diff options
Diffstat (limited to 'ffmpeg1/libswresample')
27 files changed, 0 insertions, 5336 deletions
diff --git a/ffmpeg1/libswresample/Makefile b/ffmpeg1/libswresample/Makefile deleted file mode 100644 index 0b75bd0..0000000 --- a/ffmpeg1/libswresample/Makefile +++ /dev/null @@ -1,18 +0,0 @@ -include $(SUBDIR)../config.mak - -NAME = swresample -FFLIBS = avutil - -HEADERS = swresample.h \ - version.h \ - -OBJS = audioconvert.o \ - dither.o \ - rematrix.o \ - resample.o \ - swresample.o \ - -OBJS-$(CONFIG_LIBSOXR) += soxr_resample.o -OBJS-$(CONFIG_SHARED) += log2_tab.o - -TESTPROGS = swresample diff --git a/ffmpeg1/libswresample/arm/Makefile b/ffmpeg1/libswresample/arm/Makefile deleted file mode 100644 index 55683cb..0000000 --- a/ffmpeg1/libswresample/arm/Makefile +++ /dev/null @@ -1,2 +0,0 @@ -OBJS += arm/audio_convert_init.o -NEON-OBJS += arm/audio_convert_neon.o diff --git a/ffmpeg1/libswresample/arm/audio_convert_init.c b/ffmpeg1/libswresample/arm/audio_convert_init.c deleted file mode 100644 index ec9e62e..0000000 --- a/ffmpeg1/libswresample/arm/audio_convert_init.c +++ /dev/null @@ -1,67 +0,0 @@ -/* - * This file is part of libswresample. - * - * libswresample 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. - * - * libswresample 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 libswresample; 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 "libswresample/swresample_internal.h" -#include "libswresample/audioconvert.h" - -void swri_oldapi_conv_flt_to_s16_neon(int16_t *dst, const float *src, int len); -void swri_oldapi_conv_fltp_to_s16_2ch_neon(int16_t *dst, float *const *src, int len, int channels); -void swri_oldapi_conv_fltp_to_s16_nch_neon(int16_t *dst, float *const *src, int len, int channels); - -static void conv_flt_to_s16_neon(uint8_t **dst, const uint8_t **src, int len){ - swri_oldapi_conv_flt_to_s16_neon((int16_t*)*dst, (const float*)*src, len); -} - -static void conv_fltp_to_s16_2ch_neon(uint8_t **dst, const uint8_t **src, int len){ - swri_oldapi_conv_fltp_to_s16_2ch_neon((int16_t*)*dst, (float *const*)src, len, 2); -} - -static void conv_fltp_to_s16_nch_neon(uint8_t **dst, const uint8_t **src, int len){ - int channels; - for(channels=3; channels<SWR_CH_MAX && src[channels]; channels++) - ; - swri_oldapi_conv_fltp_to_s16_nch_neon((int16_t*)*dst, (float *const*)src, len, channels); -} - -av_cold void swri_audio_convert_init_arm(struct AudioConvert *ac, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels) -{ - int cpu_flags = av_get_cpu_flags(); - - ac->simd_f= NULL; - - if (have_neon(cpu_flags)) { - if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = conv_flt_to_s16_neon; - if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP && channels == 2) - ac->simd_f = conv_fltp_to_s16_2ch_neon; - if(out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP && channels > 2) - ac->simd_f = conv_fltp_to_s16_nch_neon; - if(ac->simd_f) - ac->in_simd_align_mask = ac->out_simd_align_mask = 15; - } -} diff --git a/ffmpeg1/libswresample/arm/audio_convert_neon.S b/ffmpeg1/libswresample/arm/audio_convert_neon.S deleted file mode 100644 index 471a2d8..0000000 --- a/ffmpeg1/libswresample/arm/audio_convert_neon.S +++ /dev/null @@ -1,363 +0,0 @@ -/* - * Copyright (c) 2008 Mans Rullgard <mans@mansr.com> - * - * This file is part of libswresample. - * - * libswresample 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. - * - * libswresample 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 libswresample; 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 swri_oldapi_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 swri_oldapi_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 swri_oldapi_conv_fltp_to_s16_nch_neon, export=1 - cmp r3, #2 - itt lt - ldrlt r1, [r1] - blt swri_oldapi_conv_flt_to_s16_neon - beq swri_oldapi_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/ffmpeg1/libswresample/audioconvert.c b/ffmpeg1/libswresample/audioconvert.c deleted file mode 100644 index 4ba0ff1..0000000 --- a/ffmpeg1/libswresample/audioconvert.c +++ /dev/null @@ -1,224 +0,0 @@ -/* - * audio conversion - * Copyright (c) 2006 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 - */ - -/** - * @file - * audio conversion - * @author Michael Niedermayer <michaelni@gmx.at> - */ - -#include "libavutil/avstring.h" -#include "libavutil/avassert.h" -#include "libavutil/libm.h" -#include "libavutil/samplefmt.h" -#include "audioconvert.h" - - -#define CONV_FUNC_NAME(dst_fmt, src_fmt) conv_ ## src_fmt ## _to_ ## dst_fmt - -//FIXME rounding ? -#define CONV_FUNC(ofmt, otype, ifmt, expr)\ -static void CONV_FUNC_NAME(ofmt, ifmt)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end)\ -{\ - uint8_t *end2 = end - 3*os;\ - while(po < end2){\ - *(otype*)po = expr; pi += is; po += os;\ - *(otype*)po = expr; pi += is; po += os;\ - *(otype*)po = expr; pi += is; po += os;\ - *(otype*)po = expr; pi += is; po += os;\ - }\ - while(po < end){\ - *(otype*)po = expr; pi += is; po += os;\ - }\ -} - -//FIXME put things below under ifdefs so we do not waste space for cases no codec will need -CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi) -CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) -CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7))) -CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) -CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) -CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi) -CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15))) -CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) -CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) -CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16) -CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31))) -CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) -CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80)) -CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15)))) -CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31)))) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi) -CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi) -CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80)) -CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15)))) -CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31)))) -CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi) -CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi) - -#define FMT_PAIR_FUNC(out, in) [out + AV_SAMPLE_FMT_NB*in] = CONV_FUNC_NAME(out, in) - -static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = { - FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_U8 ), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8 ), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8 ), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8 ), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8 ), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_S16), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_S32), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_FLT), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_DBL), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), - FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), -}; - -static void cpy1(uint8_t **dst, const uint8_t **src, int len){ - memcpy(*dst, *src, len); -} -static void cpy2(uint8_t **dst, const uint8_t **src, int len){ - memcpy(*dst, *src, 2*len); -} -static void cpy4(uint8_t **dst, const uint8_t **src, int len){ - memcpy(*dst, *src, 4*len); -} -static void cpy8(uint8_t **dst, const uint8_t **src, int len){ - memcpy(*dst, *src, 8*len); -} - -AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels, const int *ch_map, - int flags) -{ - AudioConvert *ctx; - conv_func_type *f = fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt) + AV_SAMPLE_FMT_NB*av_get_packed_sample_fmt(in_fmt)]; - - if (!f) - return NULL; - ctx = av_mallocz(sizeof(*ctx)); - if (!ctx) - return NULL; - - if(channels == 1){ - in_fmt = av_get_planar_sample_fmt( in_fmt); - out_fmt = av_get_planar_sample_fmt(out_fmt); - } - - ctx->channels = channels; - ctx->conv_f = f; - ctx->ch_map = ch_map; - if (in_fmt == AV_SAMPLE_FMT_U8 || in_fmt == AV_SAMPLE_FMT_U8P) - memset(ctx->silence, 0x80, sizeof(ctx->silence)); - - if(out_fmt == in_fmt && !ch_map) { - switch(av_get_bytes_per_sample(in_fmt)){ - case 1:ctx->simd_f = cpy1; break; - case 2:ctx->simd_f = cpy2; break; - case 4:ctx->simd_f = cpy4; break; - case 8:ctx->simd_f = cpy8; break; - } - } - - if(HAVE_YASM && HAVE_MMX) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels); - if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels); - - return ctx; -} - -void swri_audio_convert_free(AudioConvert **ctx) -{ - av_freep(ctx); -} - -int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len) -{ - int ch; - int off=0; - const int os= (out->planar ? 1 :out->ch_count) *out->bps; - unsigned misaligned = 0; - - av_assert0(ctx->channels == out->ch_count); - - if (ctx->in_simd_align_mask) { - int planes = in->planar ? in->ch_count : 1; - unsigned m = 0; - for (ch = 0; ch < planes; ch++) - m |= (intptr_t)in->ch[ch]; - misaligned |= m & ctx->in_simd_align_mask; - } - if (ctx->out_simd_align_mask) { - int planes = out->planar ? out->ch_count : 1; - unsigned m = 0; - for (ch = 0; ch < planes; ch++) - m |= (intptr_t)out->ch[ch]; - misaligned |= m & ctx->out_simd_align_mask; - } - - //FIXME optimize common cases - - if(ctx->simd_f && !ctx->ch_map && !misaligned){ - off = len&~15; - av_assert1(off>=0); - av_assert1(off<=len); - av_assert2(ctx->channels == SWR_CH_MAX || !in->ch[ctx->channels]); - if(off>0){ - if(out->planar == in->planar){ - int planes = out->planar ? out->ch_count : 1; - for(ch=0; ch<planes; ch++){ - ctx->simd_f(out->ch+ch, (const uint8_t **)in->ch+ch, off * (out->planar ? 1 :out->ch_count)); - } - }else{ - ctx->simd_f(out->ch, (const uint8_t **)in->ch, off); - } - } - if(off == len) - return 0; - } - - for(ch=0; ch<ctx->channels; ch++){ - const int ich= ctx->ch_map ? ctx->ch_map[ch] : ch; - const int is= ich < 0 ? 0 : (in->planar ? 1 : in->ch_count) * in->bps; - const uint8_t *pi= ich < 0 ? ctx->silence : in->ch[ich]; - uint8_t *po= out->ch[ch]; - uint8_t *end= po + os*len; - if(!po) - continue; - ctx->conv_f(po+off*os, pi+off*is, is, os, end); - } - return 0; -} diff --git a/ffmpeg1/libswresample/audioconvert.h b/ffmpeg1/libswresample/audioconvert.h deleted file mode 100644 index 2e983df..0000000 --- a/ffmpeg1/libswresample/audioconvert.h +++ /dev/null @@ -1,78 +0,0 @@ -/* - * audio conversion - * Copyright (c) 2006 Michael Niedermayer <michaelni@gmx.at> - * Copyright (c) 2008 Peter Ross - * - * 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 SWR_AUDIOCONVERT_H -#define SWR_AUDIOCONVERT_H - -/** - * @file - * Audio format conversion routines - */ - - -#include "swresample_internal.h" -#include "libavutil/cpu.h" - - -typedef void (conv_func_type)(uint8_t *po, const uint8_t *pi, int is, int os, uint8_t *end); -typedef void (simd_func_type)(uint8_t **dst, const uint8_t **src, int len); - -typedef struct AudioConvert { - int channels; - int in_simd_align_mask; - int out_simd_align_mask; - conv_func_type *conv_f; - simd_func_type *simd_f; - const int *ch_map; - uint8_t silence[8]; ///< silence input sample -}AudioConvert; - -/** - * Create an audio sample format converter context - * @param out_fmt Output sample format - * @param in_fmt Input sample format - * @param channels Number of channels - * @param flags See AV_CPU_FLAG_xx - * @param ch_map list of the channels id to pick from the source stream, NULL - * if all channels must be selected - * @return NULL on error - */ -AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels, const int *ch_map, - int flags); - -/** - * Free audio sample format converter context. - * and set the pointer to NULL - */ -void swri_audio_convert_free(AudioConvert **ctx); - -/** - * Convert between audio sample formats - * @param[in] out array of output buffers for each channel. set to NULL to ignore processing of the given channel. - * @param[in] in array of input buffers for each channel - * @param len length of audio frame size (measured in samples) - */ -int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len); - -#endif /* AUDIOCONVERT_H */ diff --git a/ffmpeg1/libswresample/dither.c b/ffmpeg1/libswresample/dither.c deleted file mode 100644 index d0193dd..0000000 --- a/ffmpeg1/libswresample/dither.c +++ /dev/null @@ -1,147 +0,0 @@ -/* - * Copyright (C) 2012-2013 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include "libavutil/avassert.h" -#include "swresample_internal.h" - -#include "noise_shaping_data.c" - -void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt) { - double scale = s->dither.noise_scale; -#define TMP_EXTRA 2 - double *tmp = av_malloc((len + TMP_EXTRA) * sizeof(double)); - int i; - - for(i=0; i<len + TMP_EXTRA; i++){ - double v; - seed = seed* 1664525 + 1013904223; - - switch(s->dither.method){ - case SWR_DITHER_RECTANGULAR: v= ((double)seed) / UINT_MAX - 0.5; break; - default: - av_assert0(s->dither.method < SWR_DITHER_NB); - v = ((double)seed) / UINT_MAX; - seed = seed*1664525 + 1013904223; - v-= ((double)seed) / UINT_MAX; - break; - } - tmp[i] = v; - } - - for(i=0; i<len; i++){ - double v; - - switch(s->dither.method){ - default: - av_assert0(s->dither.method < SWR_DITHER_NB); - v = tmp[i]; - break; - case SWR_DITHER_TRIANGULAR_HIGHPASS : - v = (- tmp[i] + 2*tmp[i+1] - tmp[i+2]) / sqrt(6); - break; - } - - v*= scale; - - switch(noise_fmt){ - case AV_SAMPLE_FMT_S16P: ((int16_t*)dst)[i] = v; break; - case AV_SAMPLE_FMT_S32P: ((int32_t*)dst)[i] = v; break; - case AV_SAMPLE_FMT_FLTP: ((float *)dst)[i] = v; break; - case AV_SAMPLE_FMT_DBLP: ((double *)dst)[i] = v; break; - default: av_assert0(0); - } - } - - av_free(tmp); -} - -int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt) -{ - int i; - double scale = 0; - - if (s->dither.method > SWR_DITHER_TRIANGULAR_HIGHPASS && s->dither.method <= SWR_DITHER_NS) - return AVERROR(EINVAL); - - out_fmt = av_get_packed_sample_fmt(out_fmt); - in_fmt = av_get_packed_sample_fmt( in_fmt); - - if(in_fmt == AV_SAMPLE_FMT_FLT || in_fmt == AV_SAMPLE_FMT_DBL){ - if(out_fmt == AV_SAMPLE_FMT_S32) scale = 1.0/(1L<<31); - if(out_fmt == AV_SAMPLE_FMT_S16) scale = 1.0/(1L<<15); - if(out_fmt == AV_SAMPLE_FMT_U8 ) scale = 1.0/(1L<< 7); - } - if(in_fmt == AV_SAMPLE_FMT_S32 && out_fmt == AV_SAMPLE_FMT_S16) scale = 1L<<16; - if(in_fmt == AV_SAMPLE_FMT_S32 && out_fmt == AV_SAMPLE_FMT_U8 ) scale = 1L<<24; - if(in_fmt == AV_SAMPLE_FMT_S16 && out_fmt == AV_SAMPLE_FMT_U8 ) scale = 1L<<8; - - scale *= s->dither.scale; - - if (out_fmt == AV_SAMPLE_FMT_S32 && s->dither.output_sample_bits) - scale *= 1<<(32-s->dither.output_sample_bits); - - s->dither.ns_pos = 0; - s->dither.noise_scale= scale; - s->dither.ns_scale = scale; - s->dither.ns_scale_1 = scale ? 1/scale : 0; - memset(s->dither.ns_errors, 0, sizeof(s->dither.ns_errors)); - for (i=0; filters[i].coefs; i++) { - const filter_t *f = &filters[i]; - if (fabs(s->out_sample_rate - f->rate) / f->rate <= .05 && f->name == s->dither.method) { - int j; - s->dither.ns_taps = f->len; - for (j=0; j<f->len; j++) - s->dither.ns_coeffs[j] = f->coefs[j]; - s->dither.ns_scale_1 *= 1 - exp(f->gain_cB * M_LN10 * 0.005) * 2 / (1<<(8*av_get_bytes_per_sample(out_fmt))); - break; - } - } - if (!filters[i].coefs && s->dither.method > SWR_DITHER_NS) { - av_log(s, AV_LOG_WARNING, "Requested noise shaping dither not available at this sampling rate, using triangular hp dither\n"); - s->dither.method = SWR_DITHER_TRIANGULAR_HIGHPASS; - } - - av_assert0(!s->preout.count); - s->dither.noise = s->preout; - s->dither.temp = s->preout; - if (s->dither.method > SWR_DITHER_NS) { - s->dither.noise.bps = 4; - s->dither.noise.fmt = AV_SAMPLE_FMT_FLTP; - s->dither.noise_scale = 1; - } - - return 0; -} - -#define TEMPLATE_DITHER_S16 -#include "dither_template.c" -#undef TEMPLATE_DITHER_S16 - -#define TEMPLATE_DITHER_S32 -#include "dither_template.c" -#undef TEMPLATE_DITHER_S32 - -#define TEMPLATE_DITHER_FLT -#include "dither_template.c" -#undef TEMPLATE_DITHER_FLT - -#define TEMPLATE_DITHER_DBL -#include "dither_template.c" -#undef TEMPLATE_DITHER_DBL diff --git a/ffmpeg1/libswresample/dither_template.c b/ffmpeg1/libswresample/dither_template.c deleted file mode 100644 index 4af7312..0000000 --- a/ffmpeg1/libswresample/dither_template.c +++ /dev/null @@ -1,67 +0,0 @@ - -#if defined(TEMPLATE_DITHER_DBL) -# define RENAME(N) N ## _double -# define DELEM double -# define CLIP(v) - -#elif defined(TEMPLATE_DITHER_FLT) -# define RENAME(N) N ## _float -# define DELEM float -# define CLIP(v) - -#elif defined(TEMPLATE_DITHER_S32) -# define RENAME(N) N ## _int32 -# define DELEM int32_t -# define CLIP(v) v = FFMAX(FFMIN(v, INT32_MAX), INT32_MIN) - -#elif defined(TEMPLATE_DITHER_S16) -# define RENAME(N) N ## _int16 -# define DELEM int16_t -# define CLIP(v) v = FFMAX(FFMIN(v, INT16_MAX), INT16_MIN) - -#else -ERROR -#endif - -void RENAME(swri_noise_shaping)(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count){ - int pos = s->dither.ns_pos; - int i, j, ch; - int taps = s->dither.ns_taps; - float S = s->dither.ns_scale; - float S_1 = s->dither.ns_scale_1; - - av_assert2((taps&3) != 2); - av_assert2((taps&3) != 3 || s->dither.ns_coeffs[taps] == 0); - - for (ch=0; ch<srcs->ch_count; ch++) { - const float *noise = ((const float *)noises->ch[ch]) + s->dither.noise_pos; - const DELEM *src = (const DELEM*)srcs->ch[ch]; - DELEM *dst = (DELEM*)dsts->ch[ch]; - float *ns_errors = s->dither.ns_errors[ch]; - const float *ns_coeffs = s->dither.ns_coeffs; - pos = s->dither.ns_pos; - for (i=0; i<count; i++) { - double d1, d = src[i]*S_1; - for(j=0; j<taps-2; j+=4) { - d -= ns_coeffs[j ] * ns_errors[pos + j ] - +ns_coeffs[j + 1] * ns_errors[pos + j + 1] - +ns_coeffs[j + 2] * ns_errors[pos + j + 2] - +ns_coeffs[j + 3] * ns_errors[pos + j + 3]; - } - if(j < taps) - d -= ns_coeffs[j] * ns_errors[pos + j]; - pos = pos ? pos - 1 : taps - 1; - d1 = rint(d + noise[i]); - ns_errors[pos + taps] = ns_errors[pos] = d1 - d; - d1 *= S; - CLIP(d1); - dst[i] = d1; - } - } - - s->dither.ns_pos = pos; -} - -#undef RENAME -#undef DELEM -#undef CLIP diff --git a/ffmpeg1/libswresample/libswresample.pc b/ffmpeg1/libswresample/libswresample.pc deleted file mode 100644 index f71c409..0000000 --- a/ffmpeg1/libswresample/libswresample.pc +++ /dev/null @@ -1,14 +0,0 @@ -prefix=/usr/local -exec_prefix=${prefix} -libdir=${prefix}/lib -includedir=${prefix}/include - -Name: libswresample -Description: FFmpeg audio resampling library -Version: 0.17.102 -Requires: -Requires.private: libavutil = 52.22.100 -Conflicts: -Libs: -L${libdir} -lswresample -Libs.private: -lm -Cflags: -I${includedir} diff --git a/ffmpeg1/libswresample/libswresample.v b/ffmpeg1/libswresample/libswresample.v deleted file mode 100644 index 9b797bd..0000000 --- a/ffmpeg1/libswresample/libswresample.v +++ /dev/null @@ -1,4 +0,0 @@ -LIBSWRESAMPLE_$MAJOR { - global: swr_*; ff_*; swresample_*; - local: *; -}; diff --git a/ffmpeg1/libswresample/log2_tab.c b/ffmpeg1/libswresample/log2_tab.c deleted file mode 100644 index 47a1df0..0000000 --- a/ffmpeg1/libswresample/log2_tab.c +++ /dev/null @@ -1 +0,0 @@ -#include "libavutil/log2_tab.c" diff --git a/ffmpeg1/libswresample/noise_shaping_data.c b/ffmpeg1/libswresample/noise_shaping_data.c deleted file mode 100644 index 77e0f2e..0000000 --- a/ffmpeg1/libswresample/noise_shaping_data.c +++ /dev/null @@ -1,224 +0,0 @@ -/* Effect: dither/noise-shape Copyright (c) 2008-9 robs@users.sourceforge.net - * - * This library 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. - * - * This library 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 this library; if not, write to the Free Software Foundation, - * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -typedef struct { - int rate; - enum {fir, iir} type; - size_t len; - int gain_cB; /* Chosen so clips are few if any, but not guaranteed none. */ - double const * coefs; - enum SwrDitherType name; -} filter_t; - -static double const lip44[] = {2.033, -2.165, 1.959, -1.590, .6149}; -static double const fwe44[] = { - 2.412, -3.370, 3.937, -4.174, 3.353, -2.205, 1.281, -.569, .0847}; -static double const mew44[] = { - 1.662, -1.263, .4827, -.2913, .1268, -.1124, .03252, -.01265, -.03524}; -static double const iew44[] = { - 2.847, -4.685, 6.214, -7.184, 6.639, -5.032, 3.263, -1.632, .4191}; -static double const ges44[] = { - 2.2061, -.4706, -.2534, -.6214, 1.0587, .0676, -.6054, -.2738}; -static double const ges48[] = { - 2.2374, -.7339, -.1251, -.6033, .903, .0116, -.5853, -.2571}; - -static double const shi48[] = { - 2.8720729351043701172, -5.0413231849670410156, 6.2442994117736816406, - -5.8483986854553222656, 3.7067542076110839844, -1.0495119094848632812, - -1.1830236911773681641, 2.1126792430877685547, -1.9094531536102294922, - 0.99913084506988525391, -0.17090806365013122559, -0.32615602016448974609, - 0.39127644896507263184, -0.26876461505889892578, 0.097676105797290802002, - -0.023473845794796943665, -}; -static double const shi44[] = { - 2.6773197650909423828, -4.8308925628662109375, 6.570110321044921875, - -7.4572014808654785156, 6.7263274192810058594, -4.8481650352478027344, - 2.0412089824676513672, 0.7006359100341796875, -2.9537565708160400391, - 4.0800385475158691406, -4.1845216751098632812, 3.3311812877655029297, - -2.1179926395416259766, 0.879302978515625, -0.031759146600961685181, - -0.42382788658142089844, 0.47882103919982910156, -0.35490813851356506348, - 0.17496839165687561035, -0.060908168554306030273, -}; -static double const shi38[] = { - 1.6335992813110351562, -2.2615492343902587891, 2.4077029228210449219, - -2.6341717243194580078, 2.1440362930297851562, -1.8153258562088012695, - 1.0816224813461303711, -0.70302653312683105469, 0.15991993248462677002, - 0.041549518704414367676, -0.29416576027870178223, 0.2518316805362701416, - -0.27766478061676025391, 0.15785403549671173096, -0.10165894031524658203, - 0.016833892092108726501, -}; -static double const shi32[] = -{ /* dmaker 32000: bestmax=4.99659 (inverted) */ -0.82118552923202515, --1.0063692331314087, -0.62341964244842529, --1.0447187423706055, -0.64532512426376343, --0.87615132331848145, -0.52219754457473755, --0.67434263229370117, -0.44954317808151245, --0.52557498216629028, -0.34567299485206604, --0.39618203043937683, -0.26791760325431824, --0.28936097025871277, -0.1883765310049057, --0.19097308814525604, -0.10431359708309174, --0.10633844882249832, -0.046832218766212463, --0.039653312414884567, -}; -static double const shi22[] = -{ /* dmaker 22050: bestmax=5.77762 (inverted) */ -0.056581053882837296, --0.56956905126571655, --0.40727734565734863, --0.33870288729667664, --0.29810553789138794, --0.19039161503314972, --0.16510021686553955, --0.13468159735202789, --0.096633769571781158, --0.081049129366874695, --0.064953058958053589, --0.054459091275930405, --0.043378707021474838, --0.03660014271736145, --0.026256965473294258, --0.018786206841468811, --0.013387725688517094, --0.0090983230620622635, --0.0026585909072309732, --0.00042083300650119781, -}; -static double const shi16[] = -{ /* dmaker 16000: bestmax=5.97128 (inverted) */ --0.37251132726669312, --0.81423574686050415, --0.55010956525802612, --0.47405767440795898, --0.32624706625938416, --0.3161766529083252, --0.2286367267370224, --0.22916607558727264, --0.19565616548061371, --0.18160104751586914, --0.15423151850700378, --0.14104481041431427, --0.11844276636838913, --0.097583092749118805, --0.076493598520755768, --0.068106919527053833, --0.041881654411554337, --0.036922425031661987, --0.019364040344953537, --0.014994367957115173, -}; -static double const shi11[] = -{ /* dmaker 11025: bestmax=5.9406 (inverted) */ --0.9264228343963623, --0.98695987462997437, --0.631156325340271, --0.51966935396194458, --0.39738872647285461, --0.35679301619529724, --0.29720726609230042, --0.26310476660728455, --0.21719355881214142, --0.18561814725399017, --0.15404847264289856, --0.12687471508979797, --0.10339745879173279, --0.083688631653785706, --0.05875682458281517, --0.046893671154975891, --0.027950936928391457, --0.020740609616041183, --0.009366452693939209, --0.0060260160826146603, -}; -static double const shi08[] = -{ /* dmaker 8000: bestmax=5.56234 (inverted) */ --1.202863335609436, --0.94103097915649414, --0.67878556251525879, --0.57650017738342285, --0.50004476308822632, --0.44349345564842224, --0.37833768129348755, --0.34028723835945129, --0.29413089156150818, --0.24994957447052002, --0.21715600788593292, --0.18792112171649933, --0.15268312394618988, --0.12135542929172516, --0.099610626697540283, --0.075273610651493073, --0.048787496984004974, --0.042586319148540497, --0.028991291299462318, --0.011869125068187714, -}; -static double const shl48[] = { - 2.3925774097442626953, -3.4350297451019287109, 3.1853709220886230469, - -1.8117271661758422852, -0.20124770700931549072, 1.4759907722473144531, - -1.7210904359817504883, 0.97746700048446655273, -0.13790138065814971924, - -0.38185903429985046387, 0.27421241998672485352, 0.066584214568138122559, - -0.35223302245140075684, 0.37672343850135803223, -0.23964276909828186035, - 0.068674825131893157959, -}; -static double const shl44[] = { - 2.0833916664123535156, -3.0418450832366943359, 3.2047898769378662109, - -2.7571926116943359375, 1.4978630542755126953, -0.3427594602108001709, - -0.71733748912811279297, 1.0737057924270629883, -1.0225815773010253906, - 0.56649994850158691406, -0.20968692004680633545, -0.065378531813621520996, - 0.10322438180446624756, -0.067442022264003753662, -0.00495197344571352005, - 0, -}; -static double const shh44[] = { - 3.0259189605712890625, -6.0268716812133789062, 9.195003509521484375, - -11.824929237365722656, 12.767142295837402344, -11.917946815490722656, - 9.1739168167114257812, -5.3712320327758789062, 1.1393624544143676758, - 2.4484779834747314453, -4.9719839096069335938, 6.0392003059387207031, - -5.9359521865844726562, 4.903278350830078125, -3.5527443885803222656, - 2.1909697055816650391, -1.1672389507293701172, 0.4903914332389831543, - -0.16519790887832641602, 0.023217858746647834778, -}; - -static const filter_t filters[] = { - {44100, fir, 5, 210, lip44, SWR_DITHER_NS_LIPSHITZ}, - {46000, fir, 9, 276, fwe44, SWR_DITHER_NS_F_WEIGHTED}, - {46000, fir, 9, 160, mew44, SWR_DITHER_NS_MODIFIED_E_WEIGHTED}, - {46000, fir, 9, 321, iew44, SWR_DITHER_NS_IMPROVED_E_WEIGHTED}, -// {48000, iir, 4, 220, ges48, SWR_DITHER_NS_GESEMANN}, -// {44100, iir, 4, 230, ges44, SWR_DITHER_NS_GESEMANN}, - {48000, fir, 16, 301, shi48, SWR_DITHER_NS_SHIBATA}, - {44100, fir, 20, 333, shi44, SWR_DITHER_NS_SHIBATA}, - {37800, fir, 16, 240, shi38, SWR_DITHER_NS_SHIBATA}, - {32000, fir, 20, 240/*TBD*/, shi32, SWR_DITHER_NS_SHIBATA}, - {22050, fir, 20, 240/*TBD*/, shi22, SWR_DITHER_NS_SHIBATA}, - {16000, fir, 20, 240/*TBD*/, shi16, SWR_DITHER_NS_SHIBATA}, - {11025, fir, 20, 240/*TBD*/, shi11, SWR_DITHER_NS_SHIBATA}, - { 8000, fir, 20, 240/*TBD*/, shi08, SWR_DITHER_NS_SHIBATA}, - {48000, fir, 16, 250, shl48, SWR_DITHER_NS_LOW_SHIBATA}, - {44100, fir, 15, 250, shl44, SWR_DITHER_NS_LOW_SHIBATA}, - {44100, fir, 20, 383, shh44, SWR_DITHER_NS_HIGH_SHIBATA}, - { 0, fir, 0, 0, NULL, SWR_DITHER_NONE}, -}; diff --git a/ffmpeg1/libswresample/rematrix.c b/ffmpeg1/libswresample/rematrix.c deleted file mode 100644 index 51658ce..0000000 --- a/ffmpeg1/libswresample/rematrix.c +++ /dev/null @@ -1,472 +0,0 @@ -/* - * Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include "swresample_internal.h" -#include "libavutil/avassert.h" -#include "libavutil/channel_layout.h" - -#define TEMPLATE_REMATRIX_FLT -#include "rematrix_template.c" -#undef TEMPLATE_REMATRIX_FLT - -#define TEMPLATE_REMATRIX_DBL -#include "rematrix_template.c" -#undef TEMPLATE_REMATRIX_DBL - -#define TEMPLATE_REMATRIX_S16 -#include "rematrix_template.c" -#undef TEMPLATE_REMATRIX_S16 - -#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 - -int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride) -{ - int nb_in, nb_out, in, out; - - if (!s || s->in_convert) // s needs to be allocated but not initialized - return AVERROR(EINVAL); - memset(s->matrix, 0, sizeof(s->matrix)); - nb_in = av_get_channel_layout_nb_channels(s->in_ch_layout); - nb_out = av_get_channel_layout_nb_channels(s->out_ch_layout); - for (out = 0; out < nb_out; out++) { - for (in = 0; in < nb_in; in++) - s->matrix[out][in] = matrix[in]; - matrix += stride; - } - s->rematrix_custom = 1; - return 0; -} - -static int even(int64_t layout){ - if(!layout) return 1; - if(layout&(layout-1)) return 1; - return 0; -} - -static int clean_layout(SwrContext *s, int64_t layout){ - if((layout & AV_CH_LAYOUT_STEREO_DOWNMIX) == AV_CH_LAYOUT_STEREO_DOWNMIX) - return AV_CH_LAYOUT_STEREO; - - if(layout && layout != AV_CH_FRONT_CENTER && !(layout&(layout-1))) { - char buf[128]; - av_get_channel_layout_string(buf, sizeof(buf), -1, layout); - av_log(s, AV_LOG_VERBOSE, "Treating %s as mono\n", buf); - return AV_CH_FRONT_CENTER; - } - - return layout; -} - -static int sane_layout(int64_t layout){ - if(!(layout & AV_CH_LAYOUT_SURROUND)) // at least 1 front speaker - return 0; - if(!even(layout & (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT))) // no asymetric front - return 0; - if(!even(layout & (AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT))) // no asymetric side - return 0; - if(!even(layout & (AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT))) - return 0; - if(!even(layout & (AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER))) - return 0; - if(av_get_channel_layout_nb_channels(layout) >= SWR_CH_MAX) - return 0; - - return 1; -} - -av_cold static int auto_matrix(SwrContext *s) -{ - int i, j, out_i; - double matrix[64][64]={{0}}; - int64_t unaccounted, in_ch_layout, out_ch_layout; - double maxcoef=0; - char buf[128]; - const int matrix_encoding = s->matrix_encoding; - - in_ch_layout = clean_layout(s, s->in_ch_layout); - if(!sane_layout(in_ch_layout)){ - av_get_channel_layout_string(buf, sizeof(buf), -1, s->in_ch_layout); - av_log(s, AV_LOG_ERROR, "Input channel layout '%s' is not supported\n", buf); - return AVERROR(EINVAL); - } - - out_ch_layout = clean_layout(s, s->out_ch_layout); - if(!sane_layout(out_ch_layout)){ - av_get_channel_layout_string(buf, sizeof(buf), -1, s->out_ch_layout); - av_log(s, AV_LOG_ERROR, "Output channel layout '%s' is not supported\n", buf); - return AVERROR(EINVAL); - } - - memset(s->matrix, 0, sizeof(s->matrix)); - for(i=0; i<64; i++){ - if(in_ch_layout & out_ch_layout & (1ULL<<i)) - matrix[i][i]= 1.0; - } - - unaccounted= in_ch_layout & ~out_ch_layout; - -//FIXME implement dolby surround -//FIXME implement full ac3 - - - if(unaccounted & AV_CH_FRONT_CENTER){ - if((out_ch_layout & AV_CH_LAYOUT_STEREO) == AV_CH_LAYOUT_STEREO){ - if(in_ch_layout & AV_CH_LAYOUT_STEREO) { - matrix[ FRONT_LEFT][FRONT_CENTER]+= s->clev; - matrix[FRONT_RIGHT][FRONT_CENTER]+= s->clev; - } else { - matrix[ FRONT_LEFT][FRONT_CENTER]+= M_SQRT1_2; - matrix[FRONT_RIGHT][FRONT_CENTER]+= M_SQRT1_2; - } - }else - av_assert0(0); - } - if(unaccounted & AV_CH_LAYOUT_STEREO){ - if(out_ch_layout & AV_CH_FRONT_CENTER){ - matrix[FRONT_CENTER][ FRONT_LEFT]+= M_SQRT1_2; - matrix[FRONT_CENTER][FRONT_RIGHT]+= M_SQRT1_2; - if(in_ch_layout & AV_CH_FRONT_CENTER) - matrix[FRONT_CENTER][ FRONT_CENTER] = s->clev*sqrt(2); - }else - av_assert0(0); - } - - if(unaccounted & AV_CH_BACK_CENTER){ - if(out_ch_layout & AV_CH_BACK_LEFT){ - matrix[ BACK_LEFT][BACK_CENTER]+= M_SQRT1_2; - matrix[BACK_RIGHT][BACK_CENTER]+= M_SQRT1_2; - }else if(out_ch_layout & AV_CH_SIDE_LEFT){ - matrix[ SIDE_LEFT][BACK_CENTER]+= M_SQRT1_2; - matrix[SIDE_RIGHT][BACK_CENTER]+= M_SQRT1_2; - }else if(out_ch_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] -= s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_CENTER] += s->slev * M_SQRT1_2; - } else { - matrix[FRONT_LEFT ][BACK_CENTER] -= s->slev; - matrix[FRONT_RIGHT][BACK_CENTER] += s->slev; - } - } else { - matrix[ FRONT_LEFT][BACK_CENTER]+= s->slev*M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_CENTER]+= s->slev*M_SQRT1_2; - } - }else if(out_ch_layout & AV_CH_FRONT_CENTER){ - matrix[ FRONT_CENTER][BACK_CENTER]+= s->slev*M_SQRT1_2; - }else - av_assert0(0); - } - if(unaccounted & AV_CH_BACK_LEFT){ - if(out_ch_layout & AV_CH_BACK_CENTER){ - matrix[BACK_CENTER][ BACK_LEFT]+= M_SQRT1_2; - matrix[BACK_CENTER][BACK_RIGHT]+= M_SQRT1_2; - }else if(out_ch_layout & AV_CH_SIDE_LEFT){ - if(in_ch_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_ch_layout & AV_CH_FRONT_LEFT){ - if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) { - matrix[FRONT_LEFT ][BACK_LEFT ] -= s->slev * M_SQRT1_2; - matrix[FRONT_LEFT ][BACK_RIGHT] -= s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_LEFT ] += s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_RIGHT] += s->slev * M_SQRT1_2; - } else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) { - matrix[FRONT_LEFT ][BACK_LEFT ] -= s->slev * SQRT3_2; - matrix[FRONT_LEFT ][BACK_RIGHT] -= s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_LEFT ] += s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][BACK_RIGHT] += s->slev * SQRT3_2; - } else { - matrix[ FRONT_LEFT][ BACK_LEFT] += s->slev; - matrix[FRONT_RIGHT][BACK_RIGHT] += s->slev; - } - }else if(out_ch_layout & AV_CH_FRONT_CENTER){ - matrix[ FRONT_CENTER][BACK_LEFT ]+= s->slev*M_SQRT1_2; - matrix[ FRONT_CENTER][BACK_RIGHT]+= s->slev*M_SQRT1_2; - }else - av_assert0(0); - } - - if(unaccounted & AV_CH_SIDE_LEFT){ - if(out_ch_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_ch_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_ch_layout & AV_CH_BACK_CENTER){ - matrix[BACK_CENTER][ SIDE_LEFT]+= M_SQRT1_2; - matrix[BACK_CENTER][SIDE_RIGHT]+= M_SQRT1_2; - }else if(out_ch_layout & AV_CH_FRONT_LEFT){ - if (matrix_encoding == AV_MATRIX_ENCODING_DOLBY) { - matrix[FRONT_LEFT ][SIDE_LEFT ] -= s->slev * M_SQRT1_2; - matrix[FRONT_LEFT ][SIDE_RIGHT] -= s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_LEFT ] += s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_RIGHT] += s->slev * M_SQRT1_2; - } else if (matrix_encoding == AV_MATRIX_ENCODING_DPLII) { - matrix[FRONT_LEFT ][SIDE_LEFT ] -= s->slev * SQRT3_2; - matrix[FRONT_LEFT ][SIDE_RIGHT] -= s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_LEFT ] += s->slev * M_SQRT1_2; - matrix[FRONT_RIGHT][SIDE_RIGHT] += s->slev * SQRT3_2; - } else { - matrix[ FRONT_LEFT][ SIDE_LEFT] += s->slev; - matrix[FRONT_RIGHT][SIDE_RIGHT] += s->slev; - } - }else if(out_ch_layout & AV_CH_FRONT_CENTER){ - matrix[ FRONT_CENTER][SIDE_LEFT ]+= s->slev*M_SQRT1_2; - matrix[ FRONT_CENTER][SIDE_RIGHT]+= s->slev*M_SQRT1_2; - }else - av_assert0(0); - } - - if(unaccounted & AV_CH_FRONT_LEFT_OF_CENTER){ - if(out_ch_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_ch_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 - av_assert0(0); - } - /* mix LFE into front left/right or center */ - if (unaccounted & AV_CH_LOW_FREQUENCY) { - if (out_ch_layout & AV_CH_FRONT_CENTER) { - matrix[FRONT_CENTER][LOW_FREQUENCY] += s->lfe_mix_level; - } else if (out_ch_layout & AV_CH_FRONT_LEFT) { - matrix[FRONT_LEFT ][LOW_FREQUENCY] += s->lfe_mix_level * M_SQRT1_2; - matrix[FRONT_RIGHT][LOW_FREQUENCY] += s->lfe_mix_level * M_SQRT1_2; - } else - av_assert0(0); - } - - for(out_i=i=0; i<64; i++){ - double sum=0; - int in_i=0; - for(j=0; j<64; j++){ - s->matrix[out_i][in_i]= matrix[i][j]; - if(matrix[i][j]){ - sum += fabs(matrix[i][j]); - } - if(in_ch_layout & (1ULL<<j)) - in_i++; - } - maxcoef= FFMAX(maxcoef, sum); - if(out_ch_layout & (1ULL<<i)) - out_i++; - } - if(s->rematrix_volume < 0) - maxcoef = -s->rematrix_volume; - - if(( av_get_packed_sample_fmt(s->out_sample_fmt) < AV_SAMPLE_FMT_FLT - || av_get_packed_sample_fmt(s->int_sample_fmt) < AV_SAMPLE_FMT_FLT) && maxcoef > 1.0){ - for(i=0; i<SWR_CH_MAX; i++) - for(j=0; j<SWR_CH_MAX; j++){ - s->matrix[i][j] /= maxcoef; - } - } - - if(s->rematrix_volume > 0){ - for(i=0; i<SWR_CH_MAX; i++) - for(j=0; j<SWR_CH_MAX; j++){ - s->matrix[i][j] *= s->rematrix_volume; - } - } - - for(i=0; i<av_get_channel_layout_nb_channels(out_ch_layout); i++){ - for(j=0; j<av_get_channel_layout_nb_channels(in_ch_layout); j++){ - av_log(NULL, AV_LOG_DEBUG, "%f ", s->matrix[i][j]); - } - av_log(NULL, AV_LOG_DEBUG, "\n"); - } - return 0; -} - -av_cold int swri_rematrix_init(SwrContext *s){ - int i, j; - int nb_in = av_get_channel_layout_nb_channels(s->in_ch_layout); - int nb_out = av_get_channel_layout_nb_channels(s->out_ch_layout); - - s->mix_any_f = NULL; - - if (!s->rematrix_custom) { - int r = auto_matrix(s); - if (r) - return r; - } - if (s->midbuf.fmt == AV_SAMPLE_FMT_S16P){ - s->native_matrix = av_mallocz(nb_in * nb_out * sizeof(int)); - s->native_one = av_mallocz(sizeof(int)); - for (i = 0; i < nb_out; i++) - for (j = 0; j < nb_in; j++) - ((int*)s->native_matrix)[i * nb_in + j] = lrintf(s->matrix[i][j] * 32768); - *((int*)s->native_one) = 32768; - s->mix_1_1_f = (mix_1_1_func_type*)copy_s16; - s->mix_2_1_f = (mix_2_1_func_type*)sum2_s16; - s->mix_any_f = (mix_any_func_type*)get_mix_any_func_s16(s); - }else if(s->midbuf.fmt == AV_SAMPLE_FMT_FLTP){ - s->native_matrix = av_mallocz(nb_in * nb_out * sizeof(float)); - s->native_one = av_mallocz(sizeof(float)); - for (i = 0; i < nb_out; i++) - for (j = 0; j < nb_in; j++) - ((float*)s->native_matrix)[i * nb_in + j] = s->matrix[i][j]; - *((float*)s->native_one) = 1.0; - s->mix_1_1_f = (mix_1_1_func_type*)copy_float; - s->mix_2_1_f = (mix_2_1_func_type*)sum2_float; - s->mix_any_f = (mix_any_func_type*)get_mix_any_func_float(s); - }else if(s->midbuf.fmt == AV_SAMPLE_FMT_DBLP){ - s->native_matrix = av_mallocz(nb_in * nb_out * sizeof(double)); - s->native_one = av_mallocz(sizeof(double)); - for (i = 0; i < nb_out; i++) - for (j = 0; j < nb_in; j++) - ((double*)s->native_matrix)[i * nb_in + j] = s->matrix[i][j]; - *((double*)s->native_one) = 1.0; - s->mix_1_1_f = (mix_1_1_func_type*)copy_double; - s->mix_2_1_f = (mix_2_1_func_type*)sum2_double; - s->mix_any_f = (mix_any_func_type*)get_mix_any_func_double(s); - }else - av_assert0(0); - //FIXME quantize for integeres - for (i = 0; i < SWR_CH_MAX; i++) { - int ch_in=0; - for (j = 0; j < SWR_CH_MAX; j++) { - s->matrix32[i][j]= lrintf(s->matrix[i][j] * 32768); - if(s->matrix[i][j]) - s->matrix_ch[i][++ch_in]= j; - } - s->matrix_ch[i][0]= ch_in; - } - - if(HAVE_YASM && HAVE_MMX) swri_rematrix_init_x86(s); - - return 0; -} - -av_cold void swri_rematrix_free(SwrContext *s){ - av_freep(&s->native_matrix); - av_freep(&s->native_one); - av_freep(&s->native_simd_matrix); -} - -int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy){ - int out_i, in_i, i, j; - int len1 = 0; - int off = 0; - - if(s->mix_any_f) { - s->mix_any_f(out->ch, (const uint8_t **)in->ch, s->native_matrix, len); - return 0; - } - - if(s->mix_2_1_simd || s->mix_1_1_simd){ - len1= len&~15; - off = len1 * out->bps; - } - - av_assert0(out->ch_count == av_get_channel_layout_nb_channels(s->out_ch_layout)); - av_assert0(in ->ch_count == av_get_channel_layout_nb_channels(s-> in_ch_layout)); - - for(out_i=0; out_i<out->ch_count; out_i++){ - switch(s->matrix_ch[out_i][0]){ - case 0: - if(mustcopy) - memset(out->ch[out_i], 0, len * av_get_bytes_per_sample(s->int_sample_fmt)); - break; - case 1: - in_i= s->matrix_ch[out_i][1]; - if(s->matrix[out_i][in_i]!