioConvert *ac_in; /**< input sample format conversion context */ - AudioConvert *ac_out; /**< output sample format conversion context */ - ResampleContext *resample; /**< resampling context */ - AudioMix *am; /**< channel mixing context */ - enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */ - - /** - * mix matrix - * only used if avresample_set_matrix() is called before avresample_open() - */ - double *mix_matrix; - - int use_channel_map; - enum RemapPoint remap_point; - ChannelMapInfo ch_map_info; -}; - - -void ff_audio_resample_init_aarch64(ResampleContext *c, - enum AVSampleFormat sample_fmt); -void ff_audio_resample_init_arm(ResampleContext *c, - enum AVSampleFormat sample_fmt); - -#endif /* AVRESAMPLE_INTERNAL_H */ diff --git a/libavresample/libavresample.v b/libavresample/libavresample.v deleted file mode 100644 index d6fc7512ba..0000000000 --- a/libavresample/libavresample.v +++ /dev/null @@ -1,6 +0,0 @@ -LIBAVRESAMPLE_MAJOR { - global: - av*; - local: - *; -}; diff --git a/libavresample/options.c b/libavresample/options.c deleted file mode 100644 index 5f08cd7e52..0000000000 --- a/libavresample/options.c +++ /dev/null @@ -1,113 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 <stdint.h> - -#include "libavutil/mathematics.h" -#include "libavutil/mem.h" -#include "libavutil/opt.h" -#include "avresample.h" -#include "internal.h" -#include "audio_mix.h" - -/** - * @file - * Options definition for AVAudioResampleContext. - */ - -#define OFFSET(x) offsetof(AVAudioResampleContext, x) -#define PARAM AV_OPT_FLAG_AUDIO_PARAM - -static const AVOption avresample_options[] = { - { "in_channel_layout", "Input Channel Layout", OFFSET(in_channel_layout), AV_OPT_TYPE_INT64, { .i64 = 0 }, INT64_MIN, INT64_MAX, PARAM }, - { "in_sample_fmt", "Input Sample Format", OFFSET(in_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM }, - { "in_sample_rate", "Input Sample Rate", OFFSET(in_sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1, INT_MAX, PARAM }, - { "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { .i64 = 0 }, INT64_MIN, INT64_MAX, PARAM }, - { "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM }, - { "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { .i64 = 48000 }, 1, INT_MAX, PARAM }, - { "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { .i64 = AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM, "internal_sample_fmt" }, - {"u8" , "8-bit unsigned integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_U8 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"s16", "16-bit signed integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S16 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"s32", "32-bit signed integer", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S32 }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"flt", "32-bit float", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_FLT }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"dbl", "64-bit double", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_DBL }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"u8p" , "8-bit unsigned integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_U8P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"s16p", "16-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S16P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"s32p", "32-bit signed integer planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_S32P }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"fltp", "32-bit float planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_FLTP }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - {"dblp", "64-bit double planar", 0, AV_OPT_TYPE_CONST, {.i64 = AV_SAMPLE_FMT_DBLP }, INT_MIN, INT_MAX, PARAM, "internal_sample_fmt"}, - { "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { .i64 = AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" }, - { "q8", "16-bit 8.8 Fixed-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_Q8 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" }, - { "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" }, - { "flt", "Floating-Point", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MIX_COEFF_TYPE_FLT }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" }, - { "center_mix_level", "Center Mix Level", OFFSET(center_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = M_SQRT1_2 }, -32.0, 32.0, PARAM }, - { "surround_mix_level", "Surround Mix Level", OFFSET(surround_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = M_SQRT1_2 }, -32.0, 32.0, PARAM }, - { "lfe_mix_level", "LFE Mix Level", OFFSET(lfe_mix_level), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -32.0, 32.0, PARAM }, - { "normalize_mix_level", "Normalize Mix Level", OFFSET(normalize_mix_level), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, PARAM }, - { "force_resampling", "Force Resampling", OFFSET(force_resampling), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, PARAM }, - { "filter_size", "Resampling Filter Size", OFFSET(filter_size), AV_OPT_TYPE_INT, { .i64 = 16 }, 0, 32, /* ??? */ PARAM }, - { "phase_shift", "Resampling Phase Shift", OFFSET(phase_shift), AV_OPT_TYPE_INT, { .i64 = 10 }, 0, 30, /* ??? */ PARAM }, - { "linear_interp", "Use Linear Interpolation", OFFSET(linear_interp), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, PARAM }, - { "cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { .dbl = 0.8 }, 0.0, 1.0, PARAM }, - /* duplicate option in order to work with avconv */ - { "resample_cutoff", "Cutoff Frequency Ratio", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, { .dbl = 0.8 }, 0.0, 1.0, PARAM }, - { "matrix_encoding", "Matrixed Stereo Encoding", OFFSET(matrix_encoding), AV_OPT_TYPE_INT, {.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" }, - { "none", "None", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, - { "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, - { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, - { "filter_type", "Filter Type", OFFSET(filter_type), AV_OPT_TYPE_INT, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER }, AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_KAISER, PARAM, "filter_type" }, - { "cubic", "Cubic", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" }, - { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" }, - { "kaiser", "Kaiser Windowed Sinc", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" }, - { "kaiser_beta", "Kaiser Window Beta", OFFSET(kaiser_beta), AV_OPT_TYPE_INT, { .i64 = 9 }, 2, 16, PARAM }, - { "dither_method", "Dither Method", OFFSET(dither_method), AV_OPT_TYPE_INT, { .i64 = AV_RESAMPLE_DITHER_NONE }, 0, AV_RESAMPLE_DITHER_NB-1, PARAM, "dither_method"}, - {"none", "No Dithering", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_NONE }, INT_MIN, INT_MAX, PARAM, "dither_method"}, - {"rectangular", "Rectangular Dither", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_RECTANGULAR }, INT_MIN, INT_MAX, PARAM, "dither_method"}, - {"triangular", "Triangular Dither", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR }, INT_MIN, INT_MAX, PARAM, "dither_method"}, - {"triangular_hp", "Triangular Dither With High Pass", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_HP }, INT_MIN, INT_MAX, PARAM, "dither_method"}, - {"triangular_ns", "Triangular Dither With Noise Shaping", 0, AV_OPT_TYPE_CONST, { .i64 = AV_RESAMPLE_DITHER_TRIANGULAR_NS }, INT_MIN, INT_MAX, PARAM, "dither_method"}, - { NULL }, -}; - -static const AVClass av_resample_context_class = { - .class_name = "AVAudioResampleContext", - .item_name = av_default_item_name, - .option = avresample_options, - .version = LIBAVUTIL_VERSION_INT, -}; - -AVAudioResampleContext *avresample_alloc_context(void) -{ - AVAudioResampleContext *avr; - - avr = av_mallocz(sizeof(*avr)); - if (!avr) - return NULL; - - avr->av_class = &av_resample_context_class; - av_opt_set_defaults(avr); - - return avr; -} - -const AVClass *avresample_get_class(void) -{ - return &av_resample_context_class; -} diff --git a/libavresample/resample.c b/libavresample/resample.c deleted file mode 100644 index dc14cc2d2a..0000000000 --- a/libavresample/resample.c +++ /dev/null @@ -1,446 +0,0 @@ -/* - * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 "libavutil/common.h" -#include "libavutil/libm.h" -#include "libavutil/log.h" -#include "internal.h" -#include "resample.h" -#include "audio_data.h" - - -/* double template */ -#define CONFIG_RESAMPLE_DBL -#include "resample_template.c" -#undef CONFIG_RESAMPLE_DBL - -/* float template */ -#define CONFIG_RESAMPLE_FLT -#include "resample_template.c" -#undef CONFIG_RESAMPLE_FLT - -/* s32 template */ -#define CONFIG_RESAMPLE_S32 -#include "resample_template.c" -#undef CONFIG_RESAMPLE_S32 - -/* s16 template */ -#include "resample_template.c" - - -/* 0th order modified Bessel function of the first kind. */ -static double bessel(double x) -{ - double v = 1; - double lastv = 0; - double t = 1; - int i; - - x = x * x / 4; - for (i = 1; v != lastv; i++) { - lastv = v; - t *= x / (i * i); - v += t; - } - return v; -} - -/* Build a polyphase filterbank. */ -static int build_filter(ResampleContext *c, double factor) -{ - int ph, i; - double x, y, w; - double *tab; - int tap_count = c->filter_length; - int phase_count = 1 << c->phase_shift; - const int center = (tap_count - 1) / 2; - - tab = av_malloc(tap_count * sizeof(*tab)); - if (!tab) - return AVERROR(ENOMEM); - - for (ph = 0; ph < phase_count; ph++) { - double norm = 0; - for (i = 0; i < tap_count; i++) { - x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; - if (x == 0) y = 1.0; - else y = sin(x) / x; - switch (c->filter_type) { - case AV_RESAMPLE_FILTER_TYPE_CUBIC: { - const float d = -0.5; //first order derivative = -0.5 - x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); - if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x); - else y = d * (-4 + 8 * x - 5 * x*x + x*x*x); - break; - } - case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL: - w = 2.0 * x / (factor * tap_count) + M_PI; - y *= 0.3635819 - 0.4891775 * cos( w) + - 0.1365995 * cos(2 * w) - - 0.0106411 * cos(3 * w); - break; - case AV_RESAMPLE_FILTER_TYPE_KAISER: - w = 2.0 * x / (factor * tap_count * M_PI); - y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0))); - break; - } - - tab[i] = y; - norm += y; - } - /* normalize so that an uniform color remains the same */ - for (i = 0; i < tap_count; i++) - tab[i] = tab[i] / norm; - - c->set_filter(c->filter_bank, tab, ph, tap_count); - } - - av_free(tab); - return 0; -} - -ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr) -{ - ResampleContext *c; - int out_rate = avr->out_sample_rate; - int in_rate = avr->in_sample_rate; - double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0); - int phase_count = 1 << avr->phase_shift; - int felem_size; - - if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P && - avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P && - avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP && - avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) { - av_log(avr, AV_LOG_ERROR, "Unsupported internal format for " - "resampling: %s\n", - av_get_sample_fmt_name(avr->internal_sample_fmt)); - return NULL; - } - c = av_mallocz(sizeof(*c)); - if (!c) - return NULL; - - c->avr = avr; - c->phase_shift = avr->phase_shift; - c->phase_mask = phase_count - 1; - c->linear = avr->linear_interp; - c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1); - c->filter_type = avr->filter_type; - c->kaiser_beta = avr->kaiser_beta; - - switch (avr->internal_sample_fmt) { - case AV_SAMPLE_FMT_DBLP: - c->resample_one = c->linear ? resample_linear_dbl : resample_one_dbl; - c->resample_nearest = resample_nearest_dbl; - c->set_filter = set_filter_dbl; - break; - case AV_SAMPLE_FMT_FLTP: - c->resample_one = c->linear ? resample_linear_flt : resample_one_flt; - c->resample_nearest = resample_nearest_flt; - c->set_filter = set_filter_flt; - break; - case AV_SAMPLE_FMT_S32P: - c->resample_one = c->linear ? resample_linear_s32 : resample_one_s32; - c->resample_nearest = resample_nearest_s32; - c->set_filter = set_filter_s32; - break; - case AV_SAMPLE_FMT_S16P: - c->resample_one = c->linear ? resample_linear_s16 : resample_one_s16; - c->resample_nearest = resample_nearest_s16; - c->set_filter = set_filter_s16; - break; - } - - if (ARCH_AARCH64) - ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt); - if (ARCH_ARM) - ff_audio_resample_init_arm(c, avr->internal_sample_fmt); - - felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt); - c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size); - if (!c->filter_bank) - goto error; - - if (build_filter(c, factor) < 0) - goto error; - - memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size], - c->filter_bank, (c->filter_length - 1) * felem_size); - memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size], - &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size); - - c->compensation_distance = 0; - if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate, - in_rate * (int64_t)phase_count, INT32_MAX / 2)) - goto error; - c->ideal_dst_incr = c->dst_incr; - - c->padding_size = (c->filter_length - 1) / 2; - c->initial_padding_filled = 0; - c->index = 0; - c->frac = 0; - - /* allocate internal buffer */ - c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size, - avr->internal_sample_fmt, - "resample buffer"); - if (!c->buffer) - goto error; - c->buffer->nb_samples = c->padding_size; - c->initial_padding_samples = c->padding_size; - - av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n", - av_get_sample_fmt_name(avr->internal_sample_fmt), - avr->in_sample_rate, avr->out_sample_rate); - - return c; - -error: - ff_audio_data_free(&c->buffer); - av_free(c->filter_bank); - av_free(c); - return NULL; -} - -void ff_audio_resample_free(ResampleContext **c) -{ - if (!