=1.0){ - if(s->mix_1_1_simd && len1) - s->mix_1_1_simd(out->ch[out_i] , in->ch[in_i] , s->native_simd_matrix, in->ch_count*out_i + in_i, len1); - if(len != len1) - s->mix_1_1_f (out->ch[out_i]+off, in->ch[in_i]+off, s->native_matrix, in->ch_count*out_i + in_i, len-len1); - }else if(mustcopy){ - memcpy(out->ch[out_i], in->ch[in_i], len*out->bps); - }else{ - out->ch[out_i]= in->ch[in_i]; - } - break; - case 2: { - int in_i1 = s->matrix_ch[out_i][1]; - int in_i2 = s->matrix_ch[out_i][2]; - if(s->mix_2_1_simd && len1) - s->mix_2_1_simd(out->ch[out_i] , in->ch[in_i1] , in->ch[in_i2] , s->native_simd_matrix, in->ch_count*out_i + in_i1, in->ch_count*out_i + in_i2, len1); - else - s->mix_2_1_f (out->ch[out_i] , in->ch[in_i1] , in->ch[in_i2] , s->native_matrix, in->ch_count*out_i + in_i1, in->ch_count*out_i + in_i2, len1); - if(len != len1) - s->mix_2_1_f (out->ch[out_i]+off, in->ch[in_i1]+off, in->ch[in_i2]+off, s->native_matrix, in->ch_count*out_i + in_i1, in->ch_count*out_i + in_i2, len-len1); - break;} - default: - if(s->int_sample_fmt == AV_SAMPLE_FMT_FLTP){ - for(i=0; i<len; i++){ - float v=0; - for(j=0; j<s->matrix_ch[out_i][0]; j++){ - in_i= s->matrix_ch[out_i][1+j]; - v+= ((float*)in->ch[in_i])[i] * s->matrix[out_i][in_i]; - } - ((float*)out->ch[out_i])[i]= v; - } - }else if(s->int_sample_fmt == AV_SAMPLE_FMT_DBLP){ - for(i=0; i<len; i++){ - double v=0; - for(j=0; j<s->matrix_ch[out_i][0]; j++){ - in_i= s->matrix_ch[out_i][1+j]; - v+= ((double*)in->ch[in_i])[i] * s->matrix[out_i][in_i]; - } - ((double*)out->ch[out_i])[i]= v; - } - }else{ - for(i=0; i<len; i++){ - int v=0; - for(j=0; j<s->matrix_ch[out_i][0]; j++){ - in_i= s->matrix_ch[out_i][1+j]; - v+= ((int16_t*)in->ch[in_i])[i] * s->matrix32[out_i][in_i]; - } - ((int16_t*)out->ch[out_i])[i]= (v + 16384)>>15; - } - } - } - } - return 0; -} diff --git a/ffmpeg1/libswresample/rematrix_template.c b/ffmpeg1/libswresample/rematrix_template.c deleted file mode 100644 index b8ca901..0000000 --- a/ffmpeg1/libswresample/rematrix_template.c +++ /dev/null @@ -1,100 +0,0 @@ -/* - * Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#if defined(TEMPLATE_REMATRIX_FLT) -# define R(x) x -# define SAMPLE float -# define COEFF float -# define INTER float -# define RENAME(x) x ## _float -#elif defined(TEMPLATE_REMATRIX_DBL) -# define R(x) x -# define SAMPLE double -# define COEFF double -# define INTER double -# define RENAME(x) x ## _double -#elif defined(TEMPLATE_REMATRIX_S16) -# define R(x) (((x) + 16384)>>15) -# define SAMPLE int16_t -# define COEFF int -# define INTER int -# define RENAME(x) x ## _s16 -#endif - -typedef void (RENAME(mix_any_func_type))(SAMPLE **out, const SAMPLE **in1, COEFF *coeffp, integer len); - -static void RENAME(sum2)(SAMPLE *out, const SAMPLE *in1, const SAMPLE *in2, COEFF *coeffp, integer index1, integer index2, integer len){ - int i; - COEFF coeff1 = coeffp[index1]; - COEFF coeff2 = coeffp[index2]; - - for(i=0; i<len; i++) - out[i] = R(coeff1*in1[i] + coeff2*in2[i]); -} - -static void RENAME(copy)(SAMPLE *out, const SAMPLE *in, COEFF *coeffp, integer index, integer len){ - int i; - COEFF coeff = coeffp[index]; - for(i=0; i<len; i++) - out[i] = R(coeff*in[i]); -} - -static void RENAME(mix6to2)(SAMPLE **out, const SAMPLE **in, COEFF *coeffp, integer len){ - int i; - - for(i=0; i<len; i++) { - INTER t = in[2][i]*coeffp[0*6+2] + in[3][i]*coeffp[0*6+3]; - out[0][i] = R(t + in[0][i]*coeffp[0*6+0] + in[4][i]*coeffp[0*6+4]); - out[1][i] = R(t + in[1][i]*coeffp[1*6+1] + in[5][i]*coeffp[1*6+5]); - } -} - -static void RENAME(mix8to2)(SAMPLE **out, const SAMPLE **in, COEFF *coeffp, integer len){ - int i; - - for(i=0; i<len; i++) { - INTER t = in[2][i]*coeffp[0*8+2] + in[3][i]*coeffp[0*8+3]; - out[0][i] = R(t + in[0][i]*coeffp[0*8+0] + in[4][i]*coeffp[0*8+4] + in[6][i]*coeffp[0*8+6]); - out[1][i] = R(t + in[1][i]*coeffp[1*8+1] + in[5][i]*coeffp[1*8+5] + in[7][i]*coeffp[1*8+7]); - } -} - -static RENAME(mix_any_func_type) *RENAME(get_mix_any_func)(SwrContext *s){ - if( s->out_ch_layout == AV_CH_LAYOUT_STEREO && (s->in_ch_layout == AV_CH_LAYOUT_5POINT1 || s->in_ch_layout == AV_CH_LAYOUT_5POINT1_BACK) - && s->matrix[0][2] == s->matrix[1][2] && s->matrix[0][3] == s->matrix[1][3] - && !s->matrix[0][1] && !s->matrix[0][5] && !s->matrix[1][0] && !s->matrix[1][4] - ) - return RENAME(mix6to2); - - if( s->out_ch_layout == AV_CH_LAYOUT_STEREO && s->in_ch_layout == AV_CH_LAYOUT_7POINT1 - && s->matrix[0][2] == s->matrix[1][2] && s->matrix[0][3] == s->matrix[1][3] - && !s->matrix[0][1] && !s->matrix[0][5] && !s->matrix[1][0] && !s->matrix[1][4] - && !s->matrix[0][7] && !s->matrix[1][6] - ) - return RENAME(mix8to2); - - return NULL; -} - -#undef R -#undef SAMPLE -#undef COEFF -#undef INTER -#undef RENAME diff --git a/ffmpeg1/libswresample/resample.c b/ffmpeg1/libswresample/resample.c deleted file mode 100644 index fb9da7c..0000000 --- a/ffmpeg1/libswresample/resample.c +++ /dev/null @@ -1,372 +0,0 @@ -/* - * audio resampling - * Copyright (c) 2004-2012 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 - */ - -/** - * @file - * audio resampling - * @author Michael Niedermayer <michaelni@gmx.at> - */ - -#include "libavutil/log.h" -#include "libavutil/avassert.h" -#include "swresample_internal.h" - - -typedef struct ResampleContext { - const AVClass *av_class; - uint8_t *filter_bank; - int filter_length; - int filter_alloc; - int ideal_dst_incr; - int dst_incr; - int index; - int frac; - int src_incr; - int compensation_distance; - int phase_shift; - int phase_mask; - int linear; - enum SwrFilterType filter_type; - int kaiser_beta; - double factor; - enum AVSampleFormat format; - int felem_size; - int filter_shift; -} ResampleContext; - -/** - * 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; - static const double inv[100]={ - 1.0/( 1* 1), 1.0/( 2* 2), 1.0/( 3* 3), 1.0/( 4* 4), 1.0/( 5* 5), 1.0/( 6* 6), 1.0/( 7* 7), 1.0/( 8* 8), 1.0/( 9* 9), 1.0/(10*10), - 1.0/(11*11), 1.0/(12*12), 1.0/(13*13), 1.0/(14*14), 1.0/(15*15), 1.0/(16*16), 1.0/(17*17), 1.0/(18*18), 1.0/(19*19), 1.0/(20*20), - 1.0/(21*21), 1.0/(22*22), 1.0/(23*23), 1.0/(24*24), 1.0/(25*25), 1.0/(26*26), 1.0/(27*27), 1.0/(28*28), 1.0/(29*29), 1.0/(30*30), - 1.0/(31*31), 1.0/(32*32), 1.0/(33*33), 1.0/(34*34), 1.0/(35*35), 1.0/(36*36), 1.0/(37*37), 1.0/(38*38), 1.0/(39*39), 1.0/(40*40), - 1.0/(41*41), 1.0/(42*42), 1.0/(43*43), 1.0/(44*44), 1.0/(45*45), 1.0/(46*46), 1.0/(47*47), 1.0/(48*48), 1.0/(49*49), 1.0/(50*50), - 1.0/(51*51), 1.0/(52*52), 1.0/(53*53), 1.0/(54*54), 1.0/(55*55), 1.0/(56*56), 1.0/(57*57), 1.0/(58*58), 1.0/(59*59), 1.0/(60*60), - 1.0/(61*61), 1.0/(62*62), 1.0/(63*63), 1.0/(64*64), 1.0/(65*65), 1.0/(66*66), 1.0/(67*67), 1.0/(68*68), 1.0/(69*69), 1.0/(70*70), - 1.0/(71*71), 1.0/(72*72), 1.0/(73*73), 1.0/(74*74), 1.0/(75*75), 1.0/(76*76), 1.0/(77*77), 1.0/(78*78), 1.0/(79*79), 1.0/(80*80), - 1.0/(81*81), 1.0/(82*82), 1.0/(83*83), 1.0/(84*84), 1.0/(85*85), 1.0/(86*86), 1.0/(87*87), 1.0/(88*88), 1.0/(89*89), 1.0/(90*90), - 1.0/(91*91), 1.0/(92*92), 1.0/(93*93), 1.0/(94*94), 1.0/(95*95), 1.0/(96*96), 1.0/(97*97), 1.0/(98*98), 1.0/(99*99), 1.0/(10000) - }; - - x= x*x/4; - for(i=0; v != lastv; i++){ - lastv=v; - t *= x*inv[i]; - v += t; - av_assert2(i<99); - } - return v; -} - -/** - * builds a polyphase filterbank. - * @param factor resampling factor - * @param scale wanted sum of coefficients for each filter - * @param filter_type filter type - * @param kaiser_beta kaiser window beta - * @return 0 on success, negative on error - */ -static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, - int filter_type, int kaiser_beta){ - int ph, i; - double x, y, w; - double *tab = av_malloc(tap_count * sizeof(*tab)); - const int center= (tap_count-1)/2; - - if (!tab) - return AVERROR(ENOMEM); - - /* if upsampling, only need to interpolate, no filter */ - if (factor > 1.0) - factor = 1.0; - - 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(filter_type){ - case SWR_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 SWR_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 SWR_FILTER_TYPE_KAISER: - w = 2.0*x / (factor*tap_count*M_PI); - y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); - break; - default: - av_assert0(0); - } - - tab[i] = y; - norm += y; - } - - /* normalize so that an uniform color remains the same */ - switch(c->format){ - case AV_SAMPLE_FMT_S16P: - for(i=0;i<tap_count;i++) - ((int16_t*)filter)[ph * alloc + i] = av_clip(lrintf(tab[i] * scale / norm), INT16_MIN, INT16_MAX); - break; - case AV_SAMPLE_FMT_S32P: - for(i=0;i<tap_count;i++) - ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm)); - break; - case AV_SAMPLE_FMT_FLTP: - for(i=0;i<tap_count;i++) - ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm; - break; - case AV_SAMPLE_FMT_DBLP: - for(i=0;i<tap_count;i++) - ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm; - break; - } - } -#if 0 - { -#define LEN 1024 - int j,k; - double sine[LEN + tap_count]; - double filtered[LEN]; - double maxff=-2, minff=2, maxsf=-2, minsf=2; - for(i=0; i<LEN; i++){ - double ss=0, sf=0, ff=0; - for(j=0; j<LEN+tap_count; j++) - sine[j]= cos(i*j*M_PI/LEN); - for(j=0; j<LEN; j++){ - double sum=0; - ph=0; - for(k=0; k<tap_count; k++) - sum += filter[ph * tap_count + k] * sine[k+j]; - filtered[j]= sum / (1<<FILTER_SHIFT); - ss+= sine[j + center] * sine[j + center]; - ff+= filtered[j] * filtered[j]; - sf+= sine[j + center] * filtered[j]; - } - ss= sqrt(2*ss/LEN); - ff= sqrt(2*ff/LEN); - sf= 2*sf/LEN; - maxff= FFMAX(maxff, ff); - minff= FFMIN(minff, ff); - maxsf= FFMAX(maxsf, sf); - minsf= FFMIN(minsf, sf); - if(i%11==0){ - av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf); - minff=minsf= 2; - maxff=maxsf= -2; - } - } - } -#endif - - av_free(tab); - return 0; -} - -static ResampleContext *resample_init(ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, - double cutoff0, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, - double precision, int cheby){ - double cutoff = cutoff0? cutoff0 : 0.97; - double factor= FFMIN(out_rate * cutoff / in_rate, 1.0); - int phase_count= 1<<phase_shift; - - if (!c || c->phase_shift != phase_shift || c->linear!=linear || c->factor != factor - || c->filter_length != FFMAX((int)ceil(filter_size/factor), 1) || c->format != format - || c->filter_type != filter_type || c->kaiser_beta != kaiser_beta) { - c = av_mallocz(sizeof(*c)); - if (!c) - return NULL; - - c->format= format; - - c->felem_size= av_get_bytes_per_sample(c->format); - - switch(c->format){ - case AV_SAMPLE_FMT_S16P: - c->filter_shift = 15; - break; - case AV_SAMPLE_FMT_S32P: - c->filter_shift = 30; - break; - case AV_SAMPLE_FMT_FLTP: - case AV_SAMPLE_FMT_DBLP: - c->filter_shift = 0; - break; - default: - av_log(NULL, AV_LOG_ERROR, "Unsupported sample format\n"); - av_assert0(0); - } - - c->phase_shift = phase_shift; - c->phase_mask = phase_count - 1; - c->linear = linear; - c->factor = factor; - c->filter_length = FFMAX((int)ceil(filter_size/factor), 1); - c->filter_alloc = FFALIGN(c->filter_length, 8); - c->filter_bank = av_calloc(c->filter_alloc, (phase_count+1)*c->felem_size); - c->filter_type = filter_type; - c->kaiser_beta = kaiser_beta; - if (!c->filter_bank) - goto error; - if (build_filter(c, (void*)c->filter_bank, factor, c->filter_length, c->filter_alloc, phase_count, 1<<c->filter_shift, filter_type, kaiser_beta)) - goto error; - memcpy(c->filter_bank + (c->filter_alloc*phase_count+1)*c->felem_size, c->filter_bank, (c->filter_alloc-1)*c->felem_size); - memcpy(c->filter_bank + (c->filter_alloc*phase_count )*c->felem_size, c->filter_bank + (c->filter_alloc - 1)*c->felem_size, c->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->index= -phase_count*((c->filter_length-1)/2); - c->frac= 0; - - return c; -error: - av_free(c->filter_bank); - av_free(c); - return NULL; -} - -static void resample_free(ResampleContext **c){ - if(!*c) - return; - av_freep(&(*c)->filter_bank); - av_freep(c); -} - -static int set_compensation(ResampleContext *c, int sample_delta, int compensation_distance){ - 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; -} - -#define TEMPLATE_RESAMPLE_S16 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S16 - -#define TEMPLATE_RESAMPLE_S32 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S32 - -#define TEMPLATE_RESAMPLE_FLT -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_FLT - -#define TEMPLATE_RESAMPLE_DBL -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_DBL - -// XXX FIXME the whole C loop should be written in asm so this x86 specific code here isnt needed -#if HAVE_MMXEXT_INLINE - -#include "x86/resample_mmx.h" - -#define TEMPLATE_RESAMPLE_S16_MMX2 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S16_MMX2 - -#if HAVE_SSSE3_INLINE -#define TEMPLATE_RESAMPLE_S16_SSSE3 -#include "resample_template.c" -#undef TEMPLATE_RESAMPLE_S16_SSSE3 -#endif - -#endif // HAVE_MMXEXT_INLINE - -static int multiple_resample(ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed){ - int i, ret= -1; - int av_unused mm_flags = av_get_cpu_flags(); - int need_emms= 0; - - for(i=0; i<dst->ch_count; i++){ -#if HAVE_MMXEXT_INLINE -#if HAVE_SSSE3_INLINE - if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_SSSE3)) ret= swri_resample_int16_ssse3(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); - else -#endif - if(c->format == AV_SAMPLE_FMT_S16P && (mm_flags&AV_CPU_FLAG_MMX2 )){ - ret= swri_resample_int16_mmx2 (c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); - need_emms= 1; - } else -#endif - if(c->format == AV_SAMPLE_FMT_S16P) ret= swri_resample_int16(c, (int16_t*)dst->ch[i], (const int16_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); - else if(c->format == AV_SAMPLE_FMT_S32P) ret= swri_resample_int32(c, (int32_t*)dst->ch[i], (const int32_t*)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); - else if(c->format == AV_SAMPLE_FMT_FLTP) ret= swri_resample_float(c, (float *)dst->ch[i], (const float *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); - else if(c->format == AV_SAMPLE_FMT_DBLP) ret= swri_resample_double(c,(double *)dst->ch[i], (const double *)src->ch[i], consumed, src_size, dst_size, i+1==dst->ch_count); - } - if(need_emms) - emms_c(); - return ret; -} - -static int64_t get_delay(struct SwrContext *s, int64_t base){ - ResampleContext *c = s->resample; - int64_t num = s->in_buffer_count - (c->filter_length-1)/2; - num <<= c->phase_shift; - num -= c->index; - num *= c->src_incr; - num -= c->frac; - return av_rescale(num, base, s->in_sample_rate*(int64_t)c->src_incr << c->phase_shift); -} - -static int resample_flush(struct SwrContext *s) { - AudioData *a= &s->in_buffer; - int i, j, ret; - if((ret = swri_realloc_audio(a, s->in_buffer_index + 2*s->in_buffer_count)) < 0) - return ret; - av_assert0(a->planar); - for(i=0; i<a->ch_count; i++){ - for(j=0; j<s->in_buffer_count; j++){ - memcpy(a->ch[i] + (s->in_buffer_index+s->in_buffer_count+j )*a->bps, - a->ch[i] + (s->in_buffer_index+s->in_buffer_count-j-1)*a->bps, a->bps); - } - } - s->in_buffer_count += (s->in_buffer_count+1)/2; - return 0; -} - -struct Resampler const swri_resampler={ - resample_init, - resample_free, - multiple_resample, - resample_flush, - set_compensation, - get_delay, -}; diff --git a/ffmpeg1/libswresample/resample_template.c b/ffmpeg1/libswresample/resample_template.c deleted file mode 100644 index 5bc12bc..0000000 --- a/ffmpeg1/libswresample/resample_template.c +++ /dev/null @@ -1,211 +0,0 @@ -/* - * audio resampling - * Copyright (c) 2004-2012 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 - */ - -/** - * @file - * audio resampling - * @author Michael Niedermayer <michaelni@gmx.at> - */ - -#if defined(TEMPLATE_RESAMPLE_DBL) -# define RENAME(N) N ## _double -# define FILTER_SHIFT 0 -# define DELEM double -# define FELEM double -# define FELEM2 double -# define FELEML double -# define OUT(d, v) d = v - -#elif defined(TEMPLATE_RESAMPLE_FLT) -# define RENAME(N) N ## _float -# define FILTER_SHIFT 0 -# define DELEM float -# define FELEM float -# define FELEM2 float -# define FELEML float -# define OUT(d, v) d = v - -#elif defined(TEMPLATE_RESAMPLE_S32) -# define RENAME(N) N ## _int32 -# define FILTER_SHIFT 30 -# define DELEM int32_t -# define FELEM int32_t -# define FELEM2 int64_t -# define FELEML int64_t -# define FELEM_MAX INT32_MAX -# define FELEM_MIN INT32_MIN -# define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\ - d = (uint64_t)(v + 0x80000000) > 0xFFFFFFFF ? (v>>63) ^ 0x7FFFFFFF : v - -#elif defined(TEMPLATE_RESAMPLE_S16) \ - || defined(TEMPLATE_RESAMPLE_S16_MMX2) \ - || defined(TEMPLATE_RESAMPLE_S16_SSSE3) - -# define FILTER_SHIFT 15 -# define DELEM int16_t -# define FELEM int16_t -# define FELEM2 int32_t -# define FELEML int64_t -# define FELEM_MAX INT16_MAX -# define FELEM_MIN INT16_MIN -# define OUT(d, v) v = (v + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;\ - d = (unsigned)(v + 32768) > 65535 ? (v>>31) ^ 32767 : v - -# if defined(TEMPLATE_RESAMPLE_S16) -# define RENAME(N) N ## _int16 -# elif defined(TEMPLATE_RESAMPLE_S16_MMX2) -# define COMMON_CORE COMMON_CORE_INT16_MMX2 -# define RENAME(N) N ## _int16_mmx2 -# elif defined(TEMPLATE_RESAMPLE_S16_SSSE3) -# define COMMON_CORE COMMON_CORE_INT16_SSSE3 -# define RENAME(N) N ## _int16_ssse3 -# endif - -#endif - -int RENAME(swri_resample)(ResampleContext *c, DELEM *dst, const DELEM *src, int *consumed, int src_size, int dst_size, int update_ctx){ - int dst_index, i; - 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; - - av_assert1(c->filter_shift == FILTER_SHIFT); - av_assert1(c->felem_size == sizeof(FELEM)); - - if(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){ - int64_t index2= ((int64_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); - - for(dst_index=0; dst_index < dst_size; dst_index++){ - dst[dst_index] = src[index2>>32]; - index2 += incr; - } - 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; - av_assert2(index >= 0); - *consumed= index >> c->phase_shift; - index &= c->phase_mask; - }else if(compensation_distance == 0 && !c->linear && index >= 0){ - int sample_index = 0; - for(dst_index=0; dst_index < dst_size; dst_index++){ - FELEM *filter; - sample_index += index >> c->phase_shift; - index &= c->phase_mask; - filter= ((FELEM*)c->filter_bank) + c->filter_alloc*index; - - if(sample_index + c->filter_length > src_size){ - break; - }else{ -#ifdef COMMON_CORE - COMMON_CORE -#else - FELEM2 val=0; - for(i=0; i<c->filter_length; i++){ - val += src[sample_index + i] * (FELEM2)filter[i]; - } - OUT(dst[dst_index], val); -#endif - } - - frac += dst_incr_frac; - index += dst_incr; - if(frac >= c->src_incr){ - frac -= c->src_incr; - index++; - } - } - *consumed = sample_index; - }else{ - int sample_index = 0; - for(dst_index=0; dst_index < dst_size; dst_index++){ - FELEM *filter; - FELEM2 val=0; - - sample_index += index >> c->phase_shift; - index &= c->phase_mask; - filter = ((FELEM*)c->filter_bank) + c->filter_alloc*index; - - if(sample_index + c->filter_length > src_size || -sample_index >= src_size){ - break; - }else if(sample_index < 0){ - for(i=0; i<c->filter_length; i++) - val += src[FFABS(sample_index + i)] * (FELEM2)filter[i]; - }else if(c->linear){ - 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_alloc]; - } - val+=(v2-val)*(FELEML)frac / c->src_incr; - }else{ - for(i=0; i<c->filter_length; i++){ - val += src[sample_index + i] * (FELEM2)filter[i]; - } - } - - OUT(dst[dst_index], val); - - 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; - } - } - *consumed= FFMAX(sample_index, 0); - index += FFMIN(sample_index, 0) << c->phase_shift; - - if(compensation_distance){ - compensation_distance -= dst_index; - av_assert1(compensation_distance > 0); - } - } - - if(update_ctx){ - 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; -} - -#undef COMMON_CORE -#undef RENAME -#undef FILTER_SHIFT -#undef DELEM -#undef FELEM -#undef FELEM2 -#undef FELEML -#undef FELEM_MAX -#undef FELEM_MIN -#undef OUT diff --git a/ffmpeg1/libswresample/soxr_resample.c b/ffmpeg1/libswresample/soxr_resample.c deleted file mode 100644 index 4c000db..0000000 --- a/ffmpeg1/libswresample/soxr_resample.c +++ /dev/null @@ -1,93 +0,0 @@ -/* - * audio resampling with soxr - * Copyright (c) 2012 Rob Sykes <robs@users.sourceforge.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 - */ - -/** - * @file - * audio resampling with soxr - */ - -#include "libavutil/log.h" -#include "swresample_internal.h" - -#include <soxr.h> - -static struct ResampleContext *create(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, - double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby){ - soxr_error_t error; - - soxr_datatype_t type = - format == AV_SAMPLE_FMT_S16P? SOXR_INT16_S : - format == AV_SAMPLE_FMT_S16 ? SOXR_INT16_I : - format == AV_SAMPLE_FMT_S32P? SOXR_INT32_S : - format == AV_SAMPLE_FMT_S32 ? SOXR_INT32_I : - format == AV_SAMPLE_FMT_FLTP? SOXR_FLOAT32_S : - format == AV_SAMPLE_FMT_FLT ? SOXR_FLOAT32_I : - format == AV_SAMPLE_FMT_DBLP? SOXR_FLOAT64_S : - format == AV_SAMPLE_FMT_DBL ? SOXR_FLOAT64_I : (soxr_datatype_t)-1; - - soxr_io_spec_t io_spec = soxr_io_spec(type, type); - - soxr_quality_spec_t q_spec = soxr_quality_spec((int)((precision-2)/4), (SOXR_HI_PREC_CLOCK|SOXR_ROLLOFF_NONE)*!!cheby); - q_spec.precision = linear? 0 : precision; -#if !defined SOXR_VERSION /* Deprecated @ March 2013: */ - q_spec.bw_pc = cutoff? FFMAX(FFMIN(cutoff,.995),.8)*100 : q_spec.bw_pc; -#else - q_spec.passband_end = cutoff? FFMAX(FFMIN(cutoff,.995),.8) : q_spec.passband_end; -#endif - - soxr_delete((soxr_t)c); - c = (struct ResampleContext *) - soxr_create(in_rate, out_rate, 0, &error, &io_spec, &q_spec, 0); - if (!c) - av_log(NULL, AV_LOG_ERROR, "soxr_create: %s\n", error); - return c; -} - -static void destroy(struct ResampleContext * *c){ - soxr_delete((soxr_t)*c); - *c = NULL; -} - -static int flush(struct SwrContext *s){ - soxr_process((soxr_t)s->resample, NULL, 0, NULL, NULL, 0, NULL); - return 0; -} - -static int process( - struct ResampleContext * c, AudioData *dst, int dst_size, - AudioData *src, int src_size, int *consumed){ - size_t idone, odone; - soxr_error_t error = soxr_set_error((soxr_t)c, soxr_set_num_channels((soxr_t)c, src->ch_count)); - error = soxr_process((soxr_t)c, src->ch, (size_t)src_size, - &idone, dst->ch, (size_t)dst_size, &odone); - *consumed = (int)idone; - return error? -1 : odone; -} - -static int64_t get_delay(struct SwrContext *s, int64_t base){ - double delay_s = soxr_delay((soxr_t)s->resample) / s->out_sample_rate; - return (int64_t)(delay_s * base + .5); -} - -struct Resampler const soxr_resampler={ - create, destroy, process, flush, NULL /* set_compensation */, get_delay, -}; - diff --git a/ffmpeg1/libswresample/swresample-test.c b/ffmpeg1/libswresample/swresample-test.c deleted file mode 100644 index 379d385..0000000 --- a/ffmpeg1/libswresample/swresample-test.c +++ /dev/null @@ -1,414 +0,0 @@ -/* - * Copyright (C) 2011-2012 Michael Niedermayer (michaelni@gmx.at) - * Copyright (c) 2002 Fabrice Bellard - * - * This file is part of libswresample - * - * libswresample is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * libswresample 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 General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include "libavutil/avassert.h" -#include "libavutil/channel_layout.h" -#include "libavutil/common.h" -#include "libavutil/opt.h" -#include "swresample.h" - -#undef time -#include "time.h" -#undef fprintf - -#define SAMPLES 1000 - -#define ASSERT_LEVEL 2 - -static double get(uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f){ - const uint8_t *p; - if(av_sample_fmt_is_planar(f)){ - f= av_get_alt_sample_fmt(f, 0); - p= a[ch]; - }else{ - p= a[0]; - index= ch + index*ch_count; - } - - switch(f){ - case AV_SAMPLE_FMT_U8 : return ((const uint8_t*)p)[index]/127.0-1.0; - case AV_SAMPLE_FMT_S16: return ((const int16_t*)p)[index]/32767.0; - case AV_SAMPLE_FMT_S32: return ((const int32_t*)p)[index]/2147483647.0; - case AV_SAMPLE_FMT_FLT: return ((const float *)p)[index]; - case AV_SAMPLE_FMT_DBL: return ((const double *)p)[index]; - default: av_assert0(0); - } -} - -static void set(uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f, double v){ - uint8_t *p; - if(av_sample_fmt_is_planar(f)){ - f= av_get_alt_sample_fmt(f, 0); - p= a[ch]; - }else{ - p= a[0]; - index= ch + index*ch_count; - } - switch(f){ - case AV_SAMPLE_FMT_U8 : ((uint8_t*)p)[index]= av_clip_uint8 (lrint((v+1.0)*127)); break; - case AV_SAMPLE_FMT_S16: ((int16_t*)p)[index]= av_clip_int16 (lrint(v*32767)); break; - case AV_SAMPLE_FMT_S32: ((int32_t*)p)[index]= av_clipl_int32(llrint(v*2147483647)); break; - case AV_SAMPLE_FMT_FLT: ((float *)p)[index]= v; break; - case AV_SAMPLE_FMT_DBL: ((double *)p)[index]= v; break; - default: av_assert2(0); - } -} - -static void shift(uint8_t *a[], int index, int ch_count, enum AVSampleFormat f){ - int ch; - - if(av_sample_fmt_is_planar(f)){ - f= av_get_alt_sample_fmt(f, 0); - for(ch= 0; ch<ch_count; ch++) - a[ch] += index*av_get_bytes_per_sample(f); - }else{ - a[0] += index*ch_count*av_get_bytes_per_sample(f); - } -} - -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[] = { - 8000, - 11025, - 16000, - 22050, - 32000, - 48000, -}; - -uint64_t layouts[]={ - AV_CH_LAYOUT_MONO , - AV_CH_LAYOUT_STEREO , - AV_CH_LAYOUT_2_1 , - AV_CH_LAYOUT_SURROUND , - AV_CH_LAYOUT_4POINT0 , - AV_CH_LAYOUT_2_2 , - AV_CH_LAYOUT_QUAD , - AV_CH_LAYOUT_5POINT0 , - AV_CH_LAYOUT_5POINT1 , - AV_CH_LAYOUT_5POINT0_BACK , - AV_CH_LAYOUT_5POINT1_BACK , - AV_CH_LAYOUT_7POINT0 , - AV_CH_LAYOUT_7POINT1 , - AV_CH_LAYOUT_7POINT1_WIDE , -}; - -static void setup_array(uint8_t *out[SWR_CH_MAX], uint8_t *in, enum AVSampleFormat format, int samples){ - if(av_sample_fmt_is_planar(format)){ - int i; - int plane_size= av_get_bytes_per_sample(format&0xFF)*samples; - format&=0xFF; - for(i=0; i<SWR_CH_MAX; i++){ - out[i]= in + i*plane_size; - } - }else{ - out[0]= in; - } -} - -static int cmp(const int *a, const int *b){ - return *a - *b; -} - -static void audiogen(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[SWR_CH_MAX]; - double tabf2[SWR_CH_MAX]; - double taba[SWR_CH_MAX]; - unsigned static rnd; - -#define PUT_SAMPLE set(data, ch, k, channels, sample_fmt, v); -#define uint_rand(x) (x = x * 1664525 + 1013904223) -#define dbl_rand(x) (uint_rand(x)*2.0 / (double)UINT_MAX - 1) - k = 0; - - /* 1 second of single freq sinus 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 varing 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 + uint_rand(rnd) % 5000; - tabf2[ch] = 100 + uint_rand(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; - } - } -} - -int main(int argc, char **argv){ - int in_sample_rate, out_sample_rate, ch ,i, flush_count; - uint64_t in_ch_layout, out_ch_layout; - enum AVSampleFormat in_sample_fmt, out_sample_fmt; - uint8_t array_in[SAMPLES*8*8]; - uint8_t array_mid[SAMPLES*8*8*3]; - uint8_t array_out[SAMPLES*8*8+100]; - uint8_t *ain[SWR_CH_MAX]; - uint8_t *aout[SWR_CH_MAX]; - uint8_t *amid[SWR_CH_MAX]; - int flush_i=0; - int mode; - int num_tests = 10000; - uint32_t seed = 0; - uint32_t rand_seed = 0; - int remaining_tests[FF_ARRAY_ELEMS(rates) * FF_ARRAY_ELEMS(layouts) * FF_ARRAY_ELEMS(formats) * FF_ARRAY_ELEMS(layouts) * FF_ARRAY_ELEMS(formats)]; - int max_tests = FF_ARRAY_ELEMS(remaining_tests); - int test; - int specific_test= -1; - - struct SwrContext * forw_ctx= NULL; - struct SwrContext *backw_ctx= NULL; - - if (argc > 1) { - if (!strcmp(argv[1], "-h") || !strcmp(argv[1], "--help")) { - av_log(NULL, AV_LOG_INFO, "Usage: swresample-test [<num_tests>[ <test>]] \n" - "num_tests Default is %d\n", num_tests); - return 0; - } - num_tests = strtol(argv[1], NULL, 0); - if(num_tests < 0) { - num_tests = -num_tests; - rand_seed = time(0); - } - if(num_tests<= 0 || num_tests>max_tests) - num_tests = max_tests; - if(argc > 2) { - specific_test = strtol(argv[1], NULL, 0); - } - } - - for(i=0; i<max_tests; i++) - remaining_tests[i] = i; - - for(test=0; test<num_tests; test++){ - unsigned r; - uint_rand(seed); - r = (seed * (uint64_t)(max_tests - test)) >>32; - FFSWAP(int, remaining_tests[r], remaining_tests[max_tests - test - 1]); - } - qsort(remaining_tests + max_tests - num_tests, num_tests, sizeof(remaining_tests[0]), (void*)cmp); - in_sample_rate=16000; - for(test=0; test<num_tests; test++){ - char in_layout_string[256]; - char out_layout_string[256]; - unsigned vector= remaining_tests[max_tests - test - 1]; - int in_ch_count; - int out_count, mid_count, out_ch_count; - - in_ch_layout = layouts[vector % FF_ARRAY_ELEMS(layouts)]; vector /= FF_ARRAY_ELEMS(layouts); - out_ch_layout = layouts[vector % FF_ARRAY_ELEMS(layouts)]; vector /= FF_ARRAY_ELEMS(layouts); - in_sample_fmt = formats[vector % FF_ARRAY_ELEMS(formats)]; vector /= FF_ARRAY_ELEMS(formats); - out_sample_fmt = formats[vector % FF_ARRAY_ELEMS(formats)]; vector /= FF_ARRAY_ELEMS(formats); - out_sample_rate = rates [vector % FF_ARRAY_ELEMS(rates )]; vector /= FF_ARRAY_ELEMS(rates); - av_assert0(!vector); - - if(specific_test == 0){ - if(out_sample_rate != in_sample_rate || in_ch_layout != out_ch_layout) - continue; - } - - in_ch_count= av_get_channel_layout_nb_channels(in_ch_layout); - out_ch_count= av_get_channel_layout_nb_channels(out_ch_layout); - av_get_channel_layout_string( in_layout_string, sizeof( in_layout_string), in_ch_count, in_ch_layout); - av_get_channel_layout_string(out_layout_string, sizeof(out_layout_string), out_ch_count, out_ch_layout); - fprintf(stderr, "TEST: %s->%s, rate:%5d->%5d, fmt:%s->%s\n", - in_layout_string, out_layout_string, - in_sample_rate, out_sample_rate, - av_get_sample_fmt_name(in_sample_fmt), av_get_sample_fmt_name(out_sample_fmt)); - forw_ctx = swr_alloc_set_opts(forw_ctx, out_ch_layout, out_sample_fmt, out_sample_rate, - in_ch_layout, in_sample_fmt, in_sample_rate, - 0, 0); - backw_ctx = swr_alloc_set_opts(backw_ctx, in_ch_layout, in_sample_fmt, in_sample_rate, - out_ch_layout, out_sample_fmt, out_sample_rate, - 0, 0); - if(!forw_ctx) { - fprintf(stderr, "Failed to init forw_cts\n"); - return 1; - } - if(!backw_ctx) { - fprintf(stderr, "Failed to init backw_ctx\n"); - return 1; - } - if(swr_init( forw_ctx) < 0) - fprintf(stderr, "swr_init(->) failed\n"); - if(swr_init(backw_ctx) < 0) - fprintf(stderr, "swr_init(<-) failed\n"); - //FIXME test planar - setup_array(ain , array_in , in_sample_fmt, SAMPLES); - setup_array(amid, array_mid, out_sample_fmt, 3*SAMPLES); - setup_array(aout, array_out, in_sample_fmt , SAMPLES); -#if 0 - for(ch=0; ch<in_ch_count; ch++){ - for(i=0; i<SAMPLES; i++) - set(ain, ch, i, in_ch_count, in_sample_fmt, sin(i*i*3/SAMPLES)); - } -#else - audiogen(ain, in_sample_fmt, in_ch_count, SAMPLES/6+1, SAMPLES); -#endif - mode = uint_rand(rand_seed) % 3; - if(mode==0 /*|| out_sample_rate == in_sample_rate*/) { - mid_count= swr_convert(forw_ctx, amid, 3*SAMPLES, (const uint8_t **)ain, SAMPLES); - } else if(mode==1){ - mid_count= swr_convert(forw_ctx, amid, 0, (const uint8_t **)ain, SAMPLES); - mid_count+=swr_convert(forw_ctx, amid, 3*SAMPLES, (const uint8_t **)ain, 0); - } else { - int tmp_count; - mid_count= swr_convert(forw_ctx, amid, 0, (const uint8_t **)ain, 1); - av_assert0(mid_count==0); - shift(ain, 1, in_ch_count, in_sample_fmt); - mid_count+=swr_convert(forw_ctx, amid, 3*SAMPLES, (const uint8_t **)ain, 0); - shift(amid, mid_count, out_ch_count, out_sample_fmt); tmp_count = mid_count; - mid_count+=swr_convert(forw_ctx, amid, 2, (const uint8_t **)ain, 2); - shift(amid, mid_count-tmp_count, out_ch_count, out_sample_fmt); tmp_count = mid_count; - shift(ain, 2, in_ch_count, in_sample_fmt); - mid_count+=swr_convert(forw_ctx, amid, 1, (const uint8_t **)ain, SAMPLES-3); - shift(amid, mid_count-tmp_count, out_ch_count, out_sample_fmt); tmp_count = mid_count; - shift(ain, -3, in_ch_count, in_sample_fmt); - mid_count+=swr_convert(forw_ctx, amid, 3*SAMPLES, (const uint8_t **)ain, 0); - shift(amid, -tmp_count, out_ch_count, out_sample_fmt); - } - out_count= swr_convert(backw_ctx,aout, SAMPLES, (const uint8_t **)amid, mid_count); - - for(ch=0; ch<in_ch_count; ch++){ - double sse, maxdiff=0; - double sum_a= 0; - double sum_b= 0; - double sum_aa= 0; - double sum_bb= 0; - double sum_ab= 0; - for(i=0; i<out_count; i++){ - double a= get(ain , ch, i, in_ch_count, in_sample_fmt); - double b= get(aout, ch, i, in_ch_count, in_sample_fmt); - sum_a += a; - sum_b += b; - sum_aa+= a*a; - sum_bb+= b*b; - sum_ab+= a*b; - maxdiff= FFMAX(maxdiff, FFABS(a-b)); - } - sse= sum_aa + sum_bb - 2*sum_ab; - if(sse < 0 && sse > -0.00001) sse=0; //fix rounding error - - fprintf(stderr, "[e:%f c:%f max:%f] len:%5d\n", out_count ? sqrt(sse/out_count) : 0, sum_ab/(sqrt(sum_aa*sum_bb)), maxdiff, out_count); - } - - flush_i++; - flush_i%=21; - flush_count = swr_convert(backw_ctx,aout, flush_i, 0, 0); - shift(aout, flush_i, in_ch_count, in_sample_fmt); - flush_count+= swr_convert(backw_ctx,aout, SAMPLES-flush_i, 0, 0); - shift(aout, -flush_i, in_ch_count, in_sample_fmt); - if(flush_count){ - for(ch=0; ch<in_ch_count; ch++){ - double sse, maxdiff=0; - double sum_a= 0; - double sum_b= 0; - double sum_aa= 0; - double sum_bb= 0; - double sum_ab= 0; - for(i=0; i<flush_count; i++){ - double a= get(ain , ch, i+out_count, in_ch_count, in_sample_fmt); - double b= get(aout, ch, i, in_ch_count, in_sample_fmt); - sum_a += a; - sum_b += b; - sum_aa+= a*a; - sum_bb+= b*b; - sum_ab+= a*b; - maxdiff= FFMAX(maxdiff, FFABS(a-b)); - } - sse= sum_aa + sum_bb - 2*sum_ab; - if(sse < 0 && sse > -0.00001) sse=0; //fix rounding error - - fprintf(stderr, "[e:%f c:%f max:%f] len:%5d F:%3d\n", sqrt(sse/flush_count), sum_ab/(sqrt(sum_aa*sum_bb)), maxdiff, flush_count, flush_i); - } - } - - - fprintf(stderr, "\n"); - } - - return 0; -} diff --git a/ffmpeg1/libswresample/swresample.c b/ffmpeg1/libswresample/swresample.c deleted file mode 100644 index 9b71b2e..0000000 --- a/ffmpeg1/libswresample/swresample.c +++ /dev/null @@ -1,932 +0,0 @@ -/* - * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include "libavutil/opt.h" -#include "swresample_internal.h" -#include "audioconvert.h" -#include "libavutil/avassert.h" -#include "libavutil/channel_layout.h" - -#include <float.h> - -#define C30DB M_SQRT2 -#define C15DB 1.189207115 -#define C__0DB 1.0 -#define C_15DB 0.840896415 -#define C_30DB M_SQRT1_2 -#define C_45DB 0.594603558 -#define C_60DB 0.5 - -#define ALIGN 32 - -//TODO split options array out? -#define OFFSET(x) offsetof(SwrContext,x) -#define PARAM AV_OPT_FLAG_AUDIO_PARAM - -static const AVOption options[]={ -{"ich" , "set input channel count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, -{"in_channel_count" , "set