*c) - return; - ff_audio_data_free(&(*c)->buffer); - av_free((*c)->filter_bank); - av_freep(c); -} - -int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta, - int compensation_distance) -{ - ResampleContext *c; - - if (compensation_distance < 0) - return AVERROR(EINVAL); - if (!compensation_distance && sample_delta) - return AVERROR(EINVAL); - - if (!avr->resample_needed) { - av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n"); - return AVERROR(EINVAL); - } - c = avr->resample; - c->compensation_distance = compensation_distance; - if (compensation_distance) { - c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * - (int64_t)sample_delta / compensation_distance; - } else { - c->dst_incr = c->ideal_dst_incr; - } - - return 0; -} - -static int resample(ResampleContext *c, void *dst, const void *src, - int *consumed, int src_size, int dst_size, int update_ctx, - int nearest_neighbour) -{ - int dst_index; - unsigned int index = c->index; - int frac = c->frac; - int dst_incr_frac = c->dst_incr % c->src_incr; - int dst_incr = c->dst_incr / c->src_incr; - int compensation_distance = c->compensation_distance; - - if (!dst != !src) - return AVERROR(EINVAL); - - if (nearest_neighbour) { - uint64_t index2 = ((uint64_t)index) << 32; - int64_t incr = (1LL << 32) * c->dst_incr / c->src_incr; - dst_size = FFMIN(dst_size, - (src_size-1-index) * (int64_t)c->src_incr / - c->dst_incr); - - if (dst) { - for(dst_index = 0; dst_index < dst_size; dst_index++) { - c->resample_nearest(dst, dst_index, src, index2 >> 32); - index2 += incr; - } - } else { - dst_index = dst_size; - } - index += dst_index * dst_incr; - index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr; - frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr; - } else { - for (dst_index = 0; dst_index < dst_size; dst_index++) { - int sample_index = index >> c->phase_shift; - - if (sample_index + c->filter_length > src_size) - break; - - if (dst) - c->resample_one(c, dst, dst_index, src, index, frac); - - frac += dst_incr_frac; - index += dst_incr; - if (frac >= c->src_incr) { - frac -= c->src_incr; - index++; - } - if (dst_index + 1 == compensation_distance) { - compensation_distance = 0; - dst_incr_frac = c->ideal_dst_incr % c->src_incr; - dst_incr = c->ideal_dst_incr / c->src_incr; - } - } - } - if (consumed) - *consumed = index >> c->phase_shift; - - if (update_ctx) { - index &= c->phase_mask; - - if (compensation_distance) { - compensation_distance -= dst_index; - if (compensation_distance <= 0) - return AVERROR_BUG; - } - c->frac = frac; - c->index = index; - c->dst_incr = dst_incr_frac + c->src_incr*dst_incr; - c->compensation_distance = compensation_distance; - } - - return dst_index; -} - -int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src) -{ - int ch, in_samples, in_leftover, consumed = 0, out_samples = 0; - int ret = AVERROR(EINVAL); - int nearest_neighbour = (c->compensation_distance == 0 && - c->filter_length == 1 && - c->phase_shift == 0); - - in_samples = src ? src->nb_samples : 0; - in_leftover = c->buffer->nb_samples; - - /* add input samples to the internal buffer */ - if (src) { - ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples); - if (ret < 0) - return ret; - } else if (in_leftover <= c->final_padding_samples) { - /* no remaining samples to flush */ - return 0; - } - - if (!c->initial_padding_filled) { - int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt); - int i; - - if (src && c->buffer->nb_samples < 2 * c->padding_size) - return 0; - - for (i = 0; i < c->padding_size; i++) - for (ch = 0; ch < c->buffer->channels; ch++) { - if (c->buffer->nb_samples > 2 * c->padding_size - i) { - memcpy(c->buffer->data[ch] + bps * i, - c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps); - } else { - memset(c->buffer->data[ch] + bps * i, 0, bps); - } - } - c->initial_padding_filled = 1; - } - - if (!src && !c->final_padding_filled) { - int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt); - int i; - - ret = ff_audio_data_realloc(c->buffer, - FFMAX(in_samples, in_leftover) + - c->padding_size); - if (ret < 0) { - av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n"); - return AVERROR(ENOMEM); - } - - for (i = 0; i < c->padding_size; i++) - for (ch = 0; ch < c->buffer->channels; ch++) { - if (in_leftover > i) { - memcpy(c->buffer->data[ch] + bps * (in_leftover + i), - c->buffer->data[ch] + bps * (in_leftover - i - 1), - bps); - } else { - memset(c->buffer->data[ch] + bps * (in_leftover + i), - 0, bps); - } - } - c->buffer->nb_samples += c->padding_size; - c->final_padding_samples = c->padding_size; - c->final_padding_filled = 1; - } - - - /* calculate output size and reallocate output buffer if needed */ - /* TODO: try to calculate this without the dummy resample() run */ - if (!dst->read_only && dst->allow_realloc) { - out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples, - INT_MAX, 0, nearest_neighbour); - ret = ff_audio_data_realloc(dst, out_samples); - if (ret < 0) { - av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n"); - return ret; - } - } - - /* resample each channel plane */ - for (ch = 0; ch < c->buffer->channels; ch++) { - out_samples = resample(c, (void *)dst->data[ch], - (const void *)c->buffer->data[ch], &consumed, - c->buffer->nb_samples, dst->allocated_samples, - ch + 1 == c->buffer->channels, nearest_neighbour); - } - if (out_samples < 0) { - av_log(c->avr, AV_LOG_ERROR, "error during resampling\n"); - return out_samples; - } - - /* drain consumed samples from the internal buffer */ - ff_audio_data_drain(c->buffer, consumed); - c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0); - - av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n", - in_samples, in_leftover, out_samples, c->buffer->nb_samples); - - dst->nb_samples = out_samples; - return 0; -} - -int avresample_get_delay(AVAudioResampleContext *avr) -{ - ResampleContext *c = avr->resample; - - if (!avr->resample_needed || !avr->resample) - return 0; - - return FFMAX(c->buffer->nb_samples - c->padding_size, 0); -} diff --git a/libavresample/resample.h b/libavresample/resample.h deleted file mode 100644 index be9f562791..0000000000 --- a/libavresample/resample.h +++ /dev/null @@ -1,96 +0,0 @@ -/* - * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> - * - * 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 - */ - -#ifndef AVRESAMPLE_RESAMPLE_H -#define AVRESAMPLE_RESAMPLE_H - -#include "avresample.h" -#include "internal.h" -#include "audio_data.h" - -struct ResampleContext { - AVAudioResampleContext *avr; - AudioData *buffer; - uint8_t *filter_bank; - int filter_length; - int ideal_dst_incr; - int dst_incr; - unsigned int index; - int frac; - int src_incr; - int compensation_distance; - int phase_shift; - int phase_mask; - int linear; - enum AVResampleFilterType filter_type; - int kaiser_beta; - void (*set_filter)(void *filter, double *tab, int phase, int tap_count); - void (*resample_one)(struct ResampleContext *c, void *dst0, - int dst_index, const void *src0, - unsigned int index, int frac); - void (*resample_nearest)(void *dst0, int dst_index, - const void *src0, unsigned int index); - int padding_size; - int initial_padding_filled; - int initial_padding_samples; - int final_padding_filled; - int final_padding_samples; -}; - -/** - * Allocate and initialize a ResampleContext. - * - * The parameters in the AVAudioResampleContext are used to initialize the - * ResampleContext. - * - * @param avr AVAudioResampleContext - * @return newly-allocated ResampleContext - */ -ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr); - -/** - * Free a ResampleContext. - * - * @param c ResampleContext - */ -void ff_audio_resample_free(ResampleContext **c); - -/** - * Resample audio data. - * - * Changes the sample rate. - * - * @par - * All samples in the source data may not be consumed depending on the - * resampling parameters and the size of the output buffer. The unconsumed - * samples are automatically added to the start of the source in the next call. - * If the destination data can be reallocated, that may be done in this function - * in order to fit all available output. If it cannot be reallocated, fewer - * input samples will be consumed in order to have the output fit in the - * destination data buffers. - * - * @param c ResampleContext - * @param dst destination audio data - * @param src source audio data - * @return 0 on success, negative AVERROR code on failure - */ -int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src); - -#endif /* AVRESAMPLE_RESAMPLE_H */ diff --git a/libavresample/resample_template.c b/libavresample/resample_template.c deleted file mode 100644 index 863852a3fd..0000000000 --- a/libavresample/resample_template.c +++ /dev/null @@ -1,118 +0,0 @@ -/* - * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> - * - * 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 <math.h> -#include <stdint.h> - -#include "libavutil/common.h" -#include "internal.h" - -#if defined(CONFIG_RESAMPLE_DBL) -#define SET_TYPE(func) func ## _dbl -#define FELEM double -#define FELEM2 double -#define FELEML double -#define OUT(d, v) d = v -#define DBL_TO_FELEM(d, v) d = v -#elif defined(CONFIG_RESAMPLE_FLT) -#define SET_TYPE(func) func ## _flt -#define FELEM float -#define FELEM2 float -#define FELEML float -#define OUT(d, v) d = v -#define DBL_TO_FELEM(d, v) d = v -#elif defined(CONFIG_RESAMPLE_S32) -#define SET_TYPE(func) func ## _s32 -#define FELEM int32_t -#define FELEM2 int64_t -#define FELEML int64_t -#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30) -#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30))); -#else -#define SET_TYPE(func) func ## _s16 -#define FELEM int16_t -#define FELEM2 int32_t -#define FELEML int64_t -#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15) -#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15))) -#endif - -static void SET_TYPE(resample_nearest)(void *dst0, int dst_index, const void *src0, unsigned int index) -{ - FELEM *dst = dst0; - const FELEM *src = src0; - dst[dst_index] = src[index]; -} - -static void SET_TYPE(resample_linear)(ResampleContext *c, void *dst0, int dst_index, - const void *src0, unsigned int index, int frac) -{ - FELEM *dst = dst0; - const FELEM *src = src0; - int i; - unsigned int sample_index = index >> c->phase_shift; - FELEM2 val = 0; - FELEM *filter = ((FELEM *)c->filter_bank) + - c->filter_length * (index & c->phase_mask); - FELEM2 v2 = 0; - - for (i = 0; i < c->filter_length; i++) { - val += src[sample_index + i] * (FELEM2)filter[i]; - v2 += src[sample_index + i] * (FELEM2)filter[i + c->filter_length]; - } - val += (v2 - val) * (FELEML)frac / c->src_incr; - - OUT(dst[dst_index], val); -} - -static void SET_TYPE(resample_one)(ResampleContext *c, - void *dst0, int dst_index, const void *src0, - unsigned int index, int frac) -{ - FELEM *dst = dst0; - const FELEM *src = src0; - int i; - unsigned int sample_index = index >> c->phase_shift; - FELEM2 val = 0; - FELEM *filter = ((FELEM *)c->filter_bank) + - c->filter_length * (index & c->phase_mask); - - for (i = 0; i < c->filter_length; i++) - val += src[sample_index + i] * (FELEM2)filter[i]; - - OUT(dst[dst_index], val); -} - -static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase, - int tap_count) -{ - int i; - FELEM *filter = ((FELEM *)filter0) + phase * tap_count; - for (i = 0; i < tap_count; i++) { - DBL_TO_FELEM(filter[i], tab[i]); - } -} - -#undef SET_TYPE -#undef FELEM -#undef FELEM2 -#undef FELEML -#undef OUT -#undef DBL_TO_FELEM diff --git a/libavresample/tests/.gitignore b/libavresample/tests/.gitignore deleted file mode 100644 index 1e15871d54..0000000000 --- a/libavresample/tests/.gitignore +++ /dev/null @@ -1 +0,0 @@ -/avresample diff --git a/libavresample/tests/avresample.c b/libavresample/tests/avresample.c deleted file mode 100644 index 8c377bae84..0000000000 --- a/libavresample/tests/avresample.c +++ /dev/null @@ -1,342 +0,0 @@ -/* - * Copyright (c) 2002 Fabrice Bellard - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 <stdint.h> -#include <stdio.h> - -#include "libavutil/avstring.h" -#include "libavutil/common.h" -#include "libavutil/lfg.h" -#include "libavutil/libm.h" -#include "libavutil/log.h" -#include "libavutil/mem.h" -#include "libavutil/opt.h" -#include "libavutil/samplefmt.h" - -#include "libavresample/avresample.h" - -static double dbl_rand(AVLFG *lfg) -{ - return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0; -} - -#define PUT_FUNC(name, fmt, type, expr) \ -static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\ - int channels, int sample, int ch, \ - double v_dbl) \ -{ \ - type v = expr; \ - type **out = (type **)data; \ - if (av_sample_fmt_is_planar(sample_fmt)) \ - out[ch][sample] = v; \ - else \ - out[0][sample * channels + ch] = v; \ -} - -PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128)) -PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15)))) -PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31)))) -PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl) -PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl) - -static void put_sample(void **data, enum AVSampleFormat sample_fmt, - int channels, int sample, int ch, double v_dbl) -{ - switch (av_get_packed_sample_fmt(sample_fmt)) { - case AV_SAMPLE_FMT_U8: - put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl); - break; - case AV_SAMPLE_FMT_S16: - put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl); - break; - case AV_SAMPLE_FMT_S32: - put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl); - break; - case AV_SAMPLE_FMT_FLT: - put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl); - break; - case AV_SAMPLE_FMT_DBL: - put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl); - break; - } -} - -static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt, - int channels, int sample_rate, int nb_samples) -{ - int i, ch, k; - double v, f, a, ampa; - double tabf1[AVRESAMPLE_MAX_CHANNELS]; - double tabf2[AVRESAMPLE_MAX_CHANNELS]; - double taba[AVRESAMPLE_MAX_CHANNELS]; - -#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v); - - k = 0; - - /* 1 second of single freq sine at 1000 Hz */ - a = 0; - for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { - v = sin(a) * 0.30; - for (ch = 0; ch < channels; ch++) - PUT_SAMPLE - a += M_PI * 1000.0 * 2.0 / sample_rate; - } - - /* 1 second of varying frequency between 100 and 10000 Hz */ - a = 0; - for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { - v = sin(a) * 0.30; - for (ch = 0; ch < channels; ch++) - PUT_SAMPLE - f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate); - a += M_PI * f * 2.0 / sample_rate; - } - - /* 0.5 second of low amplitude white noise */ - for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { - v = dbl_rand(rnd) * 0.30; - for (ch = 0; ch < channels; ch++) - PUT_SAMPLE - } - - /* 0.5 second of high amplitude white noise */ - for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { - v = dbl_rand(rnd); - for (ch = 0; ch < channels; ch++) - PUT_SAMPLE - } - - /* 1 second of unrelated ramps for each channel */ - for (ch = 0; ch < channels; ch++) { - taba[ch] = 0; - tabf1[ch] = 100 + av_lfg_get(rnd) % 5000; - tabf2[ch] = 100 + av_lfg_get(rnd) % 5000; - } - for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { - for (ch = 0; ch < channels; ch++) { - v = sin(taba[ch]) * 0.30; - PUT_SAMPLE - f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate); - taba[ch] += M_PI * f * 2.0 / sample_rate; - } - } - - /* 2 seconds of 500 Hz with varying volume */ - a = 0; - ampa = 0; - for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) { - for (ch = 0; ch < channels; ch++) { - double amp = (1.0 + sin(ampa)) * 0.15; - if (ch & 1) - amp = 0.30 - amp; - v = sin(a) * amp; - PUT_SAMPLE - a += M_PI * 500.0 * 2.0 / sample_rate; - ampa += M_PI * 2.0 / sample_rate; - } - } -} - -/* formats, rates, and layouts are ordered for priority in testing. - e.g. 