input channel count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, -{"och" , "set output channel count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, -{"out_channel_count" , "set output channel count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, -{"uch" , "set used channel count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, -{"used_channel_count" , "set used channel count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM}, -{"isr" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, -{"in_sample_rate" , "set input sample rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, -{"osr" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, -{"out_sample_rate" , "set output sample rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM}, -{"isf" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, -{"in_sample_fmt" , "set input sample format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, -{"osf" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, -{"out_sample_fmt" , "set output sample format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, -{"tsf" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, -{"internal_sample_fmt" , "set internal sample format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_SAMPLE_FMT , {.i64=AV_SAMPLE_FMT_NONE}, -1 , AV_SAMPLE_FMT_NB-1, PARAM}, -{"icl" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, -{"in_channel_layout" , "set input channel layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, -{"ocl" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, -{"out_channel_layout" , "set output channel layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"}, -{"clev" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, -{"center_mix_level" , "set center mix level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, -{"slev" , "set surround mix level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, -{"surround_mix_level" , "set surround mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM}, -{"lfe_mix_level" , "set LFE mix level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM}, -{"rmvol" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM}, -{"rematrix_volume" , "set rematrix volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM}, - -{"flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"}, -{"swr_flags" , "set flags" , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"}, -{"res" , "force resampling" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_FLAG_RESAMPLE }, INT_MIN, INT_MAX , PARAM, "flags"}, - -{"dither_scale" , "set dither scale" , OFFSET(dither.scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM}, - -{"dither_method" , "set dither method" , OFFSET(dither.method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"}, -{"rectangular" , "select rectangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"}, -{"triangular" , "select triangular dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"}, -{"triangular_hp" , "select triangular dither with high pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"lipshitz" , "select lipshitz noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LIPSHITZ}, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"shibata" , "select shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"low_shibata" , "select low shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_LOW_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"high_shibata" , "select high shibata noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_HIGH_SHIBATA }, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"f_weighted" , "select f-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_F_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"modified_e_weighted" , "select modified-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_MODIFIED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"}, -{"improved_e_weighted" , "select improved-e-weighted noise shaping dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_NS_IMPROVED_E_WEIGHTED }, INT_MIN, INT_MAX, PARAM, "dither_method"}, - -{"filter_size" , "set swr resampling filter size", OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=32 }, 0 , INT_MAX , PARAM }, -{"phase_shift" , "set swr resampling phase shift", OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 24 , PARAM }, -{"linear_interp" , "enable linear interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM }, -{"cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM }, - -/* duplicate option in order to work with avconv */ -{"resample_cutoff" , "set cutoff frequency ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0. }, 0 , 1 , PARAM }, - -{"resampler" , "set resampling Engine" , OFFSET(engine) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_ENGINE_NB-1, PARAM, "resampler"}, -{"swr" , "select SW Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SWR }, INT_MIN, INT_MAX , PARAM, "resampler"}, -{"soxr" , "select SoX Resampler" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_ENGINE_SOXR }, INT_MIN, INT_MAX , PARAM, "resampler"}, -{"precision" , "set soxr resampling precision (in bits)" - , OFFSET(precision) , AV_OPT_TYPE_DOUBLE,{.dbl=20.0 }, 15.0 , 33.0 , PARAM }, -{"cheby" , "enable soxr Chebyshev passband & higher-precision irrational ratio approximation" - , OFFSET(cheby) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM }, -{"min_comp" , "set minimum difference between timestamps and audio data (in seconds) below which no timestamp compensation of either kind is applied" - , OFFSET(min_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=FLT_MAX }, 0 , FLT_MAX , PARAM }, -{"min_hard_comp" , "set minimum difference between timestamps and audio data (in seconds) to trigger padding/trimming the data." - , OFFSET(min_hard_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0.1 }, 0 , INT_MAX , PARAM }, -{"comp_duration" , "set duration (in seconds) over which data is stretched/squeezed to make it match the timestamps." - , OFFSET(soft_compensation_duration),AV_OPT_TYPE_FLOAT ,{.dbl=1 }, 0 , INT_MAX , PARAM }, -{"max_soft_comp" , "set maximum factor by which data is stretched/squeezed to make it match the timestamps." - , OFFSET(max_soft_compensation),AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM }, -{"async" , "simplified 1 parameter audio timestamp matching, 0(disabled), 1(filling and trimming), >1(maximum stretch/squeeze in samples per second)" - , OFFSET(async) , AV_OPT_TYPE_FLOAT ,{.dbl=0 }, INT_MIN, INT_MAX , PARAM }, -{"first_pts" , "Assume the first pts should be this value (in samples)." - , OFFSET(firstpts_in_samples), AV_OPT_TYPE_INT64 ,{.i64=AV_NOPTS_VALUE }, INT64_MIN,INT64_MAX, PARAM }, - -{ "matrix_encoding" , "set 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", "select none", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, - { "dolby", "select Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, - { "dplii", "select Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" }, - -{ "filter_type" , "select swr filter type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" }, - { "cubic" , "select cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" }, - { "blackman_nuttall", "select Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" }, - { "kaiser" , "select Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" }, - -{ "kaiser_beta" , "set swr Kaiser Window Beta" , OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM }, - -{ "output_sample_bits" , "" , OFFSET(dither.output_sample_bits) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 64 , 0 }, -{0} -}; - -static const char* context_to_name(void* ptr) { - return "SWR"; -} - -static const AVClass av_class = { - .class_name = "SWResampler", - .item_name = context_to_name, - .option = options, - .version = LIBAVUTIL_VERSION_INT, - .log_level_offset_offset = OFFSET(log_level_offset), - .parent_log_context_offset = OFFSET(log_ctx), - .category = AV_CLASS_CATEGORY_SWRESAMPLER, -}; - -unsigned swresample_version(void) -{ - av_assert0(LIBSWRESAMPLE_VERSION_MICRO >= 100); - return LIBSWRESAMPLE_VERSION_INT; -} - -const char *swresample_configuration(void) -{ - return FFMPEG_CONFIGURATION; -} - -const char *swresample_license(void) -{ -#define LICENSE_PREFIX "libswresample license: " - return LICENSE_PREFIX FFMPEG_LICENSE + sizeof(LICENSE_PREFIX) - 1; -} - -int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map){ - if(!s || s->in_convert) // s needs to be allocated but not initialized - return AVERROR(EINVAL); - s->channel_map = channel_map; - return 0; -} - -const AVClass *swr_get_class(void) -{ - return &av_class; -} - -av_cold struct SwrContext *swr_alloc(void){ - SwrContext *s= av_mallocz(sizeof(SwrContext)); - if(s){ - s->av_class= &av_class; - av_opt_set_defaults(s); - } - return s; -} - -struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, - int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, - int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, - int log_offset, void *log_ctx){ - if(!s) s= swr_alloc(); - if(!s) return NULL; - - s->log_level_offset= log_offset; - s->log_ctx= log_ctx; - - av_opt_set_int(s, "ocl", out_ch_layout, 0); - av_opt_set_int(s, "osf", out_sample_fmt, 0); - av_opt_set_int(s, "osr", out_sample_rate, 0); - av_opt_set_int(s, "icl", in_ch_layout, 0); - av_opt_set_int(s, "isf", in_sample_fmt, 0); - av_opt_set_int(s, "isr", in_sample_rate, 0); - av_opt_set_int(s, "tsf", AV_SAMPLE_FMT_NONE, 0); - av_opt_set_int(s, "ich", av_get_channel_layout_nb_channels(s-> in_ch_layout), 0); - av_opt_set_int(s, "och", av_get_channel_layout_nb_channels(s->out_ch_layout), 0); - av_opt_set_int(s, "uch", 0, 0); - return s; -} - -static void set_audiodata_fmt(AudioData *a, enum AVSampleFormat fmt){ - a->fmt = fmt; - a->bps = av_get_bytes_per_sample(fmt); - a->planar= av_sample_fmt_is_planar(fmt); -} - -static void free_temp(AudioData *a){ - av_free(a->data); - memset(a, 0, sizeof(*a)); -} - -av_cold void swr_free(SwrContext **ss){ - SwrContext *s= *ss; - if(s){ - free_temp(&s->postin); - free_temp(&s->midbuf); - free_temp(&s->preout); - free_temp(&s->in_buffer); - free_temp(&s->silence); - free_temp(&s->drop_temp); - free_temp(&s->dither.noise); - free_temp(&s->dither.temp); - swri_audio_convert_free(&s-> in_convert); - swri_audio_convert_free(&s->out_convert); - swri_audio_convert_free(&s->full_convert); - if (s->resampler) - s->resampler->free(&s->resample); - swri_rematrix_free(s); - } - - av_freep(ss); -} - -av_cold int swr_init(struct SwrContext *s){ - int ret; - s->in_buffer_index= 0; - s->in_buffer_count= 0; - s->resample_in_constraint= 0; - free_temp(&s->postin); - free_temp(&s->midbuf); - free_temp(&s->preout); - free_temp(&s->in_buffer); - free_temp(&s->silence); - free_temp(&s->drop_temp); - free_temp(&s->dither.noise); - free_temp(&s->dither.temp); - memset(s->in.ch, 0, sizeof(s->in.ch)); - memset(s->out.ch, 0, sizeof(s->out.ch)); - swri_audio_convert_free(&s-> in_convert); - swri_audio_convert_free(&s->out_convert); - swri_audio_convert_free(&s->full_convert); - swri_rematrix_free(s); - - s->flushed = 0; - - if(s-> in_sample_fmt >= AV_SAMPLE_FMT_NB){ - av_log(s, AV_LOG_ERROR, "Requested input sample format %d is invalid\n", s->in_sample_fmt); - return AVERROR(EINVAL); - } - if(s->out_sample_fmt >= AV_SAMPLE_FMT_NB){ - av_log(s, AV_LOG_ERROR, "Requested output sample format %d is invalid\n", s->out_sample_fmt); - return AVERROR(EINVAL); - } - - if(av_get_channel_layout_nb_channels(s-> in_ch_layout) > SWR_CH_MAX) { - av_log(s, AV_LOG_WARNING, "Input channel layout 0x%"PRIx64" is invalid or unsupported.\n", s-> in_ch_layout); - s->in_ch_layout = 0; - } - - if(av_get_channel_layout_nb_channels(s->out_ch_layout) > SWR_CH_MAX) { - av_log(s, AV_LOG_WARNING, "Output channel layout 0x%"PRIx64" is invalid or unsupported.\n", s->out_ch_layout); - s->out_ch_layout = 0; - } - - switch(s->engine){ -#if CONFIG_LIBSOXR - extern struct Resampler const soxr_resampler; - case SWR_ENGINE_SOXR: s->resampler = &soxr_resampler; break; -#endif - case SWR_ENGINE_SWR : s->resampler = &swri_resampler; break; - default: - av_log(s, AV_LOG_ERROR, "Requested resampling engine is unavailable\n"); - return AVERROR(EINVAL); - } - - if(!s->used_ch_count) - s->used_ch_count= s->in.ch_count; - - if(s->used_ch_count && s-> in_ch_layout && s->used_ch_count != av_get_channel_layout_nb_channels(s-> in_ch_layout)){ - av_log(s, AV_LOG_WARNING, "Input channel layout has a different number of channels than the number of used channels, ignoring layout\n"); - s-> in_ch_layout= 0; - } - - if(!s-> in_ch_layout) - s-> in_ch_layout= av_get_default_channel_layout(s->used_ch_count); - if(!s->out_ch_layout) - s->out_ch_layout= av_get_default_channel_layout(s->out.ch_count); - - s->rematrix= s->out_ch_layout !=s->in_ch_layout || s->rematrix_volume!=1.0 || - s->rematrix_custom; - - if(s->int_sample_fmt == AV_SAMPLE_FMT_NONE){ - if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_S16P){ - s->int_sample_fmt= AV_SAMPLE_FMT_S16P; - }else if( av_get_planar_sample_fmt(s-> in_sample_fmt) == AV_SAMPLE_FMT_S32P - && av_get_planar_sample_fmt(s->out_sample_fmt) == AV_SAMPLE_FMT_S32P - && !s->rematrix - && s->engine != SWR_ENGINE_SOXR){ - s->int_sample_fmt= AV_SAMPLE_FMT_S32P; - }else if(av_get_planar_sample_fmt(s->in_sample_fmt) <= AV_SAMPLE_FMT_FLTP){ - s->int_sample_fmt= AV_SAMPLE_FMT_FLTP; - }else{ - av_log(s, AV_LOG_DEBUG, "Using double precision mode\n"); - s->int_sample_fmt= AV_SAMPLE_FMT_DBLP; - } - } - - if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P - &&s->int_sample_fmt != AV_SAMPLE_FMT_S32P - &&s->int_sample_fmt != AV_SAMPLE_FMT_FLTP - &&s->int_sample_fmt != AV_SAMPLE_FMT_DBLP){ - av_log(s, AV_LOG_ERROR, "Requested sample format %s is not supported internally, S16/S32/FLT/DBL is supported\n", av_get_sample_fmt_name(s->int_sample_fmt)); - return AVERROR(EINVAL); - } - - set_audiodata_fmt(&s-> in, s-> in_sample_fmt); - set_audiodata_fmt(&s->out, s->out_sample_fmt); - - if (s->firstpts_in_samples != AV_NOPTS_VALUE) { - if (!s->async && s->min_compensation >= FLT_MAX/2) - s->async = 1; - s->firstpts = - s->outpts = s->firstpts_in_samples * s->out_sample_rate; - } else - s->firstpts = AV_NOPTS_VALUE; - - if (s->async) { - if (s->min_compensation >= FLT_MAX/2) - s->min_compensation = 0.001; - if (s->async > 1.0001) { - s->max_soft_compensation = s->async / (double) s->in_sample_rate; - } - } - - if (s->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){ - s->resample = s->resampler->init(s->resample, s->out_sample_rate, s->in_sample_rate, s->filter_size, s->phase_shift, s->linear_interp, s->cutoff, s->int_sample_fmt, s->filter_type, s->kaiser_beta, s->precision, s->cheby); - }else - s->resampler->free(&s->resample); - if( s->int_sample_fmt != AV_SAMPLE_FMT_S16P - && s->int_sample_fmt != AV_SAMPLE_FMT_S32P - && s->int_sample_fmt != AV_SAMPLE_FMT_FLTP - && s->int_sample_fmt != AV_SAMPLE_FMT_DBLP - && s->resample){ - av_log(s, AV_LOG_ERROR, "Resampling only supported with internal s16/s32/flt/dbl\n"); - return -1; - } - -#define RSC 1 //FIXME finetune - if(!s-> in.ch_count) - s-> in.ch_count= av_get_channel_layout_nb_channels(s-> in_ch_layout); - if(!s->used_ch_count) - s->used_ch_count= s->in.ch_count; - if(!s->out.ch_count) - s->out.ch_count= av_get_channel_layout_nb_channels(s->out_ch_layout); - - if(!s-> in.ch_count){ - av_assert0(!s->in_ch_layout); - av_log(s, AV_LOG_ERROR, "Input channel count and layout are unset\n"); - return -1; - } - - if ((!s->out_ch_layout || !s->in_ch_layout) && s->used_ch_count != s->out.ch_count && !s->rematrix_custom) { - char l1[1024], l2[1024]; - av_get_channel_layout_string(l1, sizeof(l1), s-> in.ch_count, s-> in_ch_layout); - av_get_channel_layout_string(l2, sizeof(l2), s->out.ch_count, s->out_ch_layout); - av_log(s, AV_LOG_ERROR, "Rematrix is needed between %s and %s " - "but there is not enough information to do it\n", l1, l2); - return -1; - } - -av_assert0(s->used_ch_count); -av_assert0(s->out.ch_count); - s->resample_first= RSC*s->out.ch_count/s->in.ch_count - RSC < s->out_sample_rate/(float)s-> in_sample_rate - 1.0; - - s->in_buffer= s->in; - s->silence = s->in; - s->drop_temp= s->out; - - if(!s->resample && !s->rematrix && !s->channel_map && !s->dither.method){ - s->full_convert = swri_audio_convert_alloc(s->out_sample_fmt, - s-> in_sample_fmt, s-> in.ch_count, NULL, 0); - return 0; - } - - s->in_convert = swri_audio_convert_alloc(s->int_sample_fmt, - s-> in_sample_fmt, s->used_ch_count, s->channel_map, 0); - s->out_convert= swri_audio_convert_alloc(s->out_sample_fmt, - s->int_sample_fmt, s->out.ch_count, NULL, 0); - - if (!s->in_convert || !s->out_convert) - return AVERROR(ENOMEM); - - s->postin= s->in; - s->preout= s->out; - s->midbuf= s->in; - - if(s->channel_map){ - s->postin.ch_count= - s->midbuf.ch_count= s->used_ch_count; - if(s->resample) - s->in_buffer.ch_count= s->used_ch_count; - } - if(!s->resample_first){ - s->midbuf.ch_count= s->out.ch_count; - if(s->resample) - s->in_buffer.ch_count = s->out.ch_count; - } - - set_audiodata_fmt(&s->postin, s->int_sample_fmt); - set_audiodata_fmt(&s->midbuf, s->int_sample_fmt); - set_audiodata_fmt(&s->preout, s->int_sample_fmt); - - if(s->resample){ - set_audiodata_fmt(&s->in_buffer, s->int_sample_fmt); - } - - if ((ret = swri_dither_init(s, s->out_sample_fmt, s->int_sample_fmt)) < 0) - return ret; - - if(s->rematrix || s->dither.method) - return swri_rematrix_init(s); - - return 0; -} - -int swri_realloc_audio(AudioData *a, int count){ - int i, countb; - AudioData old; - - if(count < 0 || count > INT_MAX/2/a->bps/a->ch_count) - return AVERROR(EINVAL); - - if(a->count >= count) - return 0; - - count*=2; - - countb= FFALIGN(count*a->bps, ALIGN); - old= *a; - - av_assert0(a->bps); - av_assert0(a->ch_count); - - a->data= av_mallocz(countb*a->ch_count); - if(!a->data) - return AVERROR(ENOMEM); - for(i=0; i<a->ch_count; i++){ - a->ch[i]= a->data + i*(a->planar ? countb : a->bps); - if(a->planar) memcpy(a->ch[i], old.ch[i], a->count*a->bps); - } - if(!a->planar) memcpy(a->ch[0], old.ch[0], a->count*a->ch_count*a->bps); - av_free(old.data); - a->count= count; - - return 1; -} - -static void copy(AudioData *out, AudioData *in, - int count){ - av_assert0(out->planar == in->planar); - av_assert0(out->bps == in->bps); - av_assert0(out->ch_count == in->ch_count); - if(out->planar){ - int ch; - for(ch=0; ch<out->ch_count; ch++) - memcpy(out->ch[ch], in->ch[ch], count*out->bps); - }else - memcpy(out->ch[0], in->ch[0], count*out->ch_count*out->bps); -} - -static void fill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ - int i; - if(!in_arg){ - memset(out->ch, 0, sizeof(out->ch)); - }else if(out->planar){ - for(i=0; i<out->ch_count; i++) - out->ch[i]= in_arg[i]; - }else{ - for(i=0; i<out->ch_count; i++) - out->ch[i]= in_arg[0] + i*out->bps; - } -} - -static void reversefill_audiodata(AudioData *out, uint8_t *in_arg [SWR_CH_MAX]){ - int i; - if(out->planar){ - for(i=0; i<out->ch_count; i++) - in_arg[i]= out->ch[i]; - }else{ - in_arg[0]= out->ch[0]; - } -} - -/** - * - * out may be equal in. - */ -static void buf_set(AudioData *out, AudioData *in, int count){ - int ch; - if(in->planar){ - for(ch=0; ch<out->ch_count; ch++) - out->ch[ch]= in->ch[ch] + count*out->bps; - }else{ - for(ch=out->ch_count-1; ch>=0; ch--) - out->ch[ch]= in->ch[0] + (ch + count*out->ch_count) * out->bps; - } -} - -/** - * - * @return number of samples output per channel - */ -static int resample(SwrContext *s, AudioData *out_param, int out_count, - const AudioData * in_param, int in_count){ - AudioData in, out, tmp; - int ret_sum=0; - int border=0; - - av_assert1(s->in_buffer.ch_count == in_param->ch_count); - av_assert1(s->in_buffer.planar == in_param->planar); - av_assert1(s->in_buffer.fmt == in_param->fmt); - - tmp=out=*out_param; - in = *in_param; - - do{ - int ret, size, consumed; - if(!s->resample_in_constraint && s->in_buffer_count){ - buf_set(&tmp, &s->in_buffer, s->in_buffer_index); - ret= s->resampler->multiple_resample(s->resample, &out, out_count, &tmp, s->in_buffer_count, &consumed); - out_count -= ret; - ret_sum += ret; - buf_set(&out, &out, ret); - s->in_buffer_count -= consumed; - s->in_buffer_index += consumed; - - if(!in_count) - break; - if(s->in_buffer_count <= border){ - buf_set(&in, &in, -s->in_buffer_count); - in_count += s->in_buffer_count; - s->in_buffer_count=0; - s->in_buffer_index=0; - border = 0; - } - } - - if((s->flushed || in_count) && !s->in_buffer_count){ - s->in_buffer_index=0; - ret= s->resampler->multiple_resample(s->resample, &out, out_count, &in, in_count, &consumed); - out_count -= ret; - ret_sum += ret; - buf_set(&out, &out, ret); - in_count -= consumed; - buf_set(&in, &in, consumed); - } - - //TODO is this check sane considering the advanced copy avoidance below - size= s->in_buffer_index + s->in_buffer_count + in_count; - if( size > s->in_buffer.count - && s->in_buffer_count + in_count <= s->in_buffer_index){ - buf_set(&tmp, &s->in_buffer, s->in_buffer_index); - copy(&s->in_buffer, &tmp, s->in_buffer_count); - s->in_buffer_index=0; - }else - if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) - return ret; - - if(in_count){ - int count= in_count; - if(s->in_buffer_count && s->in_buffer_count+2 < count && out_count) count= s->in_buffer_count+2; - - buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); - copy(&tmp, &in, /*in_*/count); - s->in_buffer_count += count; - in_count -= count; - border += count; - buf_set(&in, &in, count); - s->resample_in_constraint= 0; - if(s->in_buffer_count != count || in_count) - continue; - } - break; - }while(1); - - s->resample_in_constraint= !!out_count; - - return ret_sum; -} - -static int swr_convert_internal(struct SwrContext *s, AudioData *out, int out_count, - AudioData *in , int in_count){ - AudioData *postin, *midbuf, *preout; - int ret/*, in_max*/; - AudioData preout_tmp, midbuf_tmp; - - if(s->full_convert){ - av_assert0(!s->resample); - swri_audio_convert(s->full_convert, out, in, in_count); - return out_count; - } - -// in_max= out_count*(int64_t)s->in_sample_rate / s->out_sample_rate + resample_filter_taps; -// in_count= FFMIN(in_count, in_in + 2 - s->hist_buffer_count); - - if((ret=swri_realloc_audio(&s->postin, in_count))<0) - return ret; - if(s->resample_first){ - av_assert0(s->midbuf.ch_count == s->used_ch_count); - if((ret=swri_realloc_audio(&s->midbuf, out_count))<0) - return ret; - }else{ - av_assert0(s->midbuf.ch_count == s->out.ch_count); - if((ret=swri_realloc_audio(&s->midbuf, in_count))<0) - return ret; - } - if((ret=swri_realloc_audio(&s->preout, out_count))<0) - return ret; - - postin= &s->postin; - - midbuf_tmp= s->midbuf; - midbuf= &midbuf_tmp; - preout_tmp= s->preout; - preout= &preout_tmp; - - if(s->int_sample_fmt == s-> in_sample_fmt && s->in.planar && !s->channel_map) - postin= in; - - if(s->resample_first ? !s->resample : !s->rematrix) - midbuf= postin; - - if(s->resample_first ? !s->rematrix : !s->resample) - preout= midbuf; - - if(s->int_sample_fmt == s->out_sample_fmt && s->out.planar){ - if(preout==in){ - out_count= FFMIN(out_count, in_count); //TODO check at the end if this is needed or redundant - av_assert0(s->in.planar); //we only support planar internally so it has to be, we support copying non planar though - copy(out, in, out_count); - return out_count; - } - else if(preout==postin) preout= midbuf= postin= out; - else if(preout==midbuf) preout= midbuf= out; - else preout= out; - } - - if(in != postin){ - swri_audio_convert(s->in_convert, postin, in, in_count); - } - - if(s->resample_first){ - if(postin != midbuf) - out_count= resample(s, midbuf, out_count, postin, in_count); - if(midbuf != preout) - swri_rematrix(s, preout, midbuf, out_count, preout==out); - }else{ - if(postin != midbuf) - swri_rematrix(s, midbuf, postin, in_count, midbuf==out); - if(midbuf != preout) - out_count= resample(s, preout, out_count, midbuf, in_count); - } - - if(preout != out && out_count){ - AudioData *conv_src = preout; - if(s->dither.method){ - int ch; - int dither_count= FFMAX(out_count, 1<<16); - - if (preout == in) { - conv_src = &s->dither.temp; - if((ret=swri_realloc_audio(&s->dither.temp, dither_count))<0) - return ret; - } - - if((ret=swri_realloc_audio(&s->dither.noise, dither_count))<0) - return ret; - if(ret) - for(ch=0; ch<s->dither.noise.ch_count; ch++) - swri_get_dither(s, s->dither.noise.ch[ch], s->dither.noise.count, 12345678913579<<ch, s->dither.noise.fmt); - av_assert0(s->dither.noise.ch_count == preout->ch_count); - - if(s->dither.noise_pos + out_count > s->dither.noise.count) - s->dither.noise_pos = 0; - - if (s->dither.method < SWR_DITHER_NS){ - if (s->mix_2_1_simd) { - int len1= out_count&~15; - int off = len1 * preout->bps; - - if(len1) - for(ch=0; ch<preout->ch_count; ch++) - s->mix_2_1_simd(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, len1); - if(out_count != len1) - for(ch=0; ch<preout->ch_count; ch++) - s->mix_2_1_f(conv_src->ch[ch] + off, preout->ch[ch] + off, s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos + off + len1, s->native_one, 0, 0, out_count - len1); - } else { - for(ch=0; ch<preout->ch_count; ch++) - s->mix_2_1_f(conv_src->ch[ch], preout->ch[ch], s->dither.noise.ch[ch] + s->dither.noise.bps * s->dither.noise_pos, s->native_one, 0, 0, out_count); - } - } else { - switch(s->int_sample_fmt) { - case AV_SAMPLE_FMT_S16P :swri_noise_shaping_int16(s, conv_src, preout, &s->dither.noise, out_count); break; - case AV_SAMPLE_FMT_S32P :swri_noise_shaping_int32(s, conv_src, preout, &s->dither.noise, out_count); break; - case AV_SAMPLE_FMT_FLTP :swri_noise_shaping_float(s, conv_src, preout, &s->dither.noise, out_count); break; - case AV_SAMPLE_FMT_DBLP :swri_noise_shaping_double(s,conv_src, preout, &s->dither.noise, out_count); break; - } - } - s->dither.noise_pos += out_count; - } -//FIXME packed doesnt need more than 1 chan here! - swri_audio_convert(s->out_convert, out, conv_src, out_count); - } - return out_count; -} - -int swr_convert(struct SwrContext *s, uint8_t *out_arg[SWR_CH_MAX], int out_count, - const uint8_t *in_arg [SWR_CH_MAX], int in_count){ - AudioData * in= &s->in; - AudioData *out= &s->out; - - while(s->drop_output > 0){ - int ret; - uint8_t *tmp_arg[SWR_CH_MAX]; -#define MAX_DROP_STEP 16384 - if((ret=swri_realloc_audio(&s->drop_temp, FFMIN(s->drop_output, MAX_DROP_STEP)))<0) - return ret; - - reversefill_audiodata(&s->drop_temp, tmp_arg); - s->drop_output *= -1; //FIXME find a less hackish solution - ret = swr_convert(s, tmp_arg, FFMIN(-s->drop_output, MAX_DROP_STEP), in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter - s->drop_output *= -1; - in_count = 0; - if(ret>0) { - s->drop_output -= ret; - continue; - } - - if(s->drop_output || !out_arg) - return 0; - } - - if(!in_arg){ - if(s->resample){ - if (!s->flushed) - s->resampler->flush(s); - s->resample_in_constraint = 0; - s->flushed = 1; - }else if(!s->in_buffer_count){ - return 0; - } - }else - fill_audiodata(in , (void*)in_arg); - - fill_audiodata(out, out_arg); - - if(s->resample){ - int ret = swr_convert_internal(s, out, out_count, in, in_count); - if(ret>0 && !s->drop_output) - s->outpts += ret * (int64_t)s->in_sample_rate; - return ret; - }else{ - AudioData tmp= *in; - int ret2=0; - int ret, size; - size = FFMIN(out_count, s->in_buffer_count); - if(size){ - buf_set(&tmp, &s->in_buffer, s->in_buffer_index); - ret= swr_convert_internal(s, out, size, &tmp, size); - if(ret<0) - return ret; - ret2= ret; - s->in_buffer_count -= ret; - s->in_buffer_index += ret; - buf_set(out, out, ret); - out_count -= ret; - if(!s->in_buffer_count) - s->in_buffer_index = 0; - } - - if(in_count){ - size= s->in_buffer_index + s->in_buffer_count + in_count - out_count; - - if(in_count > out_count) { //FIXME move after swr_convert_internal - if( size > s->in_buffer.count - && s->in_buffer_count + in_count - out_count <= s->in_buffer_index){ - buf_set(&tmp, &s->in_buffer, s->in_buffer_index); - copy(&s->in_buffer, &tmp, s->in_buffer_count); - s->in_buffer_index=0; - }else - if((ret=swri_realloc_audio(&s->in_buffer, size)) < 0) - return ret; - } - - if(out_count){ - size = FFMIN(in_count, out_count); - ret= swr_convert_internal(s, out, size, in, size); - if(ret<0) - return ret; - buf_set(in, in, ret); - in_count -= ret; - ret2 += ret; - } - if(in_count){ - buf_set(&tmp, &s->in_buffer, s->in_buffer_index + s->in_buffer_count); - copy(&tmp, in, in_count); - s->in_buffer_count += in_count; - } - } - if(ret2>0 && !s->drop_output) - s->outpts += ret2 * (int64_t)s->in_sample_rate; - return ret2; - } -} - -int swr_drop_output(struct SwrContext *s, int count){ - s->drop_output += count; - - if(s->drop_output <= 0) - return 0; - - av_log(s, AV_LOG_VERBOSE, "discarding %d audio samples\n", count); - return swr_convert(s, NULL, s->drop_output, NULL, 0); -} - -int swr_inject_silence(struct SwrContext *s, int count){ - int ret, i; - uint8_t *tmp_arg[SWR_CH_MAX]; - - if(count <= 0) - return 0; - -#define MAX_SILENCE_STEP 16384 - while (count > MAX_SILENCE_STEP) { - if ((ret = swr_inject_silence(s, MAX_SILENCE_STEP)) < 0) - return ret; - count -= MAX_SILENCE_STEP; - } - - if((ret=swri_realloc_audio(&s->silence, count))<0) - return ret; - - if(s->silence.planar) for(i=0; i<s->silence.ch_count; i++) { - memset(s->silence.ch[i], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps); - } else - memset(s->silence.ch[0], s->silence.bps==1 ? 0x80 : 0, count*s->silence.bps*s->silence.ch_count); - - reversefill_audiodata(&s->silence, tmp_arg); - av_log(s, AV_LOG_VERBOSE, "adding %d audio samples of silence\n", count); - ret = swr_convert(s, NULL, 0, (const uint8_t**)tmp_arg, count); - return ret; -} - -int64_t swr_get_delay(struct SwrContext *s, int64_t base){ - if (s->resampler && s->resample){ - return s->resampler->get_delay(s, base); - }else{ - return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate; - } -} - -int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){ - int ret; - - if (!s || compensation_distance < 0) - return AVERROR(EINVAL); - if (!compensation_distance && sample_delta) - return AVERROR(EINVAL); - if (!s->resample) { - s->flags |= SWR_FLAG_RESAMPLE; - ret = swr_init(s); - if (ret < 0) - return ret; - } - if (!s->resampler->set_compensation){ - return AVERROR(EINVAL); - }else{ - return s->resampler->set_compensation(s->resample, sample_delta, compensation_distance); - } -} - -int64_t swr_next_pts(struct SwrContext *s, int64_t pts){ - if(pts == INT64_MIN) - return s->outpts; - - if (s->firstpts == AV_NOPTS_VALUE) - s->outpts = s->firstpts = pts; - - if(s->min_compensation >= FLT_MAX) { - return (s->outpts = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate)); - } else { - int64_t delta = pts - swr_get_delay(s, s->in_sample_rate * (int64_t)s->out_sample_rate) - s->outpts + s->drop_output*(int64_t)s->in_sample_rate; - double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate); - - if(fabs(fdelta) > s->min_compensation) { - if(s->outpts == s->firstpts || fabs(fdelta) > s->min_hard_compensation){ - int ret; - if(delta > 0) ret = swr_inject_silence(s, delta / s->out_sample_rate); - else ret = swr_drop_output (s, -delta / s-> in_sample_rate); - if(ret<0){ - av_log(s, AV_LOG_ERROR, "Failed to compensate for timestamp delta of %f\n", fdelta); - } - } else if(s->soft_compensation_duration && s->max_soft_compensation) { - int duration = s->out_sample_rate * s->soft_compensation_duration; - double max_soft_compensation = s->max_soft_compensation / (s->max_soft_compensation < 0 ? -s->in_sample_rate : 1); - int comp = av_clipf(fdelta, -max_soft_compensation, max_soft_compensation) * duration ; - av_log(s, AV_LOG_VERBOSE, "compensating audio timestamp drift:%f compensation:%d in:%d\n", fdelta, comp, duration); - swr_set_compensation(s, comp, duration); - } - } - - return s->outpts; - } -} diff --git a/ffmpeg1/libswresample/swresample.h b/ffmpeg1/libswresample/swresample.h deleted file mode 100644 index 95e8a5a..0000000 --- a/ffmpeg1/libswresample/swresample.h +++ /dev/null @@ -1,311 +0,0 @@ -/* - * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#ifndef SWRESAMPLE_SWRESAMPLE_H -#define SWRESAMPLE_SWRESAMPLE_H - -/** - * @file - * @ingroup lswr - * libswresample public header - */ - -/** - * @defgroup lswr Libswresample - * @{ - * - * Libswresample (lswr) is a library that handles audio resampling, sample - * format conversion and mixing. - * - * Interaction with lswr is done through SwrContext, which is - * allocated with swr_alloc() or swr_alloc_set_opts(). 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 - * SwrContext *swr = swr_alloc(); - * av_opt_set_int(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0); - * av_opt_set_int(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0); - * av_opt_set_int(swr, "in_sample_rate", 48000, 0); - * av_opt_set_int(swr, "out_sample_rate", 44100, 0); - * av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); - * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); - * @endcode - * - * Once all values have been set, it must be initialized with swr_init(). If - * you need to change the conversion parameters, you can change the parameters - * as described above, or by using swr_alloc_set_opts(), then call swr_init() - * again. - * - * The conversion itself is done by repeatedly calling swr_convert(). - * Note that the samples may get buffered in swr if you provide insufficient - * output space or if sample rate conversion is done, which requires "future" - * samples. Samples that do not require future input can be retrieved at any - * time by using swr_convert() (in_count can be set to 0). - * At the end of conversion the resampling buffer can be flushed by calling - * swr_convert() with NULL in and 0 in_count. - * - * The delay between input and output, can at any time be found by using - * swr_get_delay(). - * - * 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_samples; - * - * while (get_input(&input, &in_samples)) { - * uint8_t *output; - * int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) + - * in_samples, 44100, 48000, AV_ROUND_UP); - * av_samples_alloc(&output, NULL, 2, out_samples, - * AV_SAMPLE_FMT_S16, 0); - * out_samples = swr_convert(swr, &output, out_samples, - * input, in_samples); - * handle_output(output, out_samples); - * av_freep(&output); - * } - * @endcode - * - * When the conversion is finished, the conversion - * context and everything associated with it must be freed with swr_free(). - * There will be no memory leak if the data is not completely flushed before - * swr_free(). - */ - -#include <stdint.h> -#include "libavutil/samplefmt.h" - -#include "libswresample/version.h" - -#if LIBSWRESAMPLE_VERSION_MAJOR < 1 -#define SWR_CH_MAX 32 ///< Maximum number of channels -#endif - -#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate -//TODO use int resample ? -//long term TODO can we enable this dynamically? - -enum SwrDitherType { - SWR_DITHER_NONE = 0, - SWR_DITHER_RECTANGULAR, - SWR_DITHER_TRIANGULAR, - SWR_DITHER_TRIANGULAR_HIGHPASS, - - SWR_DITHER_NS = 64, ///< not part of API/ABI - SWR_DITHER_NS_LIPSHITZ, - SWR_DITHER_NS_F_WEIGHTED, - SWR_DITHER_NS_MODIFIED_E_WEIGHTED, - SWR_DITHER_NS_IMPROVED_E_WEIGHTED, - SWR_DITHER_NS_SHIBATA, - SWR_DITHER_NS_LOW_SHIBATA, - SWR_DITHER_NS_HIGH_SHIBATA, - SWR_DITHER_NB, ///< not part of API/ABI -}; - -/** Resampling Engines */ -enum SwrEngine { - SWR_ENGINE_SWR, /**< SW Resampler */ - SWR_ENGINE_SOXR, /**< SoX Resampler */ - SWR_ENGINE_NB, ///< not part of API/ABI -}; - -/** Resampling Filter Types */ -enum SwrFilterType { - SWR_FILTER_TYPE_CUBIC, /**< Cubic */ - SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */ - SWR_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */ -}; - -typedef struct SwrContext SwrContext; - -/** - * Get the AVClass for swrContext. It can be used in combination with - * AV_OPT_SEARCH_FAKE_OBJ for examining options. - * - * @see av_opt_find(). - */ -const AVClass *swr_get_class(void); - -/** - * Allocate SwrContext. - * - * If you use this function you will need to set the parameters (manually or - * with swr_alloc_set_opts()) before calling swr_init(). - * - * @see swr_alloc_set_opts(), swr_init(), swr_free() - * @return NULL on error, allocated context otherwise - */ -struct SwrContext *swr_alloc(void); - -/** - * Initialize context after user parameters have been set. - * - * @return AVERROR error code in case of failure. - */ -int swr_init(struct SwrContext *s); - -/** - * Allocate SwrContext if needed and set/reset common parameters. - * - * This function does not require s to be allocated with swr_alloc(). On the - * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters - * on the allocated context. - * - * @param s Swr context, can be NULL - * @param out_ch_layout output channel layout (AV_CH_LAYOUT_*) - * @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*). - * @param