'avresample-test 4 2 2' will test all input/output combinations of - S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */ - -static const enum AVSampleFormat formats[] = { - AV_SAMPLE_FMT_S16, - AV_SAMPLE_FMT_FLTP, - AV_SAMPLE_FMT_S16P, - AV_SAMPLE_FMT_FLT, - AV_SAMPLE_FMT_S32P, - AV_SAMPLE_FMT_S32, - AV_SAMPLE_FMT_U8P, - AV_SAMPLE_FMT_U8, - AV_SAMPLE_FMT_DBLP, - AV_SAMPLE_FMT_DBL, -}; - -static const int rates[] = { - 48000, - 44100, - 16000 -}; - -static const uint64_t layouts[] = { - AV_CH_LAYOUT_STEREO, - AV_CH_LAYOUT_MONO, - AV_CH_LAYOUT_5POINT1, - AV_CH_LAYOUT_7POINT1, -}; - -int main(int argc, char **argv) -{ - AVAudioResampleContext *s; - AVLFG rnd; - int ret = 0; - uint8_t *in_buf = NULL; - uint8_t *out_buf = NULL; - unsigned int in_buf_size; - unsigned int out_buf_size; - uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; - uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; - int in_linesize; - int out_linesize; - uint64_t in_ch_layout; - int in_channels; - enum AVSampleFormat in_fmt; - int in_rate; - uint64_t out_ch_layout; - int out_channels; - enum AVSampleFormat out_fmt; - int out_rate; - int num_formats, num_rates, num_layouts; - int i, j, k, l, m, n; - - num_formats = 2; - num_rates = 2; - num_layouts = 2; - if (argc > 1) { - if (!av_strncasecmp(argv[1], "-h", 3)) { - av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> " - "[<num sample rates> [<num channel layouts>]]]\n" - "Default is 2 2 2\n"); - return 0; - } - num_formats = strtol(argv[1], NULL, 0); - num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats)); - } - if (argc > 2) { - num_rates = strtol(argv[2], NULL, 0); - num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates)); - } - if (argc > 3) { - num_layouts = strtol(argv[3], NULL, 0); - num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts)); - } - - av_log_set_level(AV_LOG_DEBUG); - - av_lfg_init(&rnd, 0xC0FFEE); - - in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6, - AV_SAMPLE_FMT_DBLP, 0); - out_buf_size = in_buf_size; - - in_buf = av_malloc(in_buf_size); - if (!in_buf) - goto end; - out_buf = av_malloc(out_buf_size); - if (!out_buf) - goto end; - - s = avresample_alloc_context(); - if (!s) { - av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n"); - ret = 1; - goto end; - } - - for (i = 0; i < num_formats; i++) { - in_fmt = formats[i]; - for (k = 0; k < num_layouts; k++) { - in_ch_layout = layouts[k]; - in_channels = av_get_channel_layout_nb_channels(in_ch_layout); - for (m = 0; m < num_rates; m++) { - in_rate = rates[m]; - - ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf, - in_channels, in_rate * 6, - in_fmt, 0); - if (ret < 0) { - av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n"); - goto end; - } - audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6); - - for (j = 0; j < num_formats; j++) { - out_fmt = formats[j]; - for (l = 0; l < num_layouts; l++) { - out_ch_layout = layouts[l]; - out_channels = av_get_channel_layout_nb_channels(out_ch_layout); - for (n = 0; n < num_rates; n++) { - out_rate = rates[n]; - - av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n", - av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt), - in_channels, out_channels, in_rate, out_rate); - - ret = av_samples_fill_arrays(out_data, &out_linesize, - out_buf, out_channels, - out_rate * 6, out_fmt, 0); - if (ret < 0) { - av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n"); - goto end; - } - - av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0); - av_opt_set_int(s, "in_sample_fmt", in_fmt, 0); - av_opt_set_int(s, "in_sample_rate", in_rate, 0); - av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0); - av_opt_set_int(s, "out_sample_fmt", out_fmt, 0); - av_opt_set_int(s, "out_sample_rate", out_rate, 0); - - av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); - - ret = avresample_open(s); - if (ret < 0) { - av_log(s, AV_LOG_ERROR, "Error opening context\n"); - goto end; - } - - ret = avresample_convert(s, out_data, out_linesize, out_rate * 6, - in_data, in_linesize, in_rate * 6); - if (ret < 0) { - char errbuf[256]; - av_strerror(ret, errbuf, sizeof(errbuf)); - av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf); - goto end; - } - av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n", - in_rate * 6, ret); - if (avresample_get_delay(s) > 0) - av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n", - avresample_get_delay(s)); - if (avresample_available(s) > 0) - av_log(NULL, AV_LOG_INFO, "%d samples available for output\n", - avresample_available(s)); - av_log(NULL, AV_LOG_INFO, "\n"); - - avresample_close(s); - } - } - } - } - } - } - - ret = 0; - -end: - av_freep(&in_buf); - av_freep(&out_buf); - avresample_free(&s); - return ret; -} diff --git a/libavresample/utils.c b/libavresample/utils.c deleted file mode 100644 index b4fb906556..0000000000 --- a/libavresample/utils.c +++ /dev/null @@ -1,793 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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 "libavutil/common.h" -#include "libavutil/dict.h" -// #include "libavutil/error.h" -#include "libavutil/frame.h" -#include "libavutil/log.h" -#include "libavutil/mem.h" -#include "libavutil/opt.h" - -#include "avresample.h" -#include "internal.h" -#include "audio_data.h" -#include "audio_convert.h" -#include "audio_mix.h" -#include "resample.h" - -int avresample_open(AVAudioResampleContext *avr) -{ - int ret; - - if (avresample_is_open(avr)) { - av_log(avr, AV_LOG_ERROR, "The resampling context is already open.\n"); - return AVERROR(EINVAL); - } - - /* set channel mixing parameters */ - avr->in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); - if (avr->in_channels <= 0 || avr->in_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(avr, AV_LOG_ERROR, "Invalid input channel layout: %"PRIu64"\n", - avr->in_channel_layout); - return AVERROR(EINVAL); - } - avr->out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); - if (avr->out_channels <= 0 || avr->out_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(avr, AV_LOG_ERROR, "Invalid output channel layout: %"PRIu64"\n", - avr->out_channel_layout); - return AVERROR(EINVAL); - } - avr->resample_channels = FFMIN(avr->in_channels, avr->out_channels); - avr->downmix_needed = avr->in_channels > avr->out_channels; - avr->upmix_needed = avr->out_channels > avr->in_channels || - (!avr->downmix_needed && (avr->mix_matrix || - avr->in_channel_layout != avr->out_channel_layout)); - avr->mixing_needed = avr->downmix_needed || avr->upmix_needed; - - /* set resampling parameters */ - avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate || - avr->force_resampling; - - /* select internal sample format if not specified by the user */ - if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE && - (avr->mixing_needed || avr->resample_needed)) { - enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt); - enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); - int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt), - av_get_bytes_per_sample(out_fmt)); - if (max_bps <= 2) { - avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P; - } else if (avr->mixing_needed) { - avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; - } else { - if (max_bps <= 4) { - if (in_fmt == AV_SAMPLE_FMT_S32P || - out_fmt == AV_SAMPLE_FMT_S32P) { - if (in_fmt == AV_SAMPLE_FMT_FLTP || - out_fmt == AV_SAMPLE_FMT_FLTP) { - /* if one is s32 and the other is flt, use dbl */ - avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP; - } else { - /* if one is s32 and the other is s32, s16, or u8, use s32 */ - avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P; - } - } else { - /* if one is flt and the other is flt, s16 or u8, use flt */ - avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP; - } - } else { - /* if either is dbl, use dbl */ - avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP; - } - } - av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n", - av_get_sample_fmt_name(avr->internal_sample_fmt)); - } - - /* we may need to add an extra conversion in order to remap channels if - the output format is not planar */ - if (avr->use_channel_map && !avr->mixing_needed && !avr->resample_needed && - !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels)) { - avr->internal_sample_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt); - } - - /* set sample format conversion parameters */ - if (avr->resample_needed || avr->mixing_needed) - avr->in_convert_needed = avr->in_sample_fmt != avr->internal_sample_fmt; - else - avr->in_convert_needed = avr->use_channel_map && - !ff_sample_fmt_is_planar(avr->out_sample_fmt, avr->out_channels); - - if (avr->resample_needed || avr->mixing_needed || avr->in_convert_needed) - avr->out_convert_needed = avr->internal_sample_fmt != avr->out_sample_fmt; - else - avr->out_convert_needed = avr->in_sample_fmt != avr->out_sample_fmt; - - avr->in_copy_needed = !avr->in_convert_needed && (avr->mixing_needed || - (avr->use_channel_map && avr->resample_needed)); - - if (avr->use_channel_map) { - if (avr->in_copy_needed) { - avr->remap_point = REMAP_IN_COPY; - av_log(avr, AV_LOG_TRACE, "remap channels during in_copy\n"); - } else if (avr->in_convert_needed) { - avr->remap_point = REMAP_IN_CONVERT; - av_log(avr, AV_LOG_TRACE, "remap channels during in_convert\n"); - } else if (avr->out_convert_needed) { - avr->remap_point = REMAP_OUT_CONVERT; - av_log(avr, AV_LOG_TRACE, "remap channels during out_convert\n"); - } else { - avr->remap_point = REMAP_OUT_COPY; - av_log(avr, AV_LOG_TRACE, "remap channels during out_copy\n"); - } - -#ifdef DEBUG - { - int ch; - av_log(avr, AV_LOG_TRACE, "output map: "); - if (avr->ch_map_info.do_remap) - for (ch = 0; ch < avr->in_channels; ch++) - av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_map[ch]); - else - av_log(avr, AV_LOG_TRACE, "n/a"); - av_log(avr, AV_LOG_TRACE, "\n"); - av_log(avr, AV_LOG_TRACE, "copy map: "); - if (avr->ch_map_info.do_copy) - for (ch = 0; ch < avr->in_channels; ch++) - av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_copy[ch]); - else - av_log(avr, AV_LOG_TRACE, "n/a"); - av_log(avr, AV_LOG_TRACE, "\n"); - av_log(avr, AV_LOG_TRACE, "zero map: "); - if (avr->ch_map_info.do_zero) - for (ch = 0; ch < avr->in_channels; ch++) - av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.channel_zero[ch]); - else - av_log(avr, AV_LOG_TRACE, "n/a"); - av_log(avr, AV_LOG_TRACE, "\n"); - av_log(avr, AV_LOG_TRACE, "input map: "); - for (ch = 0; ch < avr->in_channels; ch++) - av_log(avr, AV_LOG_TRACE, " % 2d", avr->ch_map_info.input_map[ch]); - av_log(avr, AV_LOG_TRACE, "\n"); - } -#endif - } else - avr->remap_point = REMAP_NONE; - - /* allocate buffers */ - if (avr->in_copy_needed || avr->in_convert_needed) { - avr->in_buffer = ff_audio_data_alloc(FFMAX(avr->in_channels, avr->out_channels), - 0, avr->internal_sample_fmt, - "in_buffer"); - if (!avr->in_buffer) { - ret = AVERROR(EINVAL); - goto error; - } - } - if (avr->resample_needed) { - avr->resample_out_buffer = ff_audio_data_alloc(avr->out_channels, - 1024, avr->internal_sample_fmt, - "resample_out_buffer"); - if (!avr->resample_out_buffer) { - ret = AVERROR(EINVAL); - goto error; - } - } - if (avr->out_convert_needed) { - avr->out_buffer = ff_audio_data_alloc(avr->out_channels, 0, - avr->out_sample_fmt, "out_buffer"); - if (!avr->out_buffer) { - ret = AVERROR(EINVAL); - goto error; - } - } - avr->out_fifo = av_audio_fifo_alloc(avr->out_sample_fmt, avr->out_channels, - 1024); - if (!avr->out_fifo) { - ret = AVERROR(ENOMEM); - goto error; - } - - /* setup contexts */ - if (avr->in_convert_needed) { - avr->ac_in = ff_audio_convert_alloc(avr, avr->internal_sample_fmt, - avr->in_sample_fmt, avr->in_channels, - avr->in_sample_rate, - avr->remap_point == REMAP_IN_CONVERT); - if (!avr->ac_in) { - ret = AVERROR(ENOMEM); - goto error; - } - } - if (avr->out_convert_needed) { - enum AVSampleFormat src_fmt; - if (avr->in_convert_needed) - src_fmt = avr->internal_sample_fmt; - else - src_fmt = avr->in_sample_fmt; - avr->ac_out = ff_audio_convert_alloc(avr, avr->out_sample_fmt, src_fmt, - avr->out_channels, - avr->out_sample_rate, - avr->remap_point == REMAP_OUT_CONVERT); - if (!avr->ac_out) { - ret = AVERROR(ENOMEM); - goto error; - } - } - if (avr->resample_needed) { - avr->resample = ff_audio_resample_init(avr); - if (!avr->resample) { - ret = AVERROR(ENOMEM); - goto error; - } - } - if (avr->mixing_needed) { - avr->am = ff_audio_mix_alloc(avr); - if (!avr->am) { - ret = AVERROR(ENOMEM); - goto error; - } - } - - return 0; - -error: - avresample_close(avr); - return ret; -} - -int avresample_is_open(AVAudioResampleContext *avr) -{ - return !!avr->out_fifo; -} - -void avresample_close(AVAudioResampleContext *avr) -{ - ff_audio_data_free(&avr->in_buffer); - ff_audio_data_free(&avr->resample_out_buffer); - ff_audio_data_free(&avr->out_buffer); - av_audio_fifo_free(avr->out_fifo); - avr->out_fifo = NULL; - ff_audio_convert_free(&avr->ac_in); - ff_audio_convert_free(&avr->ac_out); - ff_audio_resample_free(&avr->resample); - ff_audio_mix_free(&avr->am); - av_freep(&avr->mix_matrix); - - avr->use_channel_map = 0; -} - -void avresample_free(AVAudioResampleContext **avr) -{ - if (!*avr) - return; - avresample_close(*avr); - av_opt_free(*avr); - av_freep(avr); -} - -static int handle_buffered_output(AVAudioResampleContext *avr, - AudioData *output, AudioData *converted) -{ - int ret; - - if (!output || av_audio_fifo_size(avr->out_fifo) > 0 || - (converted && output->allocated_samples < converted->nb_samples)) { - if (converted) { - /* if there are any samples in the output FIFO or if the - user-supplied output buffer is not large enough for all samples, - we add to the output FIFO */ - av_log(avr, AV_LOG_TRACE, "[FIFO] add %s to out_fifo\n", converted->name); - ret = ff_audio_data_add_to_fifo(avr->out_fifo, converted, 0, - converted->nb_samples); - if (ret < 0) - return ret; - } - - /* if the user specified an output buffer, read samples from the output - FIFO to the user output */ - if (output && output->allocated_samples > 0) { - av_log(avr, AV_LOG_TRACE, "[FIFO] read from out_fifo to output\n"); - av_log(avr, AV_LOG_TRACE, "[end conversion]\n"); - return ff_audio_data_read_from_fifo(avr->out_fifo, output, - output->allocated_samples); - } - } else if (converted) { - /* copy directly to output if it is large enough or there is not any - data in the output FIFO */ - av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", converted->name); - output->nb_samples = 0; - ret = ff_audio_data_copy(output, converted, - avr->remap_point == REMAP_OUT_COPY ? - &avr->ch_map_info : NULL); - if (ret < 0) - return ret; - av_log(avr, AV_LOG_TRACE, "[end conversion]\n"); - return output->nb_samples; - } - av_log(avr, AV_LOG_TRACE, "[end conversion]\n"); - return 0; -} - -int attribute_align_arg avresample_convert(AVAudioResampleContext *avr, - uint8_t **output, int out_plane_size, - int out_samples, - uint8_t * const *input, - int in_plane_size, int in_samples) -{ - AudioData input_buffer; - AudioData output_buffer; - AudioData *current_buffer; - int ret, direct_output; - - /* reset internal buffers */ - if (avr->in_buffer) { - avr->in_buffer->nb_samples = 0; - ff_audio_data_set_channels(avr->in_buffer, - avr->in_buffer->allocated_channels); - } - if (avr->resample_out_buffer) { - avr->resample_out_buffer->nb_samples = 0; - ff_audio_data_set_channels(avr->resample_out_buffer, - avr->resample_out_buffer->allocated_channels); - } - if (avr->out_buffer) { - avr->out_buffer->nb_samples = 0; - ff_audio_data_set_channels(avr->out_buffer, - avr->out_buffer->allocated_channels); - } - - av_log(avr, AV_LOG_TRACE, "[start conversion]\n"); - - /* initialize output_buffer with output data */ - direct_output = output && av_audio_fifo_size(avr->out_fifo) == 0; - if (output) { - ret = ff_audio_data_init(&output_buffer, output, out_plane_size, - avr->out_channels, out_samples, - avr->out_sample_fmt, 0, "output"); - if (ret < 0) - return ret; - output_buffer.nb_samples = 0; - } - - if (input) { - /* initialize input_buffer with input data */ - ret = ff_audio_data_init(&input_buffer, input, in_plane_size, - avr->in_channels, in_samples, - avr->in_sample_fmt, 1, "input"); - if (ret < 0) - return ret; - current_buffer = &input_buffer; - - if (avr->upmix_needed && !avr->in_convert_needed && !avr->resample_needed && - !avr->out_convert_needed && direct_output && out_samples >= in_samples) { - /* in some rare cases we can copy input to output and upmix - directly in the output buffer */ - av_log(avr, AV_LOG_TRACE, "[copy] %s to output\n", current_buffer->name); - ret = ff_audio_data_copy(&output_buffer, current_buffer, - avr->remap_point == REMAP_OUT_COPY ? - &avr->ch_map_info : NULL); - if (ret < 0) - return ret; - current_buffer = &output_buffer; - } else if (avr->remap_point == REMAP_OUT_COPY && - (!direct_output || out_samples < in_samples)) { - /* if remapping channels during output copy, we may need to - * use an intermediate buffer in order to remap before adding - * samples to the output fifo */ - av_log(avr, AV_LOG_TRACE, "[copy] %s to out_buffer\n", current_buffer->name); - ret = ff_audio_data_copy(avr->out_buffer, current_buffer, - &avr->ch_map_info); - if (ret < 0) - return ret; - current_buffer = avr->out_buffer; - } else if (avr->in_copy_needed || avr->in_convert_needed) { - /* if needed, copy or convert input to in_buffer, and downmix if - applicable */ - if (avr->in_convert_needed) { - ret = ff_audio_data_realloc(avr->in_buffer, - current_buffer->nb_samples); - if (ret < 0) - return ret; - av_log(avr, AV_LOG_TRACE, "[convert] %s to in_buffer\n", current_buffer->name); - ret = ff_audio_convert(avr->ac_in, avr->in_buffer, - current_buffer); - if (ret < 0) - return ret; - } else { - av_log(avr, AV_LOG_TRACE, "[copy] %s to in_buffer\n", current_buffer->name); - ret = ff_audio_data_copy(avr->in_buffer, current_buffer, - avr->remap_point == REMAP_IN_COPY ? - &avr->ch_map_info : NULL); - if (ret < 0) - return ret; - } - ff_audio_data_set_channels(avr->in_buffer, avr->in_channels); - if (avr->downmix_needed) { - av_log(avr, AV_LOG_TRACE, "[downmix] in_buffer\n"); - ret = ff_audio_mix(avr->am, avr->in_buffer); - if (ret < 0) - return ret; - } - current_buffer = avr->in_buffer; - } - } else { - /* flush resampling buffer and/or output FIFO if input is NULL */ - if (!avr->resample_needed) - return handle_buffered_output(avr, output ? &output_buffer : NULL, - NULL); - current_buffer = NULL; - } - - if (avr->resample_needed) { - AudioData *resample_out; - - if (!avr->out_convert_needed && direct_output && out_samples > 0) - resample_out = &output_buffer; - else - resample_out = avr->resample_out_buffer; - av_log(avr, AV_LOG_TRACE, "[resample] %s to %s\n", - current_buffer ? current_buffer->name : "null", - resample_out->name); - ret = ff_audio_resample(avr->resample, resample_out, - current_buffer); - if (ret < 0) - return ret; - - /* if resampling did not produce any samples, just return 0 */ - if (resample_out->nb_samples == 0) { - av_log(avr, AV_LOG_TRACE, "[end conversion]\n"); - return 0; - } - - current_buffer = resample_out; - } - - if (avr->upmix_needed) { - av_log(avr, AV_LOG_TRACE, "[upmix] %s\n", current_buffer->name); - ret = ff_audio_mix(avr->am, current_buffer); - if (ret < 0) - return ret; - } - - /* if we resampled or upmixed directly to output, return here */ - if (current_buffer == &output_buffer) { - av_log(avr, AV_LOG_TRACE, "[end conversion]\n"); - return current_buffer->nb_samples; - } - - if (avr->out_convert_needed) { - if (direct_output && out_samples >= current_buffer->nb_samples) { - /* convert directly to output */ - av_log(avr, AV_LOG_TRACE, "[convert] %s to output\n", current_buffer->name); - ret = ff_audio_convert(avr->ac_out, &output_buffer, current_buffer); - if (ret < 0) - return ret; - - av_log(avr, AV_LOG_TRACE, "[end conversion]\n"); - return output_buffer.nb_samples; - } else { - ret = ff_audio_data_realloc(avr->out_buffer, - current_buffer->nb_samples); - if (ret < 0) - return ret; - av_log(avr, AV_LOG_TRACE, "[convert] %s to out_buffer\n", current_buffer->name); - ret = ff_audio_convert(avr->ac_out, avr->out_buffer, - current_buffer); - if (ret < 0) - return ret; - current_buffer = avr->out_buffer; - } - } - - return handle_buffered_output(avr, output ? &output_buffer : NULL, - current_buffer); -} - -int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in) -{ - if (avresample_is_open(avr)) { - avresample_close(avr); - } - - if (in) { - avr->in_channel_layout = in->channel_layout; - avr->in_sample_rate = in->sample_rate; - avr->in_sample_fmt = in->format; - } - - if (out) { - avr->out_channel_layout = out->channel_layout; - avr->out_sample_rate = out->sample_rate; - avr->out_sample_fmt = out->format; - } - - return 0; -} - -static int config_changed(AVAudioResampleContext *avr, - AVFrame *out, AVFrame *in) -{ - int ret = 0; - - if (in) { - if (avr->in_channel_layout != in->channel_layout || - avr->in_sample_rate != in->sample_rate || - avr->in_sample_fmt != in->format) { - ret |= AVERROR_INPUT_CHANGED; - } - } - - if (out) { - if (avr->out_channel_layout != out->channel_layout || - avr->out_sample_rate != out->sample_rate || - avr->out_sample_fmt != out->format) { - ret |= AVERROR_OUTPUT_CHANGED; - } - } - - return ret; -} - -static inline int convert_frame(AVAudioResampleContext *avr, - AVFrame *out, AVFrame *in) -{ - int ret; - uint8_t **out_data = NULL, **in_data = NULL; - int out_linesize = 0, in_linesize = 0; - int out_nb_samples = 0, in_nb_samples = 0; - - if (out) { - out_data = out->extended_data; - out_linesize = out->linesize[0]; - out_nb_samples = out->nb_samples; - } - - if (in) { - in_data = in->extended_data; - in_linesize = in->linesize[0]; - in_nb_samples = in->nb_samples; - } - - ret = avresample_convert(avr, out_data, out_linesize, - out_nb_samples, - in_data, in_linesize, - in_nb_samples); - - if (ret < 0) { - if (out) - out->nb_samples = 0; - return ret; - } - - if (out) - out->nb_samples = ret; - - return 0; -} - -static inline int available_samples(AVFrame *out) -{ - int samples; - int bytes_per_sample = av_get_bytes_per_sample(out->format); - if (!bytes_per_sample) - return AVERROR(EINVAL); - - samples = out->linesize[0] / bytes_per_sample; - if (av_sample_fmt_is_planar(out->format)) { - return samples; - } else { - int channels = av_get_channel_layout_nb_channels(out->channel_layout); - return samples / channels; - } -} - -int avresample_convert_frame(AVAudioResampleContext *avr, - AVFrame *out, AVFrame *in) -{ - int ret, setup = 0; - - if (!avresample_is_open(avr)) { - if ((ret = avresample_config(avr, out, in)) < 0) - return ret; - if ((ret = avresample_open(avr)) < 0) - return ret; - setup = 1; - } else { - // return as is or reconfigure for input changes? - if ((ret = config_changed(avr, out, in))) - return ret; - } - - if (out) { - if (!out->linesize[0]) { - out->nb_samples = avresample_get_out_samples(avr, in->nb_samples); - if ((ret = av_frame_get_buffer(out, 0)) < 0) { - if (setup) - avresample_close(avr); - return ret; - } - } else { - if (!out->nb_samples) - out->nb_samples = available_samples(out); - } - } - - return convert_frame(avr, out, in); -} - -int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix, - int stride) -{ - int in_channels, out_channels, i, o; - - if (avr->am) - return ff_audio_mix_get_matrix(avr->am, matrix, stride); - - in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); - out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); - - if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS || - out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n"); - return AVERROR(EINVAL); - } - - if (!avr->mix_matrix) { - av_log(avr, AV_LOG_ERROR, "matrix is not set\n"); - return AVERROR(EINVAL); - } - - for (o = 0; o < out_channels; o++) - for (i = 0; i < in_channels; i++) - matrix[o * stride + i] = avr->mix_matrix[o * in_channels + i]; - - return 0; -} - -int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, - int stride) -{ - int in_channels, out_channels, i, o; - - if (avr->am) - return ff_audio_mix_set_matrix(avr->am, matrix, stride); - - in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); - out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); - - if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS || - out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n"); - return AVERROR(EINVAL); - } - - if (avr->mix_matrix) - av_freep(&avr->mix_matrix); - avr->mix_matrix = av_malloc(in_channels * out_channels * - sizeof(*avr->mix_matrix)); - if (!avr->mix_matrix) - return AVERROR(ENOMEM); - - for (o = 0; o < out_channels; o++) - for (i = 0; i < in_channels; i++) - avr->mix_matrix[o * in_channels + i] = matrix[o * stride + i]; - - return 0; -} - -int avresample_set_channel_mapping(AVAudioResampleContext *avr, - const int *channel_map) -{ - ChannelMapInfo *info = &avr->ch_map_info; - int in_channels, ch, i; - - in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); - if (in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS) { - av_log(avr, AV_LOG_ERROR, "Invalid input channel layout\n"); - return AVERROR(EINVAL); - } - - memset(info, 0, sizeof(*info)); - memset(info->input_map, -1, sizeof(info->input_map)); - - for (ch = 0; ch < in_channels; ch++) { - if (channel_map[ch] >= in_channels) { - av_log(avr, AV_LOG_ERROR, "Invalid channel map\n"); - return AVERROR(EINVAL); - } - if (channel_map[ch] < 0) { - info->channel_zero[ch] = 1; - info->channel_map[ch] = -1; - info->do_zero = 1; - } else if (info->input_map[channel_map[ch]] >= 0) { - info->channel_copy[ch] = info->input_map[channel_map[ch]]; - info->channel_map[ch] = -1; - info->do_copy = 1; - } else { - info->channel_map[ch] = channel_map[ch]; - info->input_map[channel_map[ch]] = ch; - info->do_remap = 1; - } - } - /* Fill-in unmapped input channels with unmapped output channels. - This is used when remapping during conversion from interleaved to - planar format. */ - for (ch = 0, i = 0; ch < in_channels && i < in_channels; ch++, i++) { - while (ch < in_channels && info->input_map[ch] >= 0) - ch++; - while (i < in_channels && info->channel_map[i] >= 0) - i++; - if (ch >= in_channels || i >= in_channels) - break; - info->input_map[ch] = i; - } - - avr->use_channel_map = 1; - return 0; -} - -int avresample_available(AVAudioResampleContext *avr) -{ - return av_audio_fifo_size(avr->out_fifo); -} - -int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples) -{ - int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples; - - if (avr->resample_needed) { - samples = av_rescale_rnd(samples, - avr->out_sample_rate, - avr->in_sample_rate, - AV_ROUND_UP); - } - - samples += avresample_available(avr); - - if (samples > INT_MAX) - return AVERROR(EINVAL); - - return samples; -} - -int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples) -{ - if (!output) - return av_audio_fifo_drain(avr->out_fifo, nb_samples); - return av_audio_fifo_read(avr->out_fifo, (void**)output, nb_samples); -} - -unsigned avresample_version(void) -{ - return LIBAVRESAMPLE_VERSION_INT; -} - -const char *avresample_license(void) -{ -#define LICENSE_PREFIX "libavresample license: " - return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1; -} - -const char *avresample_configuration(void) -{ - return FFMPEG_CONFIGURATION; -} diff --git a/libavresample/version.h b/libavresample/version.h deleted file mode 100644 index d5d3ea82b1..0000000000 --- a/libavresample/version.h +++ /dev/null @@ -1,50 +0,0 @@ -/* - * 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 - */ - -#ifndef AVRESAMPLE_VERSION_H -#define AVRESAMPLE_VERSION_H - -/** - * @file - * @ingroup lavr - * Libavresample version macros. - */ - -#include "libavutil/version.h" - -#define LIBAVRESAMPLE_VERSION_MAJOR 4 -#define LIBAVRESAMPLE_VERSION_MINOR 0 -#define LIBAVRESAMPLE_VERSION_MICRO 0 - -#define LIBAVRESAMPLE_VERSION_INT AV_VERSION_INT(LIBAVRESAMPLE_VERSION_MAJOR, \ - LIBAVRESAMPLE_VERSION_MINOR, \ - LIBAVRESAMPLE_VERSION_MICRO) -#define LIBAVRESAMPLE_VERSION AV_VERSION(LIBAVRESAMPLE_VERSION_MAJOR, \ - LIBAVRESAMPLE_VERSION_MINOR, \ - LIBAVRESAMPLE_VERSION_MICRO) -#define LIBAVRESAMPLE_BUILD LIBAVRESAMPLE_VERSION_INT - -#define LIBAVRESAMPLE_IDENT "Lavr" AV_STRINGIFY(LIBAVRESAMPLE_VERSION) - -/** - * FF_API_* defines may be placed below to indicate public API that will be - * dropped at a future version bump. The defines themselves are not part of - * the public API and may change, break or disappear at any time. - */ - -#endif /* AVRESAMPLE_VERSION_H */ diff --git a/libavresample/x86/Makefile b/libavresample/x86/Makefile deleted file mode 100644 index 55b709ce36..0000000000 --- a/libavresample/x86/Makefile +++ /dev/null @@ -1,9 +0,0 @@ -OBJS += x86/audio_convert_init.o \ - x86/audio_mix_init.o \ - x86/dither_init.o \ - -OBJS-$(CONFIG_XMM_CLOBBER_TEST) += x86/w64xmmtest.o - -X86ASM-OBJS += x86/audio_convert.o \ - x86/audio_mix.o \ - x86/dither.o \ diff --git a/libavresample/x86/audio_convert.asm b/libavresample/x86/audio_convert.asm deleted file mode 100644 index c6a5015282..0000000000 --- a/libavresample/x86/audio_convert.asm +++ /dev/null @@ -1,1261 +0,0 @@ -;****************************************************************************** -;* x86 optimized Format Conversion Utils -;* Copyright (c) 2008 Loren Merritt -;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> -;* -;* 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 "libavutil/x86/x86util.asm" -%include "util.asm" - -SECTION_RODATA 32 - -pf_s32_inv_scale: times 8 dd 0x30000000 -pf_s32_scale: times 8 dd 0x4f000000 -pf_s32_clip: times 8 dd 0x4effffff -pf_s16_inv_scale: times 4 dd 0x38000000 -pf_s16_scale: times 4 dd 0x47000000 -pb_shuf_unpack_even: db -1, -1, 0, 1, -1, -1, 2, 3, -1, -1, 8, 9, -1, -1, 10, 11 -pb_shuf_unpack_odd: db -1, -1, 4, 5, -1, -1, 6, 7, -1, -1, 12, 13, -1, -1, 14, 15 -pb_interleave_words: SHUFFLE_MASK_W 0, 4, 1, 5, 2, 6, 3, 7 -pb_deinterleave_words: SHUFFLE_MASK_W 0, 2, 4, 6, 1, 3, 5, 7 -pw_zero_even: times 4 dw 0x0000, 0xffff - -SECTION .text - -;------------------------------------------------------------------------------ -; void ff_conv_s16_to_s32(int32_t *dst, const int16_t *src, int len); -;------------------------------------------------------------------------------ - -INIT_XMM sse2 -cglobal conv_s16_to_s32, 3,3,3, dst, src, len - lea lenq, [2*lend] - lea dstq, [dstq+2*lenq] - add srcq, lenq - neg lenq -.loop: - mova m2, [srcq+lenq] - pxor m0, m0 - pxor m1, m1 - punpcklwd m0, m2 - punpckhwd m1, m2 - mova [dstq+2*lenq ], m0 - mova [dstq+2*lenq+mmsize], m1 - add lenq, mmsize - jl .loop - REP_RET - -;------------------------------------------------------------------------------ -; void ff_conv_s16_to_flt(float *dst, const int16_t *src, int len); -;------------------------------------------------------------------------------ - -%macro CONV_S16_TO_FLT 0 -cglobal conv_s16_to_flt, 3,3,3, dst, src, len - lea lenq, [2*lend] - add srcq, lenq - lea dstq, [dstq + 2*lenq] - neg lenq - mova m2, [pf_s16_inv_scale] - ALIGN 16 -.loop: - mova m0, [srcq+lenq] - S16_TO_S32_SX 0, 1 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - mulps m0, m2 - mulps m1, m2 - mova [dstq+2*lenq ], m0 - mova [dstq+2*lenq+mmsize], m1 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16_TO_FLT -INIT_XMM sse4 -CONV_S16_TO_FLT - -;------------------------------------------------------------------------------ -; void ff_conv_s32_to_s16(int16_t *dst, const int32_t *src, int len); -;------------------------------------------------------------------------------ - -%macro CONV_S32_TO_S16 0 -cglobal conv_s32_to_s16, 3,3,4, dst, src, len - lea lenq, [2*lend] - lea srcq, [srcq+2*lenq] - add dstq, lenq - neg lenq -.loop: - mova m0, [srcq+2*lenq ] - mova m1, [srcq+2*lenq+ mmsize] - mova m2, [srcq+2*lenq+2*mmsize] - mova m3, [srcq+2*lenq+3*mmsize] - psrad m0, 16 - psrad m1, 16 - psrad m2, 16 - psrad m3, 16 - packssdw m0, m1 - packssdw m2, m3 - mova [dstq+lenq ], m0 - mova [dstq+lenq+mmsize], m2 - add lenq, mmsize*2 - jl .loop -%if mmsize == 8 - emms - RET -%else - REP_RET -%endif -%endmacro - -INIT_MMX mmx -CONV_S32_TO_S16 -INIT_XMM sse2 -CONV_S32_TO_S16 - -;------------------------------------------------------------------------------ -; void ff_conv_s32_to_flt(float *dst, const int32_t *src, int len); -;------------------------------------------------------------------------------ - -%macro CONV_S32_TO_FLT 0 -cglobal conv_s32_to_flt, 3,3,3, dst, src, len - lea lenq, [4*lend] - add srcq, lenq - add dstq, lenq - neg lenq - mova m0, [pf_s32_inv_scale] - ALIGN 16 -.loop: - cvtdq2ps m1, [srcq+lenq ] - cvtdq2ps m2, [srcq+lenq+mmsize] - mulps m1, m1, m0 - mulps m2, m2, m0 - mova [dstq+lenq ], m1 - mova [dstq+lenq+mmsize], m2 - add lenq, mmsize*2 - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S32_TO_FLT -%if HAVE_AVX_EXTERNAL -INIT_YMM avx -CONV_S32_TO_FLT -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_flt_to_s16(int16_t *dst, const float *src, int len); -;------------------------------------------------------------------------------ - -INIT_XMM sse2 -cglobal conv_flt_to_s16, 3,3,5, dst, src, len - lea lenq, [2*lend] - lea srcq, [srcq+2*lenq] - add dstq, lenq - neg lenq - mova m4, [pf_s16_scale] -.loop: - mova m0, [srcq+2*lenq ] - mova m1, [srcq+2*lenq+1*mmsize] - mova m2, [srcq+2*lenq+2*mmsize] - mova m3, [srcq+2*lenq+3*mmsize] - mulps m0, m4 - mulps m1, m4 - mulps m2, m4 - mulps m3, m4 - cvtps2dq m0, m0 - cvtps2dq m1, m1 - cvtps2dq m2, m2 - cvtps2dq m3, m3 - packssdw m0, m1 - packssdw m2, m3 - mova [dstq+lenq ], m0 - mova [dstq+lenq+mmsize], m2 - add lenq, mmsize*2 - jl .loop - REP_RET - -;------------------------------------------------------------------------------ -; void ff_conv_flt_to_s32(int32_t *dst, const float *src, int len); -;------------------------------------------------------------------------------ - -%macro CONV_FLT_TO_S32 0 -cglobal conv_flt_to_s32, 3,3,6, dst, src, len - lea lenq, [lend*4] - add srcq, lenq - add dstq, lenq - neg lenq - mova m4, [pf_s32_scale] - mova m5, [pf_s32_clip] -.loop: - mulps m0, m4, [srcq+lenq ] - mulps m1, m4, [srcq+lenq+1*mmsize] - mulps m2, m4, [srcq+lenq+2*mmsize] - mulps m3, m4, [srcq+lenq+3*mmsize] - minps m0, m0, m5 - minps m1, m1, m5 - minps m2, m2, m5 - minps m3, m3, m5 - cvtps2dq m0, m0 - cvtps2dq m1, m1 - cvtps2dq m2, m2 - cvtps2dq m3, m3 - mova [dstq+lenq ], m0 - mova [dstq+lenq+1*mmsize], m1 - mova [dstq+lenq+2*mmsize], m2 - mova [dstq+lenq+3*mmsize], m3 - add lenq, mmsize*4 - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_FLT_TO_S32 -%if HAVE_AVX_EXTERNAL -INIT_YMM avx -CONV_FLT_TO_S32 -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16p_to_s16_2ch(int16_t *dst, int16_t *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16P_TO_S16_2CH 0 -cglobal conv_s16p_to_s16_2ch, 3,4,5, dst, src0, len, src1 - mov src1q, [src0q+gprsize] - mov src0q, [src0q ] - lea lenq, [2*lend] - add src0q, lenq - add src1q, lenq - lea dstq, [dstq+2*lenq] - neg lenq -.loop: - mova m0, [src0q+lenq ] - mova m1, [src1q+lenq ] - mova m2, [src0q+lenq+mmsize] - mova m3, [src1q+lenq+mmsize] - SBUTTERFLY2 wd, 0, 1, 4 - SBUTTERFLY2 wd, 2, 3, 4 - mova [dstq+2*lenq+0*mmsize], m0 - mova [dstq+2*lenq+1*mmsize], m1 - mova [dstq+2*lenq+2*mmsize], m2 - mova [dstq+2*lenq+3*mmsize], m3 - add lenq, 2*mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16P_TO_S16_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16P_TO_S16_2CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16p_to_s16_6ch(int16_t *dst, int16_t *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -;------------------------------------------------------------------------------ -; NOTE: In the 6-channel functions, len could be used as an index on x86-64 -; instead of just a counter, which would avoid incrementing the -; pointers, but the extra complexity and amount of code is not worth -; the small gain. On x86-32 there are not enough registers to use len -; as an index without keeping two of the pointers on the stack and -; loading them in each iteration. -;------------------------------------------------------------------------------ - -%macro CONV_S16P_TO_S16_6CH 0 -%if ARCH_X86_64 -cglobal conv_s16p_to_s16_6ch, 3,8,7, dst, src0, len, src1, src2, src3, src4, src5 -%else -cglobal conv_s16p_to_s16_6ch, 2,7,7, dst, src0, src1, src2, src3, src4, src5 -%define lend dword r2m -%endif - mov src1q, [src0q+1*gprsize] - mov src2q, [src0q+2*gprsize] - mov src3q, [src0q+3*gprsize] - mov src4q, [src0q+4*gprsize] - mov src5q, [src0q+5*gprsize] - mov src0q, [src0q] - sub src1q, src0q - sub src2q, src0q - sub src3q, src0q - sub src4q, src0q - sub src5q, src0q -.loop: -%if cpuflag(sse2slow) - movq m0, [src0q ] ; m0 = 0, 6, 12, 18, x, x, x, x - movq m1, [src0q+src1q] ; m1 = 1, 7, 13, 19, x, x, x, x - movq m2, [src0q+src2q] ; m2 = 2, 8, 14, 20, x, x, x, x - movq m3, [src0q+src3q] ; m3 = 3, 9, 15, 21, x, x, x, x - movq m4, [src0q+src4q] ; m4 = 4, 10, 16, 22, x, x, x, x - movq m5, [src0q+src5q] ; m5 = 5, 11, 17, 23, x, x, x, x - ; unpack words: - punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19 - punpcklwd m2, m3 ; m2 = 4, 5, 10, 11, 16, 17, 22, 23 - punpcklwd m4, m5 ; m4 = 2, 3, 8, 9, 14, 15, 20, 21 - ; blend dwords - shufps m1, m0, m2, q2020 ; m1 = 0, 1, 12, 13, 2, 3, 14, 15 - shufps m0, m4, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17 - shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23 - ; shuffle dwords - pshufd m0, m0, q1302 ; m0 = 4, 5, 6, 7, 16, 17, 18, 19 - pshufd m1, m1, q3120 ; m1 = 0, 1, 2, 3, 12, 13, 14, 15 - pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23 - movq [dstq+0*mmsize/2], m1 - movq [dstq+1*mmsize/2], m0 - movq [dstq+2*mmsize/2], m2 - movhps [dstq+3*mmsize/2], m1 - movhps [dstq+4*mmsize/2], m0 - movhps [dstq+5*mmsize/2], m2 - add src0q, mmsize/2 - add dstq, mmsize*3 - sub lend, mmsize/4 -%else - mova m0, [src0q ] ; m0 = 0, 6, 12, 18, 24, 30, 36, 42 - mova m1, [src0q+src1q] ; m1 = 1, 7, 13, 19, 25, 31, 37, 43 - mova m2, [src0q+src2q] ; m2 = 2, 8, 14, 20, 26, 32, 38, 44 - mova m3, [src0q+src3q] ; m3 = 3, 9, 15, 21, 27, 33, 39, 45 - mova m4, [src0q+src4q] ; m4 = 4, 10, 16, 22, 28, 34, 40, 46 - mova m5, [src0q+src5q] ; m5 = 5, 11, 17, 23, 29, 35, 41, 47 - ; unpack words: - SBUTTERFLY2 wd, 0, 1, 6 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19 - ; m1 = 24, 25, 30, 31, 36, 37, 42, 43 - SBUTTERFLY2 wd, 2, 3, 6 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21 - ; m3 = 26, 27, 32, 33, 38, 39, 44, 45 - SBUTTERFLY2 wd, 4, 5, 6 ; m4 = 4, 5, 10, 11, 16, 17, 22, 23 - ; m5 = 28, 29, 34, 35, 40, 41, 46, 47 - ; blend dwords - shufps m6, m0, m2, q2020 ; m6 = 0, 1, 12, 13, 2, 3, 14, 15 - shufps m0, m4, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17 - shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23 - SWAP 4,6 ; m4 = 0, 1, 12, 13, 2, 3, 14, 15 - shufps m6, m1, m3, q2020 ; m6 = 24, 25, 36, 37, 26, 27, 38, 39 - shufps m1, m5, q2031 ; m1 = 30, 31, 42, 43, 28, 29, 40, 41 - shufps m3, m5, q3131 ; m3 = 32, 33, 44, 45, 34, 35, 46, 47 - SWAP 5,6 ; m5 = 24, 25, 36, 37, 26, 27, 38, 39 - ; shuffle dwords - pshufd m0, m0, q1302 ; m0 = 4, 5, 6, 7, 16, 17, 18, 19 - pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23 - pshufd m4, m4, q3120 ; m4 = 0, 1, 2, 3, 12, 13, 14, 15 - pshufd m1, m1, q1302 ; m1 = 28, 29, 30, 31, 40, 41, 42, 43 - pshufd m3, m3, q3120 ; m3 = 32, 33, 34, 35, 44, 45, 46, 47 - pshufd m5, m5, q3120 ; m5 = 24, 25, 26, 27, 36, 37, 38, 39 - ; shuffle qwords - punpcklqdq m6, m4, m0 ; m6 = 0, 1, 2, 3, 4, 5, 6, 7 - punpckhqdq m0, m2 ; m0 = 16, 17, 18, 19, 20, 21, 22, 23 - shufps m2, m4, q3210 ; m2 = 8, 9, 10, 11, 12, 13, 14, 15 - SWAP 4,6 ; m4 = 0, 1, 2, 3, 4, 5, 6, 7 - punpcklqdq m6, m5, m1 ; m6 = 24, 25, 26, 27, 28, 29, 30, 31 - punpckhqdq m1, m3 ; m1 = 40, 41, 42, 43, 44, 45, 46, 47 - shufps m3, m5, q3210 ; m3 = 32, 33, 34, 35, 36, 37, 38, 39 - SWAP 5,6 ; m5 = 24, 25, 26, 27, 28, 29, 30, 31 - mova [dstq+0*mmsize], m4 - mova [dstq+1*mmsize], m2 - mova [dstq+2*mmsize], m0 - mova [dstq+3*mmsize], m5 - mova [dstq+4*mmsize], m3 - mova [dstq+5*mmsize], m1 - add src0q, mmsize - add dstq, mmsize*6 - sub lend, mmsize/2 -%endif - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16P_TO_S16_6CH -INIT_XMM sse2slow -CONV_S16P_TO_S16_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16P_TO_S16_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16p_to_flt_2ch(float *dst, int16_t *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16P_TO_FLT_2CH 0 -cglobal conv_s16p_to_flt_2ch, 3,4,6, dst, src0, len, src1 - lea lenq, [2*lend] - mov src1q, [src0q+gprsize] - mov src0q, [src0q ] - lea dstq, [dstq+4*lenq] - add src0q, lenq - add src1q, lenq - neg lenq - mova m5, [pf_s32_inv_scale] -.loop: - mova m2, [src0q+lenq] ; m2 = 0, 2, 4, 6, 8, 10, 12, 14 - mova m4, [src1q+lenq] ; m4 = 1, 3, 5, 7, 9, 11, 13, 15 - SBUTTERFLY2 wd, 2, 4, 3 ; m2 = 0, 1, 2, 3, 4, 5, 6, 7 - ; m4 = 8, 9, 10, 11, 12, 13, 14, 15 - pxor m3, m3 - punpcklwd m0, m3, m2 ; m0 = 0, 1, 2, 3 - punpckhwd m1, m3, m2 ; m1 = 4, 5, 6, 7 - punpcklwd m2, m3, m4 ; m2 = 8, 9, 10, 11 - punpckhwd m3, m4 ; m3 = 12, 13, 14, 15 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - cvtdq2ps m2, m2 - cvtdq2ps m3, m3 - mulps m0, m5 - mulps m1, m5 - mulps m2, m5 - mulps m3, m5 - mova [dstq+4*lenq ], m0 - mova [dstq+4*lenq+ mmsize], m1 - mova [dstq+4*lenq+2*mmsize], m2 - mova [dstq+4*lenq+3*mmsize], m3 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16P_TO_FLT_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16P_TO_FLT_2CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16p_to_flt_6ch(float *dst, int16_t *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16P_TO_FLT_6CH 0 -%if ARCH_X86_64 -cglobal conv_s16p_to_flt_6ch, 3,8,8, dst, src, len, src1, src2, src3, src4, src5 -%else -cglobal conv_s16p_to_flt_6ch, 2,7,8, dst, src, src1, src2, src3, src4, src5 -%define lend dword r2m -%endif - mov src1q, [srcq+1*gprsize] - mov src2q, [srcq+2*gprsize] - mov src3q, [srcq+3*gprsize] - mov src4q, [srcq+4*gprsize] - mov src5q, [srcq+5*gprsize] - mov srcq, [srcq] - sub src1q, srcq - sub src2q, srcq - sub src3q, srcq - sub src4q, srcq - sub src5q, srcq - mova m7, [pf_s32_inv_scale] -%if cpuflag(ssse3) - %define unpack_even m6 - mova m6, [pb_shuf_unpack_even] -%if ARCH_X86_64 - %define unpack_odd m8 - mova m8, [pb_shuf_unpack_odd] -%else - %define unpack_odd [pb_shuf_unpack_odd] -%endif -%endif -.loop: - movq m0, [srcq ] ; m0 = 0, 6, 12, 18, x, x, x, x - movq m1, [srcq+src1q] ; m1 = 1, 7, 13, 19, x, x, x, x - movq m2, [srcq+src2q] ; m2 = 2, 8, 14, 20, x, x, x, x - movq m3, [srcq+src3q] ; m3 = 3, 9, 15, 21, x, x, x, x - movq m4, [srcq+src4q] ; m4 = 4, 10, 16, 22, x, x, x, x - movq m5, [srcq+src5q] ; m5 = 5, 11, 17, 23, x, x, x, x - ; unpack words: - punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19 - punpcklwd m2, m3 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21 - punpcklwd m4, m5 ; m4 = 4, 5, 10, 11, 16, 17, 22, 23 - ; blend dwords - shufps m1, m4, m0, q3120 ; m1 = 4, 5, 16, 17, 6, 7, 18, 19 - shufps m0, m2, q2020 ; m0 = 0, 1, 12, 13, 2, 3, 14, 15 - shufps m2, m4, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23 -%if cpuflag(ssse3) - pshufb m3, m0, unpack_odd ; m3 = 12, 13, 14, 15 - pshufb m0, unpack_even ; m0 = 0, 1, 2, 3 - pshufb m4, m1, unpack_odd ; m4 = 16, 17, 18, 19 - pshufb m1, unpack_even ; m1 = 4, 5, 6, 7 - pshufb m5, m2, unpack_odd ; m5 = 20, 21, 22, 23 - pshufb m2, unpack_even ; m2 = 8, 9, 10, 11 -%else - ; shuffle dwords - pshufd m0, m0, q3120 ; m0 = 0, 1, 2, 3, 12, 13, 14, 15 - pshufd m1, m1, q3120 ; m1 = 4, 5, 6, 7, 16, 17, 18, 19 - pshufd m2, m2, q3120 ; m2 = 8, 9, 10, 11, 20, 21, 22, 23 - pxor m6, m6 ; convert s16 in m0-m2 to s32 in m0-m5 - punpcklwd m3, m6, m0 ; m3 = 0, 1, 2, 3 - punpckhwd m4, m6, m0 ; m4 = 12, 13, 14, 15 - punpcklwd m0, m6, m1 ; m0 = 4, 5, 6, 7 - punpckhwd m5, m6, m1 ; m5 = 16, 17, 18, 19 - punpcklwd m1, m6, m2 ; m1 = 8, 9, 10, 11 - punpckhwd m6, m2 ; m6 = 20, 21, 22, 23 - SWAP 6,2,1,0,3,4,5 ; swap registers 3,0,1,4,5,6 to 0,1,2,3,4,5 -%endif - cvtdq2ps m0, m0 ; convert s32 to float - cvtdq2ps m1, m1 - cvtdq2ps m2, m2 - cvtdq2ps m3, m3 - cvtdq2ps m4, m4 - cvtdq2ps m5, m5 - mulps m0, m7 ; scale float from s32 range to [-1.0,1.0] - mulps m1, m7 - mulps m2, m7 - mulps m3, m7 - mulps m4, m7 - mulps m5, m7 - mova [dstq ], m0 - mova [dstq+ mmsize], m1 - mova [dstq+2*mmsize], m2 - mova [dstq+3*mmsize], m3 - mova [dstq+4*mmsize], m4 - mova [dstq+5*mmsize], m5 - add srcq, mmsize/2 - add dstq, mmsize*6 - sub lend, mmsize/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16P_TO_FLT_6CH -INIT_XMM ssse3 -CONV_S16P_TO_FLT_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16P_TO_FLT_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_fltp_to_s16_2ch(int16_t *dst, float *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLTP_TO_S16_2CH 0 -cglobal conv_fltp_to_s16_2ch, 3,4,3, dst, src0, len, src1 - lea lenq, [4*lend] - mov src1q, [src0q+gprsize] - mov src0q, [src0q ] - add dstq, lenq - add src0q, lenq - add src1q, lenq - neg lenq - mova m2, [pf_s16_scale] -%if cpuflag(ssse3) - mova m3, [pb_interleave_words] -%endif -.loop: - mulps m0, m2, [src0q+lenq] ; m0 = 0, 2, 4, 6 - mulps m1, m2, [src1q+lenq] ; m1 = 1, 3, 5, 7 - cvtps2dq m0, m0 - cvtps2dq m1, m1 -%if cpuflag(ssse3) - packssdw m0, m1 ; m0 = 0, 2, 4, 6, 1, 3, 5, 7 - pshufb m0, m3 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7 -%else - packssdw m0, m0 ; m0 = 0, 2, 4, 6, x, x, x, x - packssdw m1, m1 ; m1 = 1, 3, 5, 7, x, x, x, x - punpcklwd m0, m1 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7 -%endif - mova [dstq+lenq], m0 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_FLTP_TO_S16_2CH -INIT_XMM ssse3 -CONV_FLTP_TO_S16_2CH - -;------------------------------------------------------------------------------ -; void ff_conv_fltp_to_s16_6ch(int16_t *dst, float *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLTP_TO_S16_6CH 0 -%if ARCH_X86_64 -cglobal conv_fltp_to_s16_6ch, 3,8,7, dst, src, len, src1, src2, src3, src4, src5 -%else -cglobal conv_fltp_to_s16_6ch, 2,7,7, dst, src, src1, src2, src3, src4, src5 -%define lend dword r2m -%endif - mov src1q, [srcq+1*gprsize] - mov src2q, [srcq+2*gprsize] - mov src3q, [srcq+3*gprsize] - mov src4q, [srcq+4*gprsize] - mov src5q, [srcq+5*gprsize] - mov srcq, [srcq] - sub src1q, srcq - sub src2q, srcq - sub src3q, srcq - sub src4q, srcq - sub src5q, srcq - movaps xmm6, [pf_s16_scale] -.loop: -%if cpuflag(sse2) - mulps m0, m6, [srcq ] - mulps m1, m6, [srcq+src1q] - mulps m2, m6, [srcq+src2q] - mulps m3, m6, [srcq+src3q] - mulps m4, m6, [srcq+src4q] - mulps m5, m6, [srcq+src5q] - cvtps2dq m0, m0 - cvtps2dq m1, m1 - cvtps2dq m2, m2 - cvtps2dq m3, m3 - cvtps2dq m4, m4 - cvtps2dq m5, m5 - packssdw m0, m3 ; m0 = 0, 6, 12, 18, 3, 9, 15, 21 - packssdw m1, m4 ; m1 = 1, 7, 13, 19, 4, 10, 16, 22 - packssdw m2, m5 ; m2 = 2, 8, 14, 20, 5, 11, 17, 23 - ; unpack words: - movhlps m3, m0 ; m3 = 3, 9, 15, 21, x, x, x, x - punpcklwd m0, m1 ; m0 = 0, 1, 6, 7, 12, 13, 18, 19 - punpckhwd m1, m2 ; m1 = 4, 5, 10, 11, 16, 17, 22, 23 - punpcklwd m2, m3 ; m2 = 2, 3, 8, 9, 14, 15, 20, 21 - ; blend dwords: - shufps m3, m0, m2, q2020 ; m3 = 0, 1, 12, 13, 2, 3, 14, 15 - shufps m0, m1, q2031 ; m0 = 6, 7, 18, 19, 4, 5, 16, 17 - shufps m2, m1, q3131 ; m2 = 8, 9, 20, 21, 10, 11, 22, 23 - ; shuffle dwords: - shufps m1, m2, m3, q3120 ; m1 = 8, 9, 10, 11, 12, 13, 14, 15 - shufps m3, m0, q0220 ; m3 = 0, 1, 2, 3, 4, 5, 6, 7 - shufps m0, m2, q3113 ; m0 = 16, 17, 18, 19, 20, 21, 22, 23 - mova [dstq+0*mmsize], m3 - mova [dstq+1*mmsize], m1 - mova [dstq+2*mmsize], m0 -%else ; sse - movlps xmm0, [srcq ] - movlps xmm1, [srcq+src1q] - movlps xmm2, [srcq+src2q] - movlps xmm3, [srcq+src3q] - movlps xmm4, [srcq+src4q] - movlps xmm5, [srcq+src5q] - mulps xmm0, xmm6 - mulps xmm1, xmm6 - mulps xmm2, xmm6 - mulps xmm3, xmm6 - mulps xmm4, xmm6 - mulps xmm5, xmm6 - cvtps2pi mm0, xmm0 - cvtps2pi mm1, xmm1 - cvtps2pi mm2, xmm2 - cvtps2pi mm3, xmm3 - cvtps2pi mm4, xmm4 - cvtps2pi mm5, xmm5 - packssdw mm0, mm3 ; m0 = 0, 6, 3, 9 - packssdw mm1, mm4 ; m1 = 1, 7, 4, 10 - packssdw mm2, mm5 ; m2 = 2, 8, 5, 11 - ; unpack words - pshufw mm3, mm0, q1032 ; m3 = 3, 9, 0, 6 - punpcklwd mm0, mm1 ; m0 = 0, 1, 6, 7 - punpckhwd mm1, mm2 ; m1 = 4, 5, 10, 11 - punpcklwd mm2, mm3 ; m2 = 2, 3, 8, 9 - ; unpack dwords - pshufw mm3, mm0, q1032 ; m3 = 6, 7, 0, 1 - punpckldq mm0, mm2 ; m0 = 0, 1, 2, 3 (final) - punpckhdq mm2, mm1 ; m2 = 8, 9, 10, 11 (final) - punpckldq mm1, mm3 ; m1 = 4, 5, 6, 7 (final) - mova [dstq+0*mmsize], mm0 - mova [dstq+1*mmsize], mm1 - mova [dstq+2*mmsize], mm2 -%endif - add srcq, mmsize - add dstq, mmsize*3 - sub lend, mmsize/4 - jg .loop -%if mmsize == 8 - emms - RET -%else - REP_RET -%endif -%endmacro - -INIT_MMX sse -CONV_FLTP_TO_S16_6CH -INIT_XMM sse2 -CONV_FLTP_TO_S16_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLTP_TO_S16_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_fltp_to_flt_2ch(float *dst, float *const *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLTP_TO_FLT_2CH 0 -cglobal conv_fltp_to_flt_2ch, 3,4,5, dst, src0, len, src1 - mov src1q, [src0q+gprsize] - mov src0q, [src0q] - lea lenq, [4*lend] - add src0q, lenq - add src1q, lenq - lea dstq, [dstq+2*lenq] - neg lenq -.loop: - mova m0, [src0q+lenq ] - mova m1, [src1q+lenq ] - mova m2, [src0q+lenq+mmsize] - mova m3, [src1q+lenq+mmsize] - SBUTTERFLYPS 0, 1, 4 - SBUTTERFLYPS 2, 3, 4 - mova [dstq+2*lenq+0*mmsize], m0 - mova [dstq+2*lenq+1*mmsize], m1 - mova [dstq+2*lenq+2*mmsize], m2 - mova [dstq+2*lenq+3*mmsize], m3 - add lenq, 2*mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse -CONV_FLTP_TO_FLT_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLTP_TO_FLT_2CH -%endif - -;----------------------------------------------------------------------------- -; void ff_conv_fltp_to_flt_6ch(float *dst, float *const *src, int len, -; int channels); -;----------------------------------------------------------------------------- - -%macro CONV_FLTP_TO_FLT_6CH 0 -cglobal conv_fltp_to_flt_6ch, 2,8,7, dst, src, src1, src2, src3, src4, src5, len -%if ARCH_X86_64 - mov lend, r2d -%else - %define lend dword r2m -%endif - mov src1q, [srcq+1*gprsize] - mov src2q, [srcq+2*gprsize] - mov src3q, [srcq+3*gprsize] - mov src4q, [srcq+4*gprsize] - mov src5q, [srcq+5*gprsize] - mov srcq, [srcq] - sub src1q, srcq - sub src2q, srcq - sub src3q, srcq - sub src4q, srcq - sub src5q, srcq -.loop: - mova m0, [srcq ] - mova m1, [srcq+src1q] - mova m2, [srcq+src2q] - mova m3, [srcq+src3q] - mova m4, [srcq+src4q] - mova m5, [srcq+src5q] -%if cpuflag(sse4) - SBUTTERFLYPS 0, 1, 6 - SBUTTERFLYPS 2, 3, 6 - SBUTTERFLYPS 4, 5, 6 - - blendps m6, m4, m0, 1100b - movlhps m0, m2 - movhlps m4, m2 - blendps m2, m5, m1, 1100b - movlhps m1, m3 - movhlps m5, m3 - - movaps [dstq ], m0 - movaps [dstq+16], m6 - movaps [dstq+32], m4 - movaps [dstq+48], m1 - movaps [dstq+64], m2 - movaps [dstq+80], m5 -%else ; mmx - SBUTTERFLY dq, 0, 1, 6 - SBUTTERFLY dq, 2, 3, 6 - SBUTTERFLY dq, 4, 5, 6 - - movq [dstq ], m0 - movq [dstq+ 8], m2 - movq [dstq+16], m4 - movq [dstq+24], m1 - movq [dstq+32], m3 - movq [dstq+40], m5 -%endif - add srcq, mmsize - add dstq, mmsize*6 - sub lend, mmsize/4 - jg .loop -%if mmsize == 8 - emms - RET -%else - REP_RET -%endif -%endmacro - -INIT_MMX mmx -CONV_FLTP_TO_FLT_6CH -INIT_XMM sse4 -CONV_FLTP_TO_FLT_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLTP_TO_FLT_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16_to_s16p_2ch(int16_t *const *dst, int16_t *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16_TO_S16P_2CH 0 -cglobal conv_s16_to_s16p_2ch, 3,4,4, dst0, src, len, dst1 - lea lenq, [2*lend] - mov dst1q, [dst0q+gprsize] - mov dst0q, [dst0q ] - lea srcq, [srcq+2*lenq] - add dst0q, lenq - add dst1q, lenq - neg lenq -%if cpuflag(ssse3) - mova m3, [pb_deinterleave_words] -%endif -.loop: - mova m0, [srcq+2*lenq ] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7 - mova m1, [srcq+2*lenq+mmsize] ; m1 = 8, 9, 10, 11, 12, 13, 14, 15 -%if cpuflag(ssse3) - pshufb m0, m3 ; m0 = 0, 2, 4, 6, 1, 3, 5, 7 - pshufb m1, m3 ; m1 = 8, 10, 12, 14, 9, 11, 13, 15 - SBUTTERFLY2 qdq, 0, 1, 2 ; m0 = 0, 2, 4, 6, 8, 10, 12, 14 - ; m1 = 1, 3, 5, 7, 9, 11, 13, 15 -%else ; sse2 - pshuflw m0, m0, q3120 ; m0 = 0, 2, 1, 3, 4, 5, 6, 7 - pshufhw m0, m0, q3120 ; m0 = 0, 2, 1, 3, 4, 6, 5, 7 - pshuflw m1, m1, q3120 ; m1 = 8, 10, 9, 11, 12, 13, 14, 15 - pshufhw m1, m1, q3120 ; m1 = 8, 10, 9, 11, 12, 14, 13, 15 - DEINT2_PS 0, 1, 2 ; m0 = 0, 2, 4, 6, 8, 10, 12, 14 - ; m1 = 1, 3, 5, 7, 9, 11, 13, 15 -%endif - mova [dst0q+lenq], m0 - mova [dst1q+lenq], m1 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16_TO_S16P_2CH -INIT_XMM ssse3 -CONV_S16_TO_S16P_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16_TO_S16P_2CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16_to_s16p_6ch(int16_t *const *dst, int16_t *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16_TO_S16P_6CH 0 -%if ARCH_X86_64 -cglobal conv_s16_to_s16p_6ch, 3,8,5, dst, src, len, dst1, dst2, dst3, dst4, dst5 -%else -cglobal conv_s16_to_s16p_6ch, 2,7,5, dst, src, dst1, dst2, dst3, dst4, dst5 -%define lend dword r2m -%endif - mov dst1q, [dstq+ gprsize] - mov dst2q, [dstq+2*gprsize] - mov dst3q, [dstq+3*gprsize] - mov dst4q, [dstq+4*gprsize] - mov dst5q, [dstq+5*gprsize] - mov dstq, [dstq ] - sub dst1q, dstq - sub dst2q, dstq - sub dst3q, dstq - sub dst4q, dstq - sub dst5q, dstq -.loop: - mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7 - mova m3, [srcq+1*mmsize] ; m3 = 8, 9, 10, 11, 12, 13, 14, 15 - mova m2, [srcq+2*mmsize] ; m2 = 16, 17, 18, 19, 20, 21, 22, 23 - PALIGNR m1, m3, m0, 12, m4 ; m1 = 6, 7, 8, 9, 10, 11, x, x - shufps m3, m2, q1032 ; m3 = 12, 13, 14, 15, 16, 17, 18, 19 - psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x - SBUTTERFLY2 wd, 0, 1, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9 - ; m1 = 4, 10, 5, 11, x, x, x, x - SBUTTERFLY2 wd, 3, 2, 4 ; m3 = 12, 18, 13, 19, 14, 20, 15, 21 - ; m2 = 16, 22, 17, 23, x, x, x, x - SBUTTERFLY2 dq, 0, 3, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19 - ; m3 = 2, 8, 14, 20, 3, 9, 15, 21 - punpckldq m1, m2 ; m1 = 4, 10, 16, 22, 5, 11, 17, 23 - movq [dstq ], m0 - movhps [dstq+dst1q], m0 - movq [dstq+dst2q], m3 - movhps [dstq+dst3q], m3 - movq [dstq+dst4q], m1 - movhps [dstq+dst5q], m1 - add srcq, mmsize*3 - add dstq, mmsize/2 - sub lend, mmsize/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16_TO_S16P_6CH -INIT_XMM ssse3 -CONV_S16_TO_S16P_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16_TO_S16P_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16_to_fltp_2ch(float *const *dst, int16_t *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16_TO_FLTP_2CH 0 -cglobal conv_s16_to_fltp_2ch, 3,4,5, dst0, src, len, dst1 - lea lenq, [4*lend] - mov dst1q, [dst0q+gprsize] - mov dst0q, [dst0q ] - add srcq, lenq - add dst0q, lenq - add dst1q, lenq - neg lenq - mova m3, [pf_s32_inv_scale] - mova m4, [pw_zero_even] -.loop: - mova m1, [srcq+lenq] - pslld m0, m1, 16 - pand m1, m4 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - mulps m0, m0, m3 - mulps m1, m1, m3 - mova [dst0q+lenq], m0 - mova [dst1q+lenq], m1 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16_TO_FLTP_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16_TO_FLTP_2CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_s16_to_fltp_6ch(float *const *dst, int16_t *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_S16_TO_FLTP_6CH 0 -%if ARCH_X86_64 -cglobal conv_s16_to_fltp_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5 -%else -cglobal conv_s16_to_fltp_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5 -%define lend dword r2m -%endif - mov dst1q, [dstq+ gprsize] - mov dst2q, [dstq+2*gprsize] - mov dst3q, [dstq+3*gprsize] - mov dst4q, [dstq+4*gprsize] - mov dst5q, [dstq+5*gprsize] - mov dstq, [dstq ] - sub dst1q, dstq - sub dst2q, dstq - sub dst3q, dstq - sub dst4q, dstq - sub dst5q, dstq - mova m6, [pf_s16_inv_scale] -.loop: - mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3, 4, 5, 6, 7 - mova m3, [srcq+1*mmsize] ; m3 = 8, 9, 10, 11, 12, 13, 14, 15 - mova m2, [srcq+2*mmsize] ; m2 = 16, 17, 18, 19, 20, 21, 22, 23 - PALIGNR m1, m3, m0, 12, m4 ; m1 = 6, 7, 8, 9, 10, 11, x, x - shufps m3, m2, q1032 ; m3 = 12, 13, 14, 15, 16, 17, 18, 19 - psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x - SBUTTERFLY2 wd, 0, 1, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9 - ; m1 = 4, 10, 5, 11, x, x, x, x - SBUTTERFLY2 wd, 3, 2, 4 ; m3 = 12, 18, 13, 19, 14, 20, 15, 21 - ; m2 = 16, 22, 17, 23, x, x, x, x - SBUTTERFLY2 dq, 0, 3, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19 - ; m3 = 2, 8, 14, 20, 3, 9, 15, 21 - punpckldq m1, m2 ; m1 = 4, 10, 16, 22, 5, 11, 17, 23 - S16_TO_S32_SX 0, 2 ; m0 = 0, 6, 12, 18 - ; m2 = 1, 7, 13, 19 - S16_TO_S32_SX 3, 4 ; m3 = 2, 8, 14, 20 - ; m4 = 3, 9, 15, 21 - S16_TO_S32_SX 1, 5 ; m1 = 4, 10, 16, 22 - ; m5 = 5, 11, 17, 23 - SWAP 1,2,3,4 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - cvtdq2ps m2, m2 - cvtdq2ps m3, m3 - cvtdq2ps m4, m4 - cvtdq2ps m5, m5 - mulps m0, m6 - mulps m1, m6 - mulps m2, m6 - mulps m3, m6 - mulps m4, m6 - mulps m5, m6 - mova [dstq ], m0 - mova [dstq+dst1q], m1 - mova [dstq+dst2q], m2 - mova [dstq+dst3q], m3 - mova [dstq+dst4q], m4 - mova [dstq+dst5q], m5 - add srcq, mmsize*3 - add dstq, mmsize - sub lend, mmsize/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_S16_TO_FLTP_6CH -INIT_XMM ssse3 -CONV_S16_TO_FLTP_6CH -INIT_XMM sse4 -CONV_S16_TO_FLTP_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_S16_TO_FLTP_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_flt_to_s16p_2ch(int16_t *const *dst, float *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLT_TO_S16P_2CH 0 -cglobal conv_flt_to_s16p_2ch, 3,4,6, dst0, src, len, dst1 - lea lenq, [2*lend] - mov dst1q, [dst0q+gprsize] - mov dst0q, [dst0q ] - lea srcq, [srcq+4*lenq] - add dst0q, lenq - add dst1q, lenq - neg lenq - mova m5, [pf_s16_scale] -.loop: - mova m0, [srcq+4*lenq ] - mova m1, [srcq+4*lenq+ mmsize] - mova m2, [srcq+4*lenq+2*mmsize] - mova m3, [srcq+4*lenq+3*mmsize] - DEINT2_PS 0, 1, 4 - DEINT2_PS 2, 3, 4 - mulps m0, m0, m5 - mulps m1, m1, m5 - mulps m2, m2, m5 - mulps m3, m3, m5 - cvtps2dq m0, m0 - cvtps2dq m1, m1 - cvtps2dq m2, m2 - cvtps2dq m3, m3 - packssdw m0, m2 - packssdw m1, m3 - mova [dst0q+lenq], m0 - mova [dst1q+lenq], m1 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_FLT_TO_S16P_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLT_TO_S16P_2CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_flt_to_s16p_6ch(int16_t *const *dst, float *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLT_TO_S16P_6CH 0 -%if ARCH_X86_64 -cglobal conv_flt_to_s16p_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5 -%else -cglobal conv_flt_to_s16p_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5 -%define lend dword r2m -%endif - mov dst1q, [dstq+ gprsize] - mov dst2q, [dstq+2*gprsize] - mov dst3q, [dstq+3*gprsize] - mov dst4q, [dstq+4*gprsize] - mov dst5q, [dstq+5*gprsize] - mov dstq, [dstq ] - sub dst1q, dstq - sub dst2q, dstq - sub dst3q, dstq - sub dst4q, dstq - sub dst5q, dstq - mova m6, [pf_s16_scale] -.loop: - mulps m0, m6, [srcq+0*mmsize] - mulps m3, m6, [srcq+1*mmsize] - mulps m1, m6, [srcq+2*mmsize] - mulps m4, m6, [srcq+3*mmsize] - mulps m2, m6, [srcq+4*mmsize] - mulps m5, m6, [srcq+5*mmsize] - cvtps2dq m0, m0 - cvtps2dq m1, m1 - cvtps2dq m2, m2 - cvtps2dq m3, m3 - cvtps2dq m4, m4 - cvtps2dq m5, m5 - packssdw m0, m3 ; m0 = 0, 1, 2, 3, 4, 5, 6, 7 - packssdw m1, m4 ; m1 = 8, 9, 10, 11, 12, 13, 14, 15 - packssdw m2, m5 ; m2 = 16, 17, 18, 19, 20, 21, 22, 23 - PALIGNR m3, m1, m0, 12, m4 ; m3 = 6, 7, 8, 9, 10, 11, x, x - shufps m1, m2, q1032 ; m1 = 12, 13, 14, 15, 16, 17, 18, 19 - psrldq m2, 4 ; m2 = 18, 19, 20, 21, 22, 23, x, x - SBUTTERFLY2 wd, 0, 3, 4 ; m0 = 0, 6, 1, 7, 2, 8, 3, 9 - ; m3 = 4, 10, 5, 11, x, x, x, x - SBUTTERFLY2 wd, 1, 2, 4 ; m1 = 12, 18, 13, 19, 14, 20, 15, 21 - ; m2 = 16, 22, 17, 23, x, x, x, x - SBUTTERFLY2 dq, 0, 1, 4 ; m0 = 0, 6, 12, 18, 1, 7, 13, 19 - ; m1 = 2, 8, 14, 20, 3, 9, 15, 21 - punpckldq m3, m2 ; m3 = 4, 10, 16, 22, 5, 11, 17, 23 - movq [dstq ], m0 - movhps [dstq+dst1q], m0 - movq [dstq+dst2q], m1 - movhps [dstq+dst3q], m1 - movq [dstq+dst4q], m3 - movhps [dstq+dst5q], m3 - add srcq, mmsize*6 - add dstq, mmsize/2 - sub lend, mmsize/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_FLT_TO_S16P_6CH -INIT_XMM ssse3 -CONV_FLT_TO_S16P_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLT_TO_S16P_6CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_flt_to_fltp_2ch(float *const *dst, float *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLT_TO_FLTP_2CH 0 -cglobal conv_flt_to_fltp_2ch, 3,4,3, dst0, src, len, dst1 - lea lenq, [4*lend] - mov dst1q, [dst0q+gprsize] - mov dst0q, [dst0q ] - lea srcq, [srcq+2*lenq] - add dst0q, lenq - add dst1q, lenq - neg lenq -.loop: - mova m0, [srcq+2*lenq ] - mova m1, [srcq+2*lenq+mmsize] - DEINT2_PS 0, 1, 2 - mova [dst0q+lenq], m0 - mova [dst1q+lenq], m1 - add lenq, mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse -CONV_FLT_TO_FLTP_2CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLT_TO_FLTP_2CH -%endif - -;------------------------------------------------------------------------------ -; void ff_conv_flt_to_fltp_6ch(float *const *dst, float *src, int len, -; int channels); -;------------------------------------------------------------------------------ - -%macro CONV_FLT_TO_FLTP_6CH 0 -%if ARCH_X86_64 -cglobal conv_flt_to_fltp_6ch, 3,8,7, dst, src, len, dst1, dst2, dst3, dst4, dst5 -%else -cglobal conv_flt_to_fltp_6ch, 2,7,7, dst, src, dst1, dst2, dst3, dst4, dst5 -%define lend dword r2m -%endif - mov dst1q, [dstq+ gprsize] - mov dst2q, [dstq+2*gprsize] - mov dst3q, [dstq+3*gprsize] - mov dst4q, [dstq+4*gprsize] - mov dst5q, [dstq+5*gprsize] - mov dstq, [dstq ] - sub dst1q, dstq - sub dst2q, dstq - sub dst3q, dstq - sub dst4q, dstq - sub dst5q, dstq -.loop: - mova m0, [srcq+0*mmsize] ; m0 = 0, 1, 2, 3 - mova m1, [srcq+1*mmsize] ; m1 = 4, 5, 6, 7 - mova m2, [srcq+2*mmsize] ; m2 = 8, 9, 10, 11 - mova m3, [srcq+3*mmsize] ; m3 = 12, 13, 14, 15 - mova m4, [srcq+4*mmsize] ; m4 = 16, 17, 18, 19 - mova m5, [srcq+5*mmsize] ; m5 = 20, 21, 22, 23 - - SBUTTERFLY2 dq, 0, 3, 6 ; m0 = 0, 12, 1, 13 - ; m3 = 2, 14, 3, 15 - SBUTTERFLY2 dq, 1, 4, 6 ; m1 = 4, 16, 5, 17 - ; m4 = 6, 18, 7, 19 - SBUTTERFLY2 dq, 2, 5, 6 ; m2 = 8, 20, 9, 21 - ; m5 = 10, 22, 11, 23 - SBUTTERFLY2 dq, 0, 4, 6 ; m0 = 0, 6, 12, 18 - ; m4 = 1, 7, 13, 19 - SBUTTERFLY2 dq, 3, 2, 6 ; m3 = 2, 8, 14, 20 - ; m2 = 3, 9, 15, 21 - SBUTTERFLY2 dq, 1, 5, 6 ; m1 = 4, 10, 16, 22 - ; m5 = 5, 11, 17, 23 - mova [dstq ], m0 - mova [dstq+dst1q], m4 - mova [dstq+dst2q], m3 - mova [dstq+dst3q], m2 - mova [dstq+dst4q], m1 - mova [dstq+dst5q], m5 - add srcq, mmsize*6 - add dstq, mmsize - sub lend, mmsize/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -CONV_FLT_TO_FLTP_6CH -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -CONV_FLT_TO_FLTP_6CH -%endif diff --git a/libavresample/x86/audio_convert_init.c b/libavresample/x86/audio_convert_init.c deleted file mode 100644 index 0af4222bea..0000000000 --- a/libavresample/x86/audio_convert_init.c +++ /dev/null @@ -1,265 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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/cpu.h" -#include "libavutil/x86/cpu.h" -#include "libavresample/audio_convert.h" - -/* flat conversions */ - -void ff_conv_s16_to_s32_sse2(int16_t *dst, const int32_t *src, int len); - -void ff_conv_s16_to_flt_sse2(float *dst, const int16_t *src, int len); -void ff_conv_s16_to_flt_sse4(float *dst, const int16_t *src, int len); - -void ff_conv_s32_to_s16_mmx (int16_t *dst, const int32_t *src, int len); -void ff_conv_s32_to_s16_sse2(int16_t *dst, const int32_t *src, int len); - -void ff_conv_s32_to_flt_sse2(float *dst, const int32_t *src, int len); -void ff_conv_s32_to_flt_avx (float *dst, const int32_t *src, int len); - -void ff_conv_flt_to_s16_sse2(int16_t *dst, const float *src, int len); - -void ff_conv_flt_to_s32_sse2(int32_t *dst, const float *src, int len); -void ff_conv_flt_to_s32_avx (int32_t *dst, const float *src, int len); - -/* interleave conversions */ - -void ff_conv_s16p_to_s16_2ch_sse2(int16_t *dst, int16_t *const *src, - int len, int channels); -void ff_conv_s16p_to_s16_2ch_avx (int16_t *dst, int16_t *const *src, - int len, int channels); - -void ff_conv_s16p_to_s16_6ch_sse2(int16_t *dst, int16_t *const *src, - int len, int channels); -void ff_conv_s16p_to_s16_6ch_sse2slow(int16_t *dst, int16_t *const *src, - int len, int channels); -void ff_conv_s16p_to_s16_6ch_avx (int16_t *dst, int16_t *const *src, - int len, int channels); - -void ff_conv_s16p_to_flt_2ch_sse2(float *dst, int16_t *const *src, - int len, int channels); -void ff_conv_s16p_to_flt_2ch_avx (float *dst, int16_t *const *src, - int len, int channels); - -void ff_conv_s16p_to_flt_6ch_sse2 (float *dst, int16_t *const *src, - int len, int channels); -void ff_conv_s16p_to_flt_6ch_ssse3(float *dst, int16_t *const *src, - int len, int channels); -void ff_conv_s16p_to_flt_6ch_avx (float *dst, int16_t *const *src, - int len, int channels); - -void ff_conv_fltp_to_s16_2ch_sse2 (int16_t *dst, float *const *src, - int len, int channels); -void ff_conv_fltp_to_s16_2ch_ssse3(int16_t *dst, float *const *src, - int len, int channels); - -void ff_conv_fltp_to_s16_6ch_sse (int16_t *dst, float *const *src, - int len, int channels); -void ff_conv_fltp_to_s16_6ch_sse2(int16_t *dst, float *const *src, - int len, int channels); -void ff_conv_fltp_to_s16_6ch_avx (int16_t *dst, float *const *src, - int len, int channels); - -void ff_conv_fltp_to_flt_2ch_sse(float *dst, float *const *src, int len, - int channels); -void ff_conv_fltp_to_flt_2ch_avx(float *dst, float *const *src, int len, - int channels); - -void ff_conv_fltp_to_flt_6ch_mmx (float *dst, float *const *src, int len, - int channels); -void ff_conv_fltp_to_flt_6ch_sse4(float *dst, float *const *src, int len, - int channels); -void ff_conv_fltp_to_flt_6ch_avx (float *dst, float *const *src, int len, - int channels); - -/* deinterleave conversions */ - -void ff_conv_s16_to_s16p_2ch_sse2(int16_t *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_s16p_2ch_ssse3(int16_t *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_s16p_2ch_avx (int16_t *const *dst, int16_t *src, - int len, int channels); - -void ff_conv_s16_to_s16p_6ch_sse2 (int16_t *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_s16p_6ch_ssse3(int16_t *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_s16p_6ch_avx (int16_t *const *dst, int16_t *src, - int len, int channels); - -void ff_conv_s16_to_fltp_2ch_sse2(float *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_fltp_2ch_avx (float *const *dst, int16_t *src, - int len, int channels); - -void ff_conv_s16_to_fltp_6ch_sse2 (float *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_fltp_6ch_ssse3(float *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_fltp_6ch_sse4 (float *const *dst, int16_t *src, - int len, int channels); -void ff_conv_s16_to_fltp_6ch_avx (float *const *dst, int16_t *src, - int len, int channels); - -void ff_conv_flt_to_s16p_2ch_sse2(int16_t *const *dst, float *src, - int len, int channels); -void ff_conv_flt_to_s16p_2ch_avx (int16_t *const *dst, float *src, - int len, int channels); - -void ff_conv_flt_to_s16p_6ch_sse2 (int16_t *const *dst, float *src, - int len, int channels); -void ff_conv_flt_to_s16p_6ch_ssse3(int16_t *const *dst, float *src, - int len, int channels); -void ff_conv_flt_to_s16p_6ch_avx (int16_t *const *dst, float *src, - int len, int channels); - -void ff_conv_flt_to_fltp_2ch_sse(float *const *dst, float *src, int len, - int channels); -void ff_conv_flt_to_fltp_2ch_avx(float *const *dst, float *src, int len, - int channels); - -void ff_conv_flt_to_fltp_6ch_sse2(float *const *dst, float *src, int len, - int channels); -void ff_conv_flt_to_fltp_6ch_avx (float *const *dst, float *src, int len, - int channels); - -av_cold void ff_audio_convert_init_x86(AudioConvert *ac) -{ - int cpu_flags = av_get_cpu_flags(); - - if (EXTERNAL_MMX(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32, - 0, 1, 8, "MMX", ff_conv_s32_to_s16_mmx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, - 6, 1, 4, "MMX", ff_conv_fltp_to_flt_6ch_mmx); - } - if (EXTERNAL_SSE(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 6, 1, 2, "SSE", ff_conv_fltp_to_s16_6ch_sse); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, - 2, 16, 8, "SSE", ff_conv_fltp_to_flt_2ch_sse); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT, - 2, 16, 4, "SSE", ff_conv_flt_to_fltp_2ch_sse); - } - if (EXTERNAL_SSE2(cpu_flags)) { - if (!(cpu_flags & AV_CPU_FLAG_SSE2SLOW)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32, - 0, 16, 16, "SSE2", ff_conv_s32_to_s16_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, - 6, 16, 8, "SSE2", ff_conv_s16p_to_s16_6ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 6, 16, 4, "SSE2", ff_conv_fltp_to_s16_6ch_sse2); - } else { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, - 6, 1, 4, "SSE2SLOW", ff_conv_s16p_to_s16_6ch_sse2slow); - } - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16, - 0, 16, 8, "SSE2", ff_conv_s16_to_s32_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, - 0, 16, 8, "SSE2", ff_conv_s16_to_flt_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32, - 0, 16, 8, "SSE2", ff_conv_s32_to_flt_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT, - 0, 16, 16, "SSE2", ff_conv_flt_to_s16_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT, - 0, 16, 16, "SSE2", ff_conv_flt_to_s32_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, - 2, 16, 16, "SSE2", ff_conv_s16p_to_s16_2ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P, - 2, 16, 8, "SSE2", ff_conv_s16p_to_flt_2ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P, - 6, 16, 4, "SSE2", ff_conv_s16p_to_flt_6ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 2, 16, 4, "SSE2", ff_conv_fltp_to_s16_2ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, - 2, 16, 8, "SSE2", ff_conv_s16_to_s16p_2ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, - 6, 16, 4, "SSE2", ff_conv_s16_to_s16p_6ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16, - 2, 16, 8, "SSE2", ff_conv_s16_to_fltp_2ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16, - 6, 16, 4, "SSE2", ff_conv_s16_to_fltp_6ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT, - 2, 16, 8, "SSE2", ff_conv_flt_to_s16p_2ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT, - 6, 16, 4, "SSE2", ff_conv_flt_to_s16p_6ch_sse2); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT, - 6, 16, 4, "SSE2", ff_conv_flt_to_fltp_6ch_sse2); - } - if (EXTERNAL_SSSE3(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P, - 6, 16, 4, "SSSE3", ff_conv_s16p_to_flt_6ch_ssse3); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 2, 16, 4, "SSSE3", ff_conv_fltp_to_s16_2ch_ssse3); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, - 2, 16, 8, "SSSE3", ff_conv_s16_to_s16p_2ch_ssse3); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, - 6, 16, 4, "SSSE3", ff_conv_s16_to_s16p_6ch_ssse3); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16, - 6, 16, 4, "SSSE3", ff_conv_s16_to_fltp_6ch_ssse3); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT, - 6, 16, 4, "SSSE3", ff_conv_flt_to_s16p_6ch_ssse3); - } - if (EXTERNAL_SSE4(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16, - 0, 16, 8, "SSE4", ff_conv_s16_to_flt_sse4); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, - 6, 16, 4, "SSE4", ff_conv_fltp_to_flt_6ch_sse4); - } - if (EXTERNAL_AVX_FAST(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32, - 0, 32, 16, "AVX", ff_conv_s32_to_flt_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT, - 0, 32, 32, "AVX", ff_conv_flt_to_s32_avx); - } - if (EXTERNAL_AVX(cpu_flags)) { - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, - 2, 16, 16, "AVX", ff_conv_s16p_to_s16_2ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P, - 6, 16, 8, "AVX", ff_conv_s16p_to_s16_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P, - 2, 16, 8, "AVX", ff_conv_s16p_to_flt_2ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16P, - 6, 16, 4, "AVX", ff_conv_s16p_to_flt_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLTP, - 6, 16, 4, "AVX", ff_conv_fltp_to_s16_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, - 6, 16, 4, "AVX", ff_conv_fltp_to_flt_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, - 2, 16, 8, "AVX", ff_conv_s16_to_s16p_2ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S16, - 6, 16, 4, "AVX", ff_conv_s16_to_s16p_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16, - 2, 16, 8, "AVX", ff_conv_s16_to_fltp_2ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_S16, - 6, 16, 4, "AVX", ff_conv_s16_to_fltp_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT, - 2, 16, 8, "AVX", ff_conv_flt_to_s16p_2ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLT, - 6, 16, 4, "AVX", ff_conv_flt_to_s16p_6ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT, - 2, 16, 4, "AVX", ff_conv_flt_to_fltp_2ch_avx); - ff_audio_convert_set_func(ac, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_FLT, - 6, 16, 4, "AVX", ff_conv_flt_to_fltp_6ch_avx); - } -} diff --git a/libavresample/x86/audio_mix.asm b/libavresample/x86/audio_mix.asm deleted file mode 100644 index fe27d6a6c9..0000000000 --- a/libavresample/x86/audio_mix.asm +++ /dev/null @@ -1,511 +0,0 @@ -;****************************************************************************** -;* x86 optimized channel mixing -;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> -;* -;* 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 "libavutil/x86/x86util.asm" -%include "util.asm" - -SECTION .text - -;----------------------------------------------------------------------------- -; void ff_mix_2_to_1_fltp_flt(float **src, float **matrix, int len, -; int out_ch, int in_ch); -;----------------------------------------------------------------------------- - -%macro MIX_2_TO_1_FLTP_FLT 0 -cglobal mix_2_to_1_fltp_flt, 3,4,6, src, matrix, len, src1 - mov src1q, [srcq+gprsize] - mov srcq, [srcq ] - sub src1q, srcq - mov matrixq, [matrixq ] - VBROADCASTSS m4, [matrixq ] - VBROADCASTSS m5, [matrixq+4] - ALIGN 16 -.loop: - mulps m0, m4, [srcq ] - mulps m1, m5, [srcq+src1q ] - mulps m2, m4, [srcq+ mmsize] - mulps m3, m5, [srcq+src1q+mmsize] - addps m0, m0, m1 - addps m2, m2, m3 - mova [srcq ], m0 - mova [srcq+mmsize], m2 - add srcq, mmsize*2 - sub lend, mmsize*2/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse -MIX_2_TO_1_FLTP_FLT -%if HAVE_AVX_EXTERNAL -INIT_YMM avx -MIX_2_TO_1_FLTP_FLT -%endif - -;----------------------------------------------------------------------------- -; void ff_mix_2_to_1_s16p_flt(int16_t **src, float **matrix, int len, -; int out_ch, int in_ch); -;----------------------------------------------------------------------------- - -%macro MIX_2_TO_1_S16P_FLT 0 -cglobal mix_2_to_1_s16p_flt, 3,4,6, src, matrix, len, src1 - mov src1q, [srcq+gprsize] - mov srcq, [srcq] - sub src1q, srcq - mov matrixq, [matrixq ] - VBROADCASTSS m4, [matrixq ] - VBROADCASTSS m5, [matrixq+4] - ALIGN 16 -.loop: - mova m0, [srcq ] - mova m2, [srcq+src1q] - S16_TO_S32_SX 0, 1 - S16_TO_S32_SX 2, 3 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - cvtdq2ps m2, m2 - cvtdq2ps m3, m3 - mulps m0, m4 - mulps m1, m4 - mulps m2, m5 - mulps m3, m5 - addps m0, m2 - addps m1, m3 - cvtps2dq m0, m0 - cvtps2dq m1, m1 - packssdw m0, m1 - mova [srcq], m0 - add srcq, mmsize - sub lend, mmsize/2 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -MIX_2_TO_1_S16P_FLT -INIT_XMM sse4 -MIX_2_TO_1_S16P_FLT - -;----------------------------------------------------------------------------- -; void ff_mix_2_to_1_s16p_q8(int16_t **src, int16_t **matrix, int len, -; int out_ch, int in_ch); -;----------------------------------------------------------------------------- - -INIT_XMM sse2 -cglobal mix_2_to_1_s16p_q8, 3,4,6, src, matrix, len, src1 - mov src1q, [srcq+gprsize] - mov srcq, [srcq] - sub src1q, srcq - mov matrixq, [matrixq] - movd m4, [matrixq] - movd m5, [matrixq] - SPLATW m4, m4, 0 - SPLATW m5, m5, 1 - pxor m0, m0 - punpcklwd m4, m0 - punpcklwd m5, m0 - ALIGN 16 -.loop: - mova m0, [srcq ] - mova m2, [srcq+src1q] - punpckhwd m1, m0, m0 - punpcklwd m0, m0 - punpckhwd m3, m2, m2 - punpcklwd m2, m2 - pmaddwd m0, m4 - pmaddwd m1, m4 - pmaddwd m2, m5 - pmaddwd m3, m5 - paddd m0, m2 - paddd m1, m3 - psrad m0, 8 - psrad m1, 8 - packssdw m0, m1 - mova [srcq], m0 - add srcq, mmsize - sub lend, mmsize/2 - jg .loop - REP_RET - -;----------------------------------------------------------------------------- -; void ff_mix_1_to_2_fltp_flt(float **src, float **matrix, int len, -; int out_ch, int in_ch); -;----------------------------------------------------------------------------- - -%macro MIX_1_TO_2_FLTP_FLT 0 -cglobal mix_1_to_2_fltp_flt, 3,5,4, src0, matrix0, len, src1, matrix1 - mov src1q, [src0q+gprsize] - mov src0q, [src0q] - sub src1q, src0q - mov matrix1q, [matrix0q+gprsize] - mov matrix0q, [matrix0q] - VBROADCASTSS m2, [matrix0q] - VBROADCASTSS m3, [matrix1q] - ALIGN 16 -.loop: - mova m0, [src0q] - mulps m1, m0, m3 - mulps m0, m0, m2 - mova [src0q ], m0 - mova [src0q+src1q], m1 - add src0q, mmsize - sub lend, mmsize/4 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse -MIX_1_TO_2_FLTP_FLT -%if HAVE_AVX_EXTERNAL -INIT_YMM avx -MIX_1_TO_2_FLTP_FLT -%endif - -;----------------------------------------------------------------------------- -; void ff_mix_1_to_2_s16p_flt(int16_t **src, float **matrix, int len, -; int out_ch, int in_ch); -;----------------------------------------------------------------------------- - -%macro MIX_1_TO_2_S16P_FLT 0 -cglobal mix_1_to_2_s16p_flt, 3,5,6, src0, matrix0, len, src1, matrix1 - mov src1q, [src0q+gprsize] - mov src0q, [src0q] - sub src1q, src0q - mov matrix1q, [matrix0q+gprsize] - mov matrix0q, [matrix0q] - VBROADCASTSS m4, [matrix0q] - VBROADCASTSS m5, [matrix1q] - ALIGN 16 -.loop: - mova m0, [src0q] - S16_TO_S32_SX 0, 2 - cvtdq2ps m0, m0 - cvtdq2ps m2, m2 - mulps m1, m0, m5 - mulps m0, m0, m4 - mulps m3, m2, m5 - mulps m2, m2, m4 - cvtps2dq m0, m0 - cvtps2dq m1, m1 - cvtps2dq m2, m2 - cvtps2dq m3, m3 - packssdw m0, m2 - packssdw m1, m3 - mova [src0q ], m0 - mova [src0q+src1q], m1 - add src0q, mmsize - sub lend, mmsize/2 - jg .loop - REP_RET -%endmacro - -INIT_XMM sse2 -MIX_1_TO_2_S16P_FLT -INIT_XMM sse4 -MIX_1_TO_2_S16P_FLT -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -MIX_1_TO_2_S16P_FLT -%endif - -;----------------------------------------------------------------------------- -; void ff_mix_3_8_to_1_2_fltp/s16p_flt(float/int16_t **src, float **matrix, -; int len, int out_ch, int in_ch); -;----------------------------------------------------------------------------- - -%macro MIX_3_8_TO_1_2_FLT 3 ; %1 = in channels, %2 = out channels, %3 = s16p or fltp -; define some names to make the code clearer -%assign in_channels %1 -%assign out_channels %2 -%assign stereo out_channels - 1 -%ifidn %3, s16p - %assign is_s16 1 -%else - %assign is_s16 0 -%endif - -; determine how many matrix elements must go on the stack vs. mmregs -%assign matrix_elements in_channels * out_channels -%if is_s16 - %if stereo - %assign needed_mmregs 7 - %else - %assign needed_mmregs 5 - %endif -%else - %if stereo - %assign needed_mmregs 4 - %else - %assign needed_mmregs 3 - %endif -%endif -%assign matrix_elements_mm num_mmregs - needed_mmregs -%if matrix_elements < matrix_elements_mm - %assign matrix_elements_mm matrix_elements -%endif -%if matrix_elements_mm < matrix_elements - %assign matrix_elements_stack matrix_elements - matrix_elements_mm -%else - %assign matrix_elements_stack 0 -%endif -%assign matrix_stack_size matrix_elements_stack * mmsize - -%assign needed_stack_size -1 * matrix_stack_size -%if ARCH_X86_32 && in_channels >= 7 -%assign needed_stack_size needed_stack_size - 16 -%endif - -cglobal mix_%1_to_%2_%3_flt, 3,in_channels+2,needed_mmregs+matrix_elements_mm, needed_stack_size, src0, src1, len, src2, src3, src4, src5, src6, src7 - -; define src pointers on stack if needed -%if matrix_elements_stack > 0 && ARCH_X86_32 && in_channels >= 7 - %define src5m [rsp+matrix_stack_size+0] - %define src6m [rsp+matrix_stack_size+4] - %define src7m [rsp+matrix_stack_size+8] -%endif - -; load matrix pointers -%define matrix0q r1q -%define matrix1q r3q -%if stereo - mov matrix1q, [matrix0q+gprsize] -%endif - mov matrix0q, [matrix0q] - -; define matrix coeff names -%assign %%i 0 -%assign %%j needed_mmregs -%rep in_channels - %if %%i >= matrix_elements_mm - CAT_XDEFINE mx_stack_0_, %%i, 1 - CAT_XDEFINE mx_0_, %%i, [rsp+(%%i-matrix_elements_mm)*mmsize] - %else - CAT_XDEFINE mx_stack_0_, %%i, 0 - CAT_XDEFINE mx_0_, %%i, m %+ %%j - %assign %%j %%j+1 - %endif - %assign %%i %%i+1 -%endrep -%if stereo -%assign %%i 0 -%rep in_channels - %if in_channels + %%i >= matrix_elements_mm - CAT_XDEFINE mx_stack_1_, %%i, 1 - CAT_XDEFINE mx_1_, %%i, [rsp+(in_channels+%%i-matrix_elements_mm)*mmsize] - %else - CAT_XDEFINE mx_stack_1_, %%i, 0 - CAT_XDEFINE mx_1_, %%i, m %+ %%j - %assign %%j %%j+1 - %endif - %assign %%i %%i+1 -%endrep -%endif - -; load/splat matrix coeffs -%assign %%i 0 -%rep in_channels - %if mx_stack_0_ %+ %%i - VBROADCASTSS m0, [matrix0q+4*%%i] - mova mx_0_ %+ %%i, m0 - %else - VBROADCASTSS mx_0_ %+ %%i, [matrix0q+4*%%i] - %endif - %if stereo - %if mx_stack_1_ %+ %%i - VBROADCASTSS m0, [matrix1q+4*%%i] - mova mx_1_ %+ %%i, m0 - %else - VBROADCASTSS mx_1_ %+ %%i, [matrix1q+4*%%i] - %endif - %endif - %assign %%i %%i+1 -%endrep - -; load channel pointers to registers as offsets from the first channel pointer -%if ARCH_X86_64 - movsxd lenq, r2d -%endif - shl lenq, 2-is_s16 -%assign %%i 1 -%rep (in_channels - 1) - %if ARCH_X86_32 && in_channels >= 7 && %%i >= 5 - mov src5q, [src0q+%%i*gprsize] - add src5q, lenq - mov src %+ %%i %+ m, src5q - %else - mov src %+ %%i %+ q, [src0q+%%i*gprsize] - add src %+ %%i %+ q, lenq - %endif - %assign %%i %%i+1 -%endrep - mov src0q, [src0q] - add src0q, lenq - neg lenq -.loop: -; for x86-32 with 7-8 channels we do not have enough gp registers for all src -; pointers, so we have to load some of them from the stack each time -%define copy_src_from_stack ARCH_X86_32 && in_channels >= 7 && %%i >= 5 -%if is_s16 - ; mix with s16p input - mova m0, [src0q+lenq] - S16_TO_S32_SX 0, 1 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - %if stereo - mulps m2, m0, mx_1_0 - mulps m3, m1, mx_1_0 - %endif - mulps m0, m0, mx_0_0 - mulps m1, m1, mx_0_0 -%assign %%i 1 -%rep (in_channels - 1) - %if copy_src_from_stack - %define src_ptr src5q - %else - %define src_ptr src %+ %%i %+ q - %endif - %if stereo - %if copy_src_from_stack - mov src_ptr, src %+ %%i %+ m - %endif - mova m4, [src_ptr+lenq] - S16_TO_S32_SX 4, 5 - cvtdq2ps m4, m4 - cvtdq2ps m5, m5 - FMULADD_PS m2, m4, mx_1_ %+ %%i, m2, m6 - FMULADD_PS m3, m5, mx_1_ %+ %%i, m3, m6 - FMULADD_PS m0, m4, mx_0_ %+ %%i, m0, m4 - FMULADD_PS m1, m5, mx_0_ %+ %%i, m1, m5 - %else - %if copy_src_from_stack - mov src_ptr, src %+ %%i %+ m - %endif - mova m2, [src_ptr+lenq] - S16_TO_S32_SX 2, 3 - cvtdq2ps m2, m2 - cvtdq2ps m3, m3 - FMULADD_PS m0, m2, mx_0_ %+ %%i, m0, m4 - FMULADD_PS m1, m3, mx_0_ %+ %%i, m1, m4 - %endif - %assign %%i %%i+1 -%endrep - %if stereo - cvtps2dq m2, m2 - cvtps2dq m3, m3 - packssdw m2, m3 - mova [src1q+lenq], m2 - %endif - cvtps2dq m0, m0 - cvtps2dq m1, m1 - packssdw m0, m1 - mova [src0q+lenq], m0 -%else - ; mix with fltp input - %if stereo || mx_stack_0_0 - mova m0, [src0q+lenq] - %endif - %if stereo - mulps m1, m0, mx_1_0 - %endif - %if stereo || mx_stack_0_0 - mulps m0, m0, mx_0_0 - %else - mulps m0, mx_0_0, [src0q+lenq] - %endif -%assign %%i 1 -%rep (in_channels - 1) - %if copy_src_from_stack - %define src_ptr src5q - mov src_ptr, src %+ %%i %+ m - %else - %define src_ptr src %+ %%i %+ q - %endif - ; avoid extra load for mono if matrix is in a mm register - %if stereo || mx_stack_0_ %+ %%i - mova m2, [src_ptr+lenq] - %endif - %if stereo - FMULADD_PS m1, m2, mx_1_ %+ %%i, m1, m3 - %endif - %if stereo || mx_stack_0_ %+ %%i - FMULADD_PS m0, m2, mx_0_ %+ %%i, m0, m2 - %else - FMULADD_PS m0, mx_0_ %+ %%i, [src_ptr+lenq], m0, m1 - %endif - %assign %%i %%i+1 -%endrep - mova [src0q+lenq], m0 - %if stereo - mova [src1q+lenq], m1 - %endif -%endif - - add lenq, mmsize - jl .loop -; zero ymm high halves -%if mmsize == 32 - vzeroupper -%endif - RET -%endmacro - -%macro MIX_3_8_TO_1_2_FLT_FUNCS 0 -%assign %%i 3 -%rep 6 - INIT_XMM sse - MIX_3_8_TO_1_2_FLT %%i, 1, fltp - MIX_3_8_TO_1_2_FLT %%i, 2, fltp - INIT_XMM sse2 - MIX_3_8_TO_1_2_FLT %%i, 1, s16p - MIX_3_8_TO_1_2_FLT %%i, 2, s16p - INIT_XMM sse4 - MIX_3_8_TO_1_2_FLT %%i, 1, s16p - MIX_3_8_TO_1_2_FLT %%i, 2, s16p - ; do not use ymm AVX or FMA4 in x86-32 for 6 or more channels due to stack alignment issues - %if HAVE_AVX_EXTERNAL - %if ARCH_X86_64 || %%i < 6 - INIT_YMM avx - %else - INIT_XMM avx - %endif - MIX_3_8_TO_1_2_FLT %%i, 1, fltp - MIX_3_8_TO_1_2_FLT %%i, 2, fltp - INIT_XMM avx - MIX_3_8_TO_1_2_FLT %%i, 1, s16p - MIX_3_8_TO_1_2_FLT %%i, 2, s16p - %endif - %if HAVE_FMA4_EXTERNAL - %if ARCH_X86_64 || %%i < 6 - INIT_YMM fma4 - %else - INIT_XMM fma4 - %endif - MIX_3_8_TO_1_2_FLT %%i, 1, fltp - MIX_3_8_TO_1_2_FLT %%i, 2, fltp - INIT_XMM fma4 - MIX_3_8_TO_1_2_FLT %%i, 1, s16p - MIX_3_8_TO_1_2_FLT %%i, 2, s16p - %endif - %assign %%i %%i+1 -%endrep -%endmacro - -MIX_3_8_TO_1_2_FLT_FUNCS diff --git a/libavresample/x86/audio_mix_init.c b/libavresample/x86/audio_mix_init.c deleted file mode 100644 index 9b86be2847..0000000000 --- a/libavresample/x86/audio_mix_init.c +++ /dev/null @@ -1,215 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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/cpu.h" -#include "libavutil/x86/cpu.h" -#include "libavresample/audio_mix.h" - -void ff_mix_2_to_1_fltp_flt_sse(float **src, float **matrix, int len, - int out_ch, int in_ch); -void ff_mix_2_to_1_fltp_flt_avx(float **src, float **matrix, int len, - int out_ch, int in_ch); - -void ff_mix_2_to_1_s16p_flt_sse2(int16_t **src, float **matrix, int len, - int out_ch, int in_ch); -void ff_mix_2_to_1_s16p_flt_sse4(int16_t **src, float **matrix, int len, - int out_ch, int in_ch); - -void ff_mix_2_to_1_s16p_q8_sse2(int16_t **src, int16_t **matrix, - int len, int out_ch, int in_ch); - -void ff_mix_1_to_2_fltp_flt_sse(float **src, float **matrix, int len, - int out_ch, int in_ch); -void ff_mix_1_to_2_fltp_flt_avx(float **src, float **matrix, int len, - int out_ch, int in_ch); - -void ff_mix_1_to_2_s16p_flt_sse2(int16_t **src, float **matrix, int len, - int out_ch, int in_ch); -void ff_mix_1_to_2_s16p_flt_sse4(int16_t **src, float **matrix, int len, - int out_ch, int in_ch); -void ff_mix_1_to_2_s16p_flt_avx (int16_t **src, float **matrix, int len, - int out_ch, int in_ch); - -#define DEFINE_MIX_3_8_TO_1_2(chan) \ -void ff_mix_ ## chan ## _to_1_fltp_flt_sse(float **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_fltp_flt_sse(float **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ - \ -void ff_mix_ ## chan ## _to_1_s16p_flt_sse2(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_s16p_flt_sse2(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ - \ -void ff_mix_ ## chan ## _to_1_s16p_flt_sse4(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_s16p_flt_sse4(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ - \ -void ff_mix_ ## chan ## _to_1_fltp_flt_avx(float **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_fltp_flt_avx(float **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ - \ -void ff_mix_ ## chan ## _to_1_s16p_flt_avx(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_s16p_flt_avx(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ - \ -void ff_mix_ ## chan ## _to_1_fltp_flt_fma4(float **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_fltp_flt_fma4(float **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ - \ -void ff_mix_ ## chan ## _to_1_s16p_flt_fma4(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); \ -void ff_mix_ ## chan ## _to_2_s16p_flt_fma4(int16_t **src, \ - float **matrix, int len, \ - int out_ch, int in_ch); - -DEFINE_MIX_3_8_TO_1_2(3) -DEFINE_MIX_3_8_TO_1_2(4) -DEFINE_MIX_3_8_TO_1_2(5) -DEFINE_MIX_3_8_TO_1_2(6) -DEFINE_MIX_3_8_TO_1_2(7) -DEFINE_MIX_3_8_TO_1_2(8) - -#define SET_MIX_3_8_TO_1_2(chan) \ - if (EXTERNAL_SSE(cpu_flags)) { \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, 16, 4, "SSE", \ - ff_mix_ ## chan ## _to_1_fltp_flt_sse); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, 16, 4, "SSE", \ - ff_mix_## chan ##_to_2_fltp_flt_sse); \ - } \ - if (EXTERNAL_SSE2(cpu_flags)) { \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, 16, 8, "SSE2", \ - ff_mix_ ## chan ## _to_1_s16p_flt_sse2); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, 16, 8, "SSE2", \ - ff_mix_ ## chan ## _to_2_s16p_flt_sse2); \ - } \ - if (EXTERNAL_SSE4(cpu_flags)) { \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, 16, 8, "SSE4", \ - ff_mix_ ## chan ## _to_1_s16p_flt_sse4); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, 16, 8, "SSE4", \ - ff_mix_ ## chan ## _to_2_s16p_flt_sse4); \ - } \ - if (EXTERNAL_AVX(cpu_flags)) { \ - int ptr_align = 32; \ - int smp_align = 8; \ - if (ARCH_X86_32 || chan >= 6) { \ - ptr_align = 16; \ - smp_align = 4; \ - } \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, ptr_align, smp_align, "AVX", \ - ff_mix_ ## chan ## _to_1_fltp_flt_avx); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, ptr_align, smp_align, "AVX", \ - ff_mix_ ## chan ## _to_2_fltp_flt_avx); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, 16, 8, "AVX", \ - ff_mix_ ## chan ## _to_1_s16p_flt_avx); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, 16, 8, "AVX", \ - ff_mix_ ## chan ## _to_2_s16p_flt_avx); \ - } \ - if (EXTERNAL_FMA4(cpu_flags)) { \ - int ptr_align = 32; \ - int smp_align = 8; \ - if (ARCH_X86_32 || chan >= 6) { \ - ptr_align = 16; \ - smp_align = 4; \ - } \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, ptr_align, smp_align, "FMA4", \ - ff_mix_ ## chan ## _to_1_fltp_flt_fma4); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, ptr_align, smp_align, "FMA4", \ - ff_mix_ ## chan ## _to_2_fltp_flt_fma4); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 1, 16, 8, "FMA4", \ - ff_mix_ ## chan ## _to_1_s16p_flt_fma4); \ - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT,\ - chan, 2, 16, 8, "FMA4", \ - ff_mix_ ## chan ## _to_2_s16p_flt_fma4); \ - } - -av_cold void ff_audio_mix_init_x86(AudioMix *am) -{ - int cpu_flags = av_get_cpu_flags(); - - if (EXTERNAL_SSE(cpu_flags)) { - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 2, 1, 16, 8, "SSE", ff_mix_2_to_1_fltp_flt_sse); - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 1, 2, 16, 4, "SSE", ff_mix_1_to_2_fltp_flt_sse); - } - if (EXTERNAL_SSE2(cpu_flags)) { - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_flt_sse2); - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_Q8, - 2, 1, 16, 8, "SSE2", ff_mix_2_to_1_s16p_q8_sse2); - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 1, 2, 16, 8, "SSE2", ff_mix_1_to_2_s16p_flt_sse2); - } - if (EXTERNAL_SSE4(cpu_flags)) { - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 2, 1, 16, 8, "SSE4", ff_mix_2_to_1_s16p_flt_sse4); - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 1, 2, 16, 8, "SSE4", ff_mix_1_to_2_s16p_flt_sse4); - } - if (EXTERNAL_AVX_FAST(cpu_flags)) { - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 2, 1, 32, 16, "AVX", ff_mix_2_to_1_fltp_flt_avx); - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_FLTP, AV_MIX_COEFF_TYPE_FLT, - 1, 2, 32, 8, "AVX", ff_mix_1_to_2_fltp_flt_avx); - } - if (EXTERNAL_AVX(cpu_flags)) { - ff_audio_mix_set_func(am, AV_SAMPLE_FMT_S16P, AV_MIX_COEFF_TYPE_FLT, - 1, 2, 16, 8, "AVX", ff_mix_1_to_2_s16p_flt_avx); - } - - SET_MIX_3_8_TO_1_2(3) - SET_MIX_3_8_TO_1_2(4) - SET_MIX_3_8_TO_1_2(5) - SET_MIX_3_8_TO_1_2(6) - SET_MIX_3_8_TO_1_2(7) - SET_MIX_3_8_TO_1_2(8) -} diff --git a/libavresample/x86/dither.asm b/libavresample/x86/dither.asm deleted file mode 100644 index d677c7179a..0000000000 --- a/libavresample/x86/dither.asm +++ /dev/null @@ -1,117 +0,0 @@ -;****************************************************************************** -;* x86 optimized dithering format conversion -;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> -;* -;* 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 "libavutil/x86/x86util.asm" - -SECTION_RODATA 32 - -; 1.0f / (2.0f * INT32_MAX) -pf_dither_scale: times 8 dd 2.32830643762e-10 - -pf_s16_scale: times 4 dd 32753.0 - -SECTION .text - -;------------------------------------------------------------------------------ -; void ff_quantize(int16_t *dst, float *src, float *dither, int len); -;------------------------------------------------------------------------------ - -INIT_XMM sse2 -cglobal quantize, 4,4,3, dst, src, dither, len - lea lenq, [2*lend] - add dstq, lenq - lea srcq, [srcq+2*lenq] - lea ditherq, [ditherq+2*lenq] - neg lenq - mova m2, [pf_s16_scale] -.loop: - mulps m0, m2, [srcq+2*lenq] - mulps m1, m2, [srcq+2*lenq+mmsize] - addps m0, [ditherq+2*lenq] - addps m1, [ditherq+2*lenq+mmsize] - cvtps2dq m0, m0 - cvtps2dq m1, m1 - packssdw m0, m1 - mova [dstq+lenq], m0 - add lenq, mmsize - jl .loop - REP_RET - -;------------------------------------------------------------------------------ -; void ff_dither_int_to_float_rectangular(float *dst, int *src, int len) -;------------------------------------------------------------------------------ - -%macro DITHER_INT_TO_FLOAT_RECTANGULAR 0 -cglobal dither_int_to_float_rectangular, 3,3,3, dst, src, len - lea lenq, [4*lend] - add srcq, lenq - add dstq, lenq - neg lenq - mova m0, [pf_dither_scale] -.loop: - cvtdq2ps m1, [srcq+lenq] - cvtdq2ps m2, [srcq+lenq+mmsize] - mulps m1, m1, m0 - mulps m2, m2, m0 - mova [dstq+lenq], m1 - mova [dstq+lenq+mmsize], m2 - add lenq, 2*mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -DITHER_INT_TO_FLOAT_RECTANGULAR -INIT_YMM avx -DITHER_INT_TO_FLOAT_RECTANGULAR - -;------------------------------------------------------------------------------ -; void ff_dither_int_to_float_triangular(float *dst, int *src0, int len) -;------------------------------------------------------------------------------ - -%macro DITHER_INT_TO_FLOAT_TRIANGULAR 0 -cglobal dither_int_to_float_triangular, 3,4,5, dst, src0, len, src1 - lea lenq, [4*lend] - lea src1q, [src0q+2*lenq] - add src0q, lenq - add dstq, lenq - neg lenq - mova m0, [pf_dither_scale] -.loop: - cvtdq2ps m1, [src0q+lenq] - cvtdq2ps m2, [src0q+lenq+mmsize] - cvtdq2ps m3, [src1q+lenq] - cvtdq2ps m4, [src1q+lenq+mmsize] - addps m1, m1, m3 - addps m2, m2, m4 - mulps m1, m1, m0 - mulps m2, m2, m0 - mova [dstq+lenq], m1 - mova [dstq+lenq+mmsize], m2 - add lenq, 2*mmsize - jl .loop - REP_RET -%endmacro - -INIT_XMM sse2 -DITHER_INT_TO_FLOAT_TRIANGULAR -INIT_YMM avx -DITHER_INT_TO_FLOAT_TRIANGULAR diff --git a/libavresample/x86/dither_init.c b/libavresample/x86/dither_init.c deleted file mode 100644 index ad157b96b1..0000000000 --- a/libavresample/x86/dither_init.c +++ /dev/null @@ -1,60 +0,0 @@ -/* - * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> - * - * 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/cpu.h" -#include "libavutil/x86/cpu.h" -#include "libavresample/dither.h" - -void ff_quantize_sse2(int16_t *dst, const float *src, float *dither, int len); - -void ff_dither_int_to_float_rectangular_sse2(float *dst, int *src, int len); -void ff_dither_int_to_float_rectangular_avx(float *dst, int *src, int len); - -void ff_dither_int_to_float_triangular_sse2(float *dst, int *src0, int len); -void ff_dither_int_to_float_triangular_avx(float *dst, int *src0, int len); - -av_cold void ff_dither_init_x86(DitherDSPContext *ddsp, - enum AVResampleDitherMethod method) -{ - int cpu_flags = av_get_cpu_flags(); - - if (EXTERNAL_SSE2(cpu_flags)) { - ddsp->quantize = ff_quantize_sse2; - ddsp->ptr_align = 16; - ddsp->samples_align = 8; - } - - if (method == AV_RESAMPLE_DITHER_RECTANGULAR) { - if (EXTERNAL_SSE2(cpu_flags)) { - ddsp->dither_int_to_float = ff_dither_int_to_float_rectangular_sse2; - } - if (EXTERNAL_AVX_FAST(cpu_flags)) { - ddsp->dither_int_to_float = ff_dither_int_to_float_rectangular_avx; - } - } else { - if (EXTERNAL_SSE2(cpu_flags)) { - ddsp->dither_int_to_float = ff_dither_int_to_float_triangular_sse2; - } - if (EXTERNAL_AVX_FAST(cpu_flags)) { - ddsp->dither_int_to_float = ff_dither_int_to_float_triangular_avx; - } - } -} diff --git a/libavresample/x86/util.asm b/libavresample/x86/util.asm deleted file mode 100644 index 187a4a21ba..0000000000 --- a/libavresample/x86/util.asm +++ /dev/null @@ -1,41 +0,0 @@ -;****************************************************************************** -;* x86 utility macros for libavresample -;* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> -;* -;* 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 -;****************************************************************************** - -%macro S16_TO_S32_SX 2 ; src/low dst, high dst -%if cpuflag(sse4) - pmovsxwd m%2, m%1 - psrldq m%1, 8 - pmovsxwd m%1, m%1 - SWAP %1, %2 -%else - mova m%2, m%1 - punpckhwd m%2, m%2 - punpcklwd m%1, m%1 - psrad m%2, 16 - psrad m%1, 16 -%endif -%endmacro - -%macro DEINT2_PS 3 ; src0/even dst, src1/odd dst, temp - shufps m%3, m%1, m%2, q3131 - shufps m%1, m%2, q2020 - SWAP %2,%3 -%endmacro diff --git a/libavresample/x86/w64xmmtest.c b/libavresample/x86/w64xmmtest.c deleted file mode 100644 index 0f42bd185c..0000000000 --- a/libavresample/x86/w64xmmtest.c +++ /dev/null @@ -1,31 +0,0 @@ -/* - * check XMM registers for clobbers on Win64 - * Copyright (c) 2013 Martin Storsjo - * - * 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 "libavresample/avresample.h" -#include "libavutil/x86/w64xmmtest.h" - -wrap(avresample_convert(AVAudioResampleContext *avr, uint8_t **output, - int out_plane_size, int out_samples, uint8_t **input, - int in_plane_size, int in_samples)) -{ - testxmmclobbers(avresample_convert, avr, output, out_plane_size, - out_samples, input, in_plane_size, in_samples); -} |