out_sample_rate output sample rate (frequency in Hz) - * @param in_ch_layout input channel layout (AV_CH_LAYOUT_*) - * @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*). - * @param in_sample_rate input sample rate (frequency in Hz) - * @param log_offset logging level offset - * @param log_ctx parent logging context, can be NULL - * - * @see swr_init(), swr_free() - * @return NULL on error, allocated context otherwise - */ -struct SwrContext *swr_alloc_set_opts(struct SwrContext *s, - int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate, - int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate, - int log_offset, void *log_ctx); - -/** - * Free the given SwrContext and set the pointer to NULL. - */ -void swr_free(struct SwrContext **s); - -/** - * Convert audio. - * - * in and in_count can be set to 0 to flush the last few samples out at the - * end. - * - * If more input is provided than output space then the input will be buffered. - * You can avoid this buffering by providing more output space than input. - * Convertion will run directly without copying whenever possible. - * - * @param s allocated Swr context, with parameters set - * @param out output buffers, only the first one need be set in case of packed audio - * @param out_count amount of space available for output in samples per channel - * @param in input buffers, only the first one need to be set in case of packed audio - * @param in_count number of input samples available in one channel - * - * @return number of samples output per channel, negative value on error - */ -int swr_convert(struct SwrContext *s, uint8_t **out, int out_count, - const uint8_t **in , int in_count); - -/** - * Convert the next timestamp from input to output - * timestamps are in 1/(in_sample_rate * out_sample_rate) units. - * - * @note There are 2 slightly differently behaving modes. - * First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX) - * in this case timestamps will be passed through with delays compensated - * Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX) - * in this case the output timestamps will match output sample numbers - * - * @param pts timestamp for the next input sample, INT64_MIN if unknown - * @return the output timestamp for the next output sample - */ -int64_t swr_next_pts(struct SwrContext *s, int64_t pts); - -/** - * Activate resampling compensation. - */ -int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance); - -/** - * Set a customized input channel mapping. - * - * @param s allocated Swr context, not yet initialized - * @param channel_map customized input channel mapping (array of channel - * indexes, -1 for a muted channel) - * @return AVERROR error code in case of failure. - */ -int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map); - -/** - * Set a customized remix matrix. - * - * @param s allocated Swr context, not yet initialized - * @param matrix remix coefficients; matrix[i + stride * o] is - * the weight of input channel i in output channel o - * @param stride offset between lines of the matrix - * @return AVERROR error code in case of failure. - */ -int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride); - -/** - * Drops the specified number of output samples. - */ -int swr_drop_output(struct SwrContext *s, int count); - -/** - * Injects the specified number of silence samples. - */ -int swr_inject_silence(struct SwrContext *s, int count); - -/** - * Gets the delay the next input sample will experience relative to the next output sample. - * - * Swresample can buffer data if more input has been provided than available - * output space, also converting between sample rates needs a delay. - * This function returns the sum of all such delays. - * The exact delay is not necessarily an integer value in either input or - * output sample rate. Especially when downsampling by a large value, the - * output sample rate may be a poor choice to represent the delay, similarly - * for upsampling and the input sample rate. - * - * @param s swr context - * @param base timebase in which the returned delay will be - * if its set to 1 the returned delay is in seconds - * if its set to 1000 the returned delay is in milli seconds - * if its set to the input sample rate then the returned delay is in input samples - * if its set to the output sample rate then the returned delay is in output samples - * an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate) - * @returns the delay in 1/base units. - */ -int64_t swr_get_delay(struct SwrContext *s, int64_t base); - -/** - * Return the LIBSWRESAMPLE_VERSION_INT constant. - */ -unsigned swresample_version(void); - -/** - * Return the swr build-time configuration. - */ -const char *swresample_configuration(void); - -/** - * Return the swr license. - */ -const char *swresample_license(void); - -/** - * @} - */ - -#endif /* SWRESAMPLE_SWRESAMPLE_H */ diff --git a/ffmpeg1/libswresample/swresample_internal.h b/ffmpeg1/libswresample/swresample_internal.h deleted file mode 100644 index 17b85d5..0000000 --- a/ffmpeg1/libswresample/swresample_internal.h +++ /dev/null @@ -1,197 +0,0 @@ -/* - * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#ifndef SWR_INTERNAL_H -#define SWR_INTERNAL_H - -#include "swresample.h" -#include "libavutil/channel_layout.h" -#include "config.h" - -#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */ - -#define NS_TAPS 20 - -#if ARCH_X86_64 -typedef int64_t integer; -#else -typedef int integer; -#endif - -typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len); -typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len); - -typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len); - -typedef struct AudioData{ - uint8_t *ch[SWR_CH_MAX]; ///< samples buffer per channel - uint8_t *data; ///< samples buffer - int ch_count; ///< number of channels - int bps; ///< bytes per sample - int count; ///< number of samples - int planar; ///< 1 if planar audio, 0 otherwise - enum AVSampleFormat fmt; ///< sample format -} AudioData; - -struct DitherContext { - enum SwrDitherType method; - int noise_pos; - float scale; - float noise_scale; ///< Noise scale - int ns_taps; ///< Noise shaping dither taps - float ns_scale; ///< Noise shaping dither scale - float ns_scale_1; ///< Noise shaping dither scale^-1 - int ns_pos; ///< Noise shaping dither position - float ns_coeffs[NS_TAPS]; ///< Noise shaping filter coefficients - float ns_errors[SWR_CH_MAX][2*NS_TAPS]; - AudioData noise; ///< noise used for dithering - AudioData temp; ///< temporary storage when writing into the input buffer isnt possible - int output_sample_bits; ///< the number of used output bits, needed to scale dither correctly -}; - -struct SwrContext { - const AVClass *av_class; ///< AVClass used for AVOption and av_log() - int log_level_offset; ///< logging level offset - void *log_ctx; ///< parent logging context - enum AVSampleFormat in_sample_fmt; ///< input sample format - enum AVSampleFormat int_sample_fmt; ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P) - enum AVSampleFormat out_sample_fmt; ///< output sample format - int64_t in_ch_layout; ///< input channel layout - int64_t out_ch_layout; ///< output channel layout - int in_sample_rate; ///< input sample rate - int out_sample_rate; ///< output sample rate - int flags; ///< miscellaneous flags such as SWR_FLAG_RESAMPLE - float slev; ///< surround mixing level - float clev; ///< center mixing level - float lfe_mix_level; ///< LFE mixing level - float rematrix_volume; ///< rematrixing volume coefficient - enum AVMatrixEncoding matrix_encoding; /**< matrixed stereo encoding */ - const int *channel_map; ///< channel index (or -1 if muted channel) map - int used_ch_count; ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count) - enum SwrEngine engine; - - struct DitherContext dither; - - 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 (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */ - enum SwrFilterType filter_type; /**< swr resampling filter type */ - int kaiser_beta; /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */ - double precision; /**< soxr resampling precision (in bits) */ - int cheby; /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */ - - float min_compensation; ///< swr minimum below which no compensation will happen - float min_hard_compensation; ///< swr minimum below which no silence inject / sample drop will happen - float soft_compensation_duration; ///< swr duration over which soft compensation is applied - float max_soft_compensation; ///< swr maximum soft compensation in seconds over soft_compensation_duration - float async; ///< swr simple 1 parameter async, similar to ffmpegs -async - int64_t firstpts_in_samples; ///< swr first pts in samples - - int resample_first; ///< 1 if resampling must come first, 0 if rematrixing - int rematrix; ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch) - int rematrix_custom; ///< flag to indicate that a custom matrix has been defined - - AudioData in; ///< input audio data - AudioData postin; ///< post-input audio data: used for rematrix/resample - AudioData midbuf; ///< intermediate audio data (postin/preout) - AudioData preout; ///< pre-output audio data: used for rematrix/resample - AudioData out; ///< converted output audio data - AudioData in_buffer; ///< cached audio data (convert and resample purpose) - AudioData silence; ///< temporary with silence - AudioData drop_temp; ///< temporary used to discard output - int in_buffer_index; ///< cached buffer position - int in_buffer_count; ///< cached buffer length - int resample_in_constraint; ///< 1 if the input end was reach before the output end, 0 otherwise - int flushed; ///< 1 if data is to be flushed and no further input is expected - int64_t outpts; ///< output PTS - int64_t firstpts; ///< first PTS - int drop_output; ///< number of output samples to drop - - struct AudioConvert *in_convert; ///< input conversion context - struct AudioConvert *out_convert; ///< output conversion context - struct AudioConvert *full_convert; ///< full conversion context (single conversion for input and output) - struct ResampleContext *resample; ///< resampling context - struct Resampler const *resampler; ///< resampler virtual function table - - float matrix[SWR_CH_MAX][SWR_CH_MAX]; ///< floating point rematrixing coefficients - uint8_t *native_matrix; - uint8_t *native_one; - uint8_t *native_simd_matrix; - int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX]; ///< 17.15 fixed point rematrixing coefficients - uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1]; ///< Lists of input channels per output channel that have non zero rematrixing coefficients - mix_1_1_func_type *mix_1_1_f; - mix_1_1_func_type *mix_1_1_simd; - - mix_2_1_func_type *mix_2_1_f; - mix_2_1_func_type *mix_2_1_simd; - - mix_any_func_type *mix_any_f; - - /* TODO: callbacks for ASM optimizations */ -}; - -typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, - double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby); -typedef void (* resample_free_func)(struct ResampleContext **c); -typedef int (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed); -typedef int (* resample_flush_func)(struct SwrContext *c); -typedef int (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance); -typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base); - -struct Resampler { - resample_init_func init; - resample_free_func free; - multiple_resample_func multiple_resample; - resample_flush_func flush; - set_compensation_func set_compensation; - get_delay_func get_delay; -}; - -extern struct Resampler const swri_resampler; - -int swri_realloc_audio(AudioData *a, int count); -int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx); -int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx); -int swri_resample_float(struct ResampleContext *c, float *dst, const float *src, int *consumed, int src_size, int dst_size, int update_ctx); -int swri_resample_double(struct ResampleContext *c,double *dst, const double *src, int *consumed, int src_size, int dst_size, int update_ctx); - -void swri_noise_shaping_int16 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); -void swri_noise_shaping_int32 (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); -void swri_noise_shaping_float (SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); -void swri_noise_shaping_double(SwrContext *s, AudioData *dsts, const AudioData *srcs, const AudioData *noises, int count); - -int swri_rematrix_init(SwrContext *s); -void swri_rematrix_free(SwrContext *s); -int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy); -void swri_rematrix_init_x86(struct SwrContext *s); - -void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt); -int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt); - -void swri_audio_convert_init_arm(struct AudioConvert *ac, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels); -void swri_audio_convert_init_x86(struct AudioConvert *ac, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels); -#endif diff --git a/ffmpeg1/libswresample/version.h b/ffmpeg1/libswresample/version.h deleted file mode 100644 index df9df48..0000000 --- a/ffmpeg1/libswresample/version.h +++ /dev/null @@ -1,45 +0,0 @@ -/* - * Version macros. - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#ifndef SWR_VERSION_H -#define SWR_VERSION_H - -/** - * @file - * Libswresample version macros - */ - -#include "libavutil/avutil.h" - -#define LIBSWRESAMPLE_VERSION_MAJOR 0 -#define LIBSWRESAMPLE_VERSION_MINOR 17 -#define LIBSWRESAMPLE_VERSION_MICRO 102 - -#define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \ - LIBSWRESAMPLE_VERSION_MINOR, \ - LIBSWRESAMPLE_VERSION_MICRO) -#define LIBSWRESAMPLE_VERSION AV_VERSION(LIBSWRESAMPLE_VERSION_MAJOR, \ - LIBSWRESAMPLE_VERSION_MINOR, \ - LIBSWRESAMPLE_VERSION_MICRO) -#define LIBSWRESAMPLE_BUILD LIBSWRESAMPLE_VERSION_INT - -#define LIBSWRESAMPLE_IDENT "SwR" AV_STRINGIFY(LIBSWRESAMPLE_VERSION) - -#endif /* SWR_VERSION_H */ diff --git a/ffmpeg1/libswresample/x86/Makefile b/ffmpeg1/libswresample/x86/Makefile deleted file mode 100644 index e8feede..0000000 --- a/ffmpeg1/libswresample/x86/Makefile +++ /dev/null @@ -1,3 +0,0 @@ -YASM-OBJS += x86/swresample_x86.o\ - x86/audio_convert.o\ - x86/rematrix.o\ diff --git a/ffmpeg1/libswresample/x86/audio_convert.asm b/ffmpeg1/libswresample/x86/audio_convert.asm deleted file mode 100644 index ad46977..0000000 --- a/ffmpeg1/libswresample/x86/audio_convert.asm +++ /dev/null @@ -1,461 +0,0 @@ -;****************************************************************************** -;* Copyright (c) 2012 Michael Niedermayer -;* -;* 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 -align 32 -flt2pm31: times 8 dd 4.6566129e-10 -flt2p31 : times 8 dd 2147483648.0 -flt2p15 : times 8 dd 32768.0 - -word_unpack_shuf : db 0, 1, 4, 5, 8, 9,12,13, 2, 3, 6, 7,10,11,14,15 - -SECTION .text - - -;to, from, a/u, log2_outsize, log_intsize, const -%macro PACK_2CH 5-7 -cglobal pack_2ch_%2_to_%1_%3, 3, 4, 6, dst, src, len, src2 - mov src2q , [srcq+gprsize] - mov srcq , [srcq] - mov dstq , [dstq] -%ifidn %3, a - test dstq, mmsize-1 - jne pack_2ch_%2_to_%1_u_int %+ SUFFIX - test srcq, mmsize-1 - jne pack_2ch_%2_to_%1_u_int %+ SUFFIX - test src2q, mmsize-1 - jne pack_2ch_%2_to_%1_u_int %+ SUFFIX -%else -pack_2ch_%2_to_%1_u_int %+ SUFFIX -%endif - lea srcq , [srcq + (1<<%5)*lenq] - lea src2q, [src2q + (1<<%5)*lenq] - lea dstq , [dstq + (2<<%4)*lenq] - neg lenq - %7 m0,m1,m2,m3,m4,m5 -.next: -%if %4 >= %5 - mov%3 m0, [ srcq +(1<<%5)*lenq] - mova m1, m0 - mov%3 m2, [ src2q+(1<<%5)*lenq] -%if %5 == 1 - punpcklwd m0, m2 - punpckhwd m1, m2 -%else - punpckldq m0, m2 - punpckhdq m1, m2 -%endif - %6 m0,m1,m2,m3,m4,m5 -%else - mov%3 m0, [ srcq +(1<<%5)*lenq] - mov%3 m1, [mmsize + srcq +(1<<%5)*lenq] - mov%3 m2, [ src2q+(1<<%5)*lenq] - mov%3 m3, [mmsize + src2q+(1<<%5)*lenq] - %6 m0,m1,m2,m3,m4,m5 - mova m2, m0 - punpcklwd m0, m1 - punpckhwd m2, m1 - SWAP 1,2 -%endif - mov%3 [ dstq+(2<<%4)*lenq], m0 - mov%3 [ mmsize + dstq+(2<<%4)*lenq], m1 -%if %4 > %5 - mov%3 [2*mmsize + dstq+(2<<%4)*lenq], m2 - mov%3 [3*mmsize + dstq+(2<<%4)*lenq], m3 - add lenq, 4*mmsize/(2<<%4) -%else - add lenq, 2*mmsize/(2<<%4) -%endif - jl .next - REP_RET -%endmacro - -%macro UNPACK_2CH 5-7 -cglobal unpack_2ch_%2_to_%1_%3, 3, 4, 7, dst, src, len, dst2 - mov dst2q , [dstq+gprsize] - mov srcq , [srcq] - mov dstq , [dstq] -%ifidn %3, a - test dstq, mmsize-1 - jne unpack_2ch_%2_to_%1_u_int %+ SUFFIX - test srcq, mmsize-1 - jne unpack_2ch_%2_to_%1_u_int %+ SUFFIX - test dst2q, mmsize-1 - jne unpack_2ch_%2_to_%1_u_int %+ SUFFIX -%else -unpack_2ch_%2_to_%1_u_int %+ SUFFIX -%endif - lea srcq , [srcq + (2<<%5)*lenq] - lea dstq , [dstq + (1<<%4)*lenq] - lea dst2q, [dst2q + (1<<%4)*lenq] - neg lenq - %7 m0,m1,m2,m3,m4,m5 - mova m6, [word_unpack_shuf] -.next: - mov%3 m0, [ srcq +(2<<%5)*lenq] - mov%3 m2, [ mmsize + srcq +(2<<%5)*lenq] -%if %5 == 1 -%ifidn SUFFIX, _ssse3 - pshufb m0, m6 - mova m1, m0 - pshufb m2, m6 - punpcklqdq m0,m2 - punpckhqdq m1,m2 -%else - mova m1, m0 - punpcklwd m0,m2 - punpckhwd m1,m2 - - mova m2, m0 - punpcklwd m0,m1 - punpckhwd m2,m1 - - mova m1, m0 - punpcklwd m0,m2 - punpckhwd m1,m2 -%endif -%else - mova m1, m0 - shufps m0, m2, 10001000b - shufps m1, m2, 11011101b -%endif -%if %4 < %5 - mov%3 m2, [2*mmsize + srcq +(2<<%5)*lenq] - mova m3, m2 - mov%3 m4, [3*mmsize + srcq +(2<<%5)*lenq] - shufps m2, m4, 10001000b - shufps m3, m4, 11011101b - SWAP 1,2 -%endif - %6 m0,m1,m2,m3,m4,m5 - mov%3 [ dstq+(1<<%4)*lenq], m0 -%if %4 > %5 - mov%3 [ dst2q+(1<<%4)*lenq], m2 - mov%3 [ mmsize + dstq+(1<<%4)*lenq], m1 - mov%3 [ mmsize + dst2q+(1<<%4)*lenq], m3 - add lenq, 2*mmsize/(1<<%4) -%else - mov%3 [ dst2q+(1<<%4)*lenq], m1 - add lenq, mmsize/(1<<%4) -%endif - jl .next - REP_RET -%endmacro - -%macro CONV 5-7 -cglobal %2_to_%1_%3, 3, 3, 6, dst, src, len - mov srcq , [srcq] - mov dstq , [dstq] -%ifidn %3, a - test dstq, mmsize-1 - jne %2_to_%1_u_int %+ SUFFIX - test srcq, mmsize-1 - jne %2_to_%1_u_int %+ SUFFIX -%else -%2_to_%1_u_int %+ SUFFIX -%endif - lea srcq , [srcq + (1<<%5)*lenq] - lea dstq , [dstq + (1<<%4)*lenq] - neg lenq - %7 m0,m1,m2,m3,m4,m5 -.next: - mov%3 m0, [ srcq +(1<<%5)*lenq] - mov%3 m1, [ mmsize + srcq +(1<<%5)*lenq] -%if %4 < %5 - mov%3 m2, [2*mmsize + srcq +(1<<%5)*lenq] - mov%3 m3, [3*mmsize + srcq +(1<<%5)*lenq] -%endif - %6 m0,m1,m2,m3,m4,m5 - mov%3 [ dstq+(1<<%4)*lenq], m0 - mov%3 [ mmsize + dstq+(1<<%4)*lenq], m1 -%if %4 > %5 - mov%3 [2*mmsize + dstq+(1<<%4)*lenq], m2 - mov%3 [3*mmsize + dstq+(1<<%4)*lenq], m3 - add lenq, 4*mmsize/(1<<%4) -%else - add lenq, 2*mmsize/(1<<%4) -%endif - jl .next - REP_RET -%endmacro - -%macro PACK_6CH 5-7 -cglobal pack_6ch_%2_to_%1_%3, 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] - mov dstq, [dstq] -%ifidn %3, a - test dstq, mmsize-1 - jne pack_6ch_%2_to_%1_u_int %+ SUFFIX - test srcq, mmsize-1 - jne pack_6ch_%2_to_%1_u_int %+ SUFFIX - test src2q, mmsize-1 - jne pack_6ch_%2_to_%1_u_int %+ SUFFIX - test src3q, mmsize-1 - jne pack_6ch_%2_to_%1_u_int %+ SUFFIX - test src4q, mmsize-1 - jne pack_6ch_%2_to_%1_u_int %+ SUFFIX - test src5q, mmsize-1 - jne pack_6ch_%2_to_%1_u_int %+ SUFFIX -%else -pack_6ch_%2_to_%1_u_int %+ SUFFIX -%endif - sub src1q, srcq - sub src2q, srcq - sub src3q, srcq - sub src4q, srcq - sub src5q, srcq -.loop: - mov%3 m0, [srcq ] - mov%3 m1, [srcq+src1q] - mov%3 m2, [srcq+src2q] - mov%3 m3, [srcq+src3q] - mov%3 m4, [srcq+src4q] - mov%3 m5, [srcq+src5q] - %7 x,x,x,x,m7,x -%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 - - %6 m0,m6,x,x,m7,m3 - %6 m4,m1,x,x,m7,m3 - %6 m2,m5,x,x,m7,m3 - - mov %+ %3 %+ ps [dstq ], m0 - mov %+ %3 %+ ps [dstq+16], m6 - mov %+ %3 %+ ps [dstq+32], m4 - mov %+ %3 %+ ps [dstq+48], m1 - mov %+ %3 %+ ps [dstq+64], m2 - mov %+ %3 %+ ps [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 - -%macro INT16_TO_INT32_N 6 - pxor m2, m2 - pxor m3, m3 - punpcklwd m2, m1 - punpckhwd m3, m1 - SWAP 4,0 - pxor m0, m0 - pxor m1, m1 - punpcklwd m0, m4 - punpckhwd m1, m4 -%endmacro - -%macro INT32_TO_INT16_N 6 - psrad m0, 16 - psrad m1, 16 - psrad m2, 16 - psrad m3, 16 - packssdw m0, m1 - packssdw m2, m3 - SWAP 1,2 -%endmacro - -%macro INT32_TO_FLOAT_INIT 6 - mova %5, [flt2pm31] -%endmacro -%macro INT32_TO_FLOAT_N 6 - cvtdq2ps %1, %1 - cvtdq2ps %2, %2 - mulps %1, %1, %5 - mulps %2, %2, %5 -%endmacro - -%macro FLOAT_TO_INT32_INIT 6 - mova %5, [flt2p31] -%endmacro -%macro FLOAT_TO_INT32_N 6 - mulps %1, %5 - mulps %2, %5 - cvtps2dq %6, %1 - cmpnltps %1, %5 - paddd %1, %6 - cvtps2dq %6, %2 - cmpnltps %2, %5 - paddd %2, %6 -%endmacro - -%macro INT16_TO_FLOAT_INIT 6 - mova m5, [flt2pm31] -%endmacro -%macro INT16_TO_FLOAT_N 6 - INT16_TO_INT32_N %1,%2,%3,%4,%5,%6 - cvtdq2ps m0, m0 - cvtdq2ps m1, m1 - cvtdq2ps m2, m2 - cvtdq2ps m3, m3 - mulps m0, m0, m5 - mulps m1, m1, m5 - mulps m2, m2, m5 - mulps m3, m3, m5 -%endmacro - -%macro FLOAT_TO_INT16_INIT 6 - mova m5, [flt2p15] -%endmacro -%macro FLOAT_TO_INT16_N 6 - mulps m0, m5 - mulps m1, m5 - mulps m2, m5 - mulps m3, m5 - cvtps2dq m0, m0 - cvtps2dq m1, m1 - packssdw m0, m1 - cvtps2dq m1, m2 - cvtps2dq m3, m3 - packssdw m1, m3 -%endmacro - -%macro NOP_N 0-6 -%endmacro - -INIT_MMX mmx -CONV int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N -CONV int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N -CONV int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N -CONV int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N - -PACK_6CH float, float, u, 2, 2, NOP_N, NOP_N -PACK_6CH float, float, a, 2, 2, NOP_N, NOP_N - -INIT_XMM sse2 -CONV int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N -CONV int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N -CONV int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N -CONV int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N - -PACK_2CH int16, int16, u, 1, 1, NOP_N, NOP_N -PACK_2CH int16, int16, a, 1, 1, NOP_N, NOP_N -PACK_2CH int32, int32, u, 2, 2, NOP_N, NOP_N -PACK_2CH int32, int32, a, 2, 2, NOP_N, NOP_N -PACK_2CH int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N -PACK_2CH int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N -PACK_2CH int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N -PACK_2CH int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N - -UNPACK_2CH int16, int16, u, 1, 1, NOP_N, NOP_N -UNPACK_2CH int16, int16, a, 1, 1, NOP_N, NOP_N -UNPACK_2CH int32, int32, u, 2, 2, NOP_N, NOP_N -UNPACK_2CH int32, int32, a, 2, 2, NOP_N, NOP_N -UNPACK_2CH int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N -UNPACK_2CH int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N -UNPACK_2CH int16, int32, u, 1, 2, INT32_TO_INT16_N, NOP_N -UNPACK_2CH int16, int32, a, 1, 2, INT32_TO_INT16_N, NOP_N - -CONV float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -CONV float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -CONV int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -CONV int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -CONV float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -CONV float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -CONV int16, float, u, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT -CONV int16, float, a, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT - -PACK_2CH float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -PACK_2CH float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -PACK_2CH int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -PACK_2CH int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -PACK_2CH float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -PACK_2CH float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -PACK_2CH int16, float, u, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT -PACK_2CH int16, float, a, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT - -UNPACK_2CH float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -UNPACK_2CH float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -UNPACK_2CH int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -UNPACK_2CH int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -UNPACK_2CH float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -UNPACK_2CH float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -UNPACK_2CH int16, float, u, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT -UNPACK_2CH int16, float, a, 1, 2, FLOAT_TO_INT16_N, FLOAT_TO_INT16_INIT - - -INIT_XMM ssse3 -UNPACK_2CH int16, int16, u, 1, 1, NOP_N, NOP_N -UNPACK_2CH int16, int16, a, 1, 1, NOP_N, NOP_N -UNPACK_2CH int32, int16, u, 2, 1, INT16_TO_INT32_N, NOP_N -UNPACK_2CH int32, int16, a, 2, 1, INT16_TO_INT32_N, NOP_N -UNPACK_2CH float, int16, u, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT -UNPACK_2CH float, int16, a, 2, 1, INT16_TO_FLOAT_N, INT16_TO_FLOAT_INIT - -INIT_XMM sse4 -PACK_6CH float, float, u, 2, 2, NOP_N, NOP_N -PACK_6CH float, float, a, 2, 2, NOP_N, NOP_N - -PACK_6CH float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -PACK_6CH float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -PACK_6CH int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -PACK_6CH int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT - -%if HAVE_AVX_EXTERNAL -INIT_XMM avx -PACK_6CH float, float, u, 2, 2, NOP_N, NOP_N -PACK_6CH float, float, a, 2, 2, NOP_N, NOP_N - -PACK_6CH float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -PACK_6CH float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -PACK_6CH int32, float, u, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT -PACK_6CH int32, float, a, 2, 2, FLOAT_TO_INT32_N, FLOAT_TO_INT32_INIT - -INIT_YMM avx -CONV float, int32, u, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -CONV float, int32, a, 2, 2, INT32_TO_FLOAT_N, INT32_TO_FLOAT_INIT -%endif diff --git a/ffmpeg1/libswresample/x86/rematrix.asm b/ffmpeg1/libswresample/x86/rematrix.asm deleted file mode 100644 index 84448e8..0000000 --- a/ffmpeg1/libswresample/x86/rematrix.asm +++ /dev/null @@ -1,251 +0,0 @@ -;****************************************************************************** -;* Copyright (c) 2012 Michael Niedermayer -;* -;* 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 -align 32 -dw1: times 8 dd 1 -w1 : times 16 dw 1 - -SECTION .text - -%macro MIX2_FLT 1 -cglobal mix_2_1_%1_float, 7, 7, 6, out, in1, in2, coeffp, index1, index2, len -%ifidn %1, a - test in1q, mmsize-1 - jne mix_2_1_float_u_int %+ SUFFIX - test in2q, mmsize-1 - jne mix_2_1_float_u_int %+ SUFFIX - test outq, mmsize-1 - jne mix_2_1_float_u_int %+ SUFFIX -%else -mix_2_1_float_u_int %+ SUFFIX -%endif - VBROADCASTSS m4, [coeffpq + 4*index1q] - VBROADCASTSS m5, [coeffpq + 4*index2q] - shl lend , 2 - add in1q , lenq - add in2q , lenq - add outq , lenq - neg lenq -.next: -%ifidn %1, a - mulps m0, m4, [in1q + lenq ] - mulps m1, m5, [in2q + lenq ] - mulps m2, m4, [in1q + lenq + mmsize] - mulps m3, m5, [in2q + lenq + mmsize] -%else - movu m0, [in1q + lenq ] - movu m1, [in2q + lenq ] - movu m2, [in1q + lenq + mmsize] - movu m3, [in2q + lenq + mmsize] - mulps m0, m0, m4 - mulps m1, m1, m5 - mulps m2, m2, m4 - mulps m3, m3, m5 -%endif - addps m0, m0, m1 - addps m2, m2, m3 - mov%1 [outq + lenq ], m0 - mov%1 [outq + lenq + mmsize], m2 - add lenq, mmsize*2 - jl .next - REP_RET -%endmacro - -%macro MIX1_FLT 1 -cglobal mix_1_1_%1_float, 5, 5, 3, out, in, coeffp, index, len -%ifidn %1, a - test inq, mmsize-1 - jne mix_1_1_float_u_int %+ SUFFIX - test outq, mmsize-1 - jne mix_1_1_float_u_int %+ SUFFIX -%else -mix_1_1_float_u_int %+ SUFFIX -%endif - VBROADCASTSS m2, [coeffpq + 4*indexq] - shl lenq , 2 - add inq , lenq - add outq , lenq - neg lenq -.next: -%ifidn %1, a - mulps m0, m2, [inq + lenq ] - mulps m1, m2, [inq + lenq + mmsize] -%else - movu m0, [inq + lenq ] - movu m1, [inq + lenq + mmsize] - mulps m0, m0, m2 - mulps m1, m1, m2 -%endif - mov%1 [outq + lenq ], m0 - mov%1 [outq + lenq + mmsize], m1 - add lenq, mmsize*2 - jl .next - REP_RET -%endmacro - -%macro MIX1_INT16 1 -cglobal mix_1_1_%1_int16, 5, 5, 6, out, in, coeffp, index, len -%ifidn %1, a - test inq, mmsize-1 - jne mix_1_1_int16_u_int %+ SUFFIX - test outq, mmsize-1 - jne mix_1_1_int16_u_int %+ SUFFIX -%else -mix_1_1_int16_u_int %+ SUFFIX -%endif - movd m4, [coeffpq + 4*indexq] - SPLATW m5, m4 - psllq m4, 32 - psrlq m4, 48 - mova m0, [w1] - psllw m0, m4 - psrlw m0, 1 - punpcklwd m5, m0 - add lenq , lenq - add inq , lenq - add outq , lenq - neg lenq -.next: - mov%1 m0, [inq + lenq ] - mov%1 m2, [inq + lenq + mmsize] - mova m1, m0 - mova m3, m2 - punpcklwd m0, [w1] - punpckhwd m1, [w1] - punpcklwd m2, [w1] - punpckhwd m3, [w1] - pmaddwd m0, m5 - pmaddwd m1, m5 - pmaddwd m2, m5 - pmaddwd m3, m5 - psrad m0, m4 - psrad m1, m4 - psrad m2, m4 - psrad m3, m4 - packssdw m0, m1 - packssdw m2, m3 - mov%1 [outq + lenq ], m0 - mov%1 [outq + lenq + mmsize], m2 - add lenq, mmsize*2 - jl .next -%if mmsize == 8 - emms - RET -%else - REP_RET -%endif -%endmacro - -%macro MIX2_INT16 1 -cglobal mix_2_1_%1_int16, 7, 7, 8, out, in1, in2, coeffp, index1, index2, len -%ifidn %1, a - test in1q, mmsize-1 - jne mix_2_1_int16_u_int %+ SUFFIX - test in2q, mmsize-1 - jne mix_2_1_int16_u_int %+ SUFFIX - test outq, mmsize-1 - jne mix_2_1_int16_u_int %+ SUFFIX -%else -mix_2_1_int16_u_int %+ SUFFIX -%endif - movd m4, [coeffpq + 4*index1q] - movd m6, [coeffpq + 4*index2q] - SPLATW m5, m4 - SPLATW m6, m6 - psllq m4, 32 - psrlq m4, 48 - mova m7, [dw1] - pslld m7, m4 - psrld m7, 1 - punpcklwd m5, m6 - add lend , lend - add in1q , lenq - add in2q , lenq - add outq , lenq - neg lenq -.next: - mov%1 m0, [in1q + lenq ] - mov%1 m2, [in2q + lenq ] - mova m1, m0 - punpcklwd m0, m2 - punpckhwd m1, m2 - - mov%1 m2, [in1q + lenq + mmsize] - mov%1 m6, [in2q + lenq + mmsize] - mova m3, m2 - punpcklwd m2, m6 - punpckhwd m3, m6 - - pmaddwd m0, m5 - pmaddwd m1, m5 - pmaddwd m2, m5 - pmaddwd m3, m5 - paddd m0, m7 - paddd m1, m7 - paddd m2, m7 - paddd m3, m7 - psrad m0, m4 - psrad m1, m4 - psrad m2, m4 - psrad m3, m4 - packssdw m0, m1 - packssdw m2, m3 - mov%1 [outq + lenq ], m0 - mov%1 [outq + lenq + mmsize], m2 - add lenq, mmsize*2 - jl .next -%if mmsize == 8 - emms - RET -%else - REP_RET -%endif -%endmacro - - -INIT_MMX mmx -MIX1_INT16 u -MIX1_INT16 a -MIX2_INT16 u -MIX2_INT16 a - -INIT_XMM sse -MIX2_FLT u -MIX2_FLT a -MIX1_FLT u -MIX1_FLT a - -INIT_XMM sse2 -MIX1_INT16 u -MIX1_INT16 a -MIX2_INT16 u -MIX2_INT16 a - -%if HAVE_AVX_EXTERNAL -INIT_YMM avx -MIX2_FLT u -MIX2_FLT a -MIX1_FLT u -MIX1_FLT a -%endif diff --git a/ffmpeg1/libswresample/x86/resample_mmx.h b/ffmpeg1/libswresample/x86/resample_mmx.h deleted file mode 100644 index d96fd5a..0000000 --- a/ffmpeg1/libswresample/x86/resample_mmx.h +++ /dev/null @@ -1,70 +0,0 @@ -/* - * Copyright (c) 2012 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 "libavutil/x86/asm.h" -#include "libavutil/cpu.h" -#include "libswresample/swresample_internal.h" - -int swri_resample_int16_mmx2 (struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx); -int swri_resample_int16_ssse3(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx); - -DECLARE_ALIGNED(16, const uint64_t, ff_resample_int16_rounder)[2] = { 0x0000000000004000ULL, 0x0000000000000000ULL}; - -#define COMMON_CORE_INT16_MMX2 \ - x86_reg len= -2*c->filter_length;\ -__asm__ volatile(\ - "movq "MANGLE(ff_resample_int16_rounder)", %%mm0 \n\t"\ - "1: \n\t"\ - "movq (%1, %0), %%mm1 \n\t"\ - "pmaddwd (%2, %0), %%mm1 \n\t"\ - "paddd %%mm1, %%mm0 \n\t"\ - "add $8, %0 \n\t"\ - " js 1b \n\t"\ - "pshufw $0x0E, %%mm0, %%mm1 \n\t"\ - "paddd %%mm1, %%mm0 \n\t"\ - "psrad $15, %%mm0 \n\t"\ - "packssdw %%mm0, %%mm0 \n\t"\ - "movd %%mm0, (%3) \n\t"\ - : "+r" (len)\ - : "r" (((uint8_t*)(src+sample_index))-len),\ - "r" (((uint8_t*)filter)-len),\ - "r" (dst+dst_index)\ -); - -#define COMMON_CORE_INT16_SSSE3 \ - x86_reg len= -2*c->filter_length;\ -__asm__ volatile(\ - "movdqa "MANGLE(ff_resample_int16_rounder)", %%xmm0 \n\t"\ - "1: \n\t"\ - "movdqu (%1, %0), %%xmm1 \n\t"\ - "pmaddwd (%2, %0), %%xmm1 \n\t"\ - "paddd %%xmm1, %%xmm0 \n\t"\ - "add $16, %0 \n\t"\ - " js 1b \n\t"\ - "phaddd %%xmm0, %%xmm0 \n\t"\ - "phaddd %%xmm0, %%xmm0 \n\t"\ - "psrad $15, %%xmm0 \n\t"\ - "packssdw %%xmm0, %%xmm0 \n\t"\ - "movd %%xmm0, (%3) \n\t"\ - : "+r" (len)\ - : "r" (((uint8_t*)(src+sample_index))-len),\ - "r" (((uint8_t*)filter)-len),\ - "r" (dst+dst_index)\ -); diff --git a/ffmpeg1/libswresample/x86/swresample_x86.c b/ffmpeg1/libswresample/x86/swresample_x86.c deleted file mode 100644 index e18f0c5..0000000 --- a/ffmpeg1/libswresample/x86/swresample_x86.c +++ /dev/null @@ -1,195 +0,0 @@ -/* - * Copyright (C) 2012 Michael Niedermayer (michaelni@gmx.at) - * - * This file is part of libswresample - * - * libswresample 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. - * - * libswresample 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 libswresample; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include "libswresample/swresample_internal.h" -#include "libswresample/audioconvert.h" - -#define PROTO(pre, in, out, cap) void ff ## pre ## _ ##in## _to_ ##out## _a_ ##cap(uint8_t **dst, const uint8_t **src, int len); -#define PROTO2(pre, out, cap) PROTO(pre, int16, out, cap) PROTO(pre, int32, out, cap) PROTO(pre, float, out, cap) -#define PROTO3(pre, cap) PROTO2(pre, int16, cap) PROTO2(pre, int32, cap) PROTO2(pre, float, cap) -#define PROTO4(pre) PROTO3(pre, mmx) PROTO3(pre, sse) PROTO3(pre, sse2) PROTO3(pre, ssse3) PROTO3(pre, sse4) PROTO3(pre, avx) -PROTO4() -PROTO4(_pack_2ch) -PROTO4(_pack_6ch) -PROTO4(_unpack_2ch) - -av_cold void swri_audio_convert_init_x86(struct AudioConvert *ac, - enum AVSampleFormat out_fmt, - enum AVSampleFormat in_fmt, - int channels){ - int mm_flags = av_get_cpu_flags(); - - ac->simd_f= NULL; - -//FIXME add memcpy case - -#define MULTI_CAPS_FUNC(flag, cap) \ - if (mm_flags & flag) {\ - if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S16 || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S16P)\ - ac->simd_f = ff_int16_to_int32_a_ ## cap;\ - if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_S32 || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S32P)\ - ac->simd_f = ff_int32_to_int16_a_ ## cap;\ - } - -MULTI_CAPS_FUNC(AV_CPU_FLAG_MMX, mmx) -MULTI_CAPS_FUNC(AV_CPU_FLAG_SSE2, sse2) - - if(mm_flags & AV_CPU_FLAG_MMX) { - if(channels == 6) { - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_6ch_float_to_float_a_mmx; - } - } - - if(mm_flags & AV_CPU_FLAG_SSE2) { - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32 || out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_int32_to_float_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S16 || out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S16P) - ac->simd_f = ff_int16_to_float_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = ff_float_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = ff_float_to_int16_a_sse2; - - if(channels == 2) { - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_2ch_int32_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_S16P) - ac->simd_f = ff_pack_2ch_int16_to_int16_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S16P) - ac->simd_f = ff_pack_2ch_int16_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_2ch_int32_to_int16_a_sse2; - - if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_FLT || out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S32) - ac->simd_f = ff_unpack_2ch_int32_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S16) - ac->simd_f = ff_unpack_2ch_int16_to_int16_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S16) - ac->simd_f = ff_unpack_2ch_int16_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S32) - ac->simd_f = ff_unpack_2ch_int32_to_int16_a_sse2; - - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_2ch_int32_to_float_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = ff_pack_2ch_float_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S16P) - ac->simd_f = ff_pack_2ch_int16_to_float_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16 && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = ff_pack_2ch_float_to_int16_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32) - ac->simd_f = ff_unpack_2ch_int32_to_float_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_FLT) - ac->simd_f = ff_unpack_2ch_float_to_int32_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S16) - ac->simd_f = ff_unpack_2ch_int16_to_float_a_sse2; - if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_FLT) - ac->simd_f = ff_unpack_2ch_float_to_int16_a_sse2; - } - } - if(mm_flags & AV_CPU_FLAG_SSSE3) { - if(channels == 2) { - if( out_fmt == AV_SAMPLE_FMT_S16P && in_fmt == AV_SAMPLE_FMT_S16) - ac->simd_f = ff_unpack_2ch_int16_to_int16_a_ssse3; - if( out_fmt == AV_SAMPLE_FMT_S32P && in_fmt == AV_SAMPLE_FMT_S16) - ac->simd_f = ff_unpack_2ch_int16_to_int32_a_ssse3; - if( out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S16) - ac->simd_f = ff_unpack_2ch_int16_to_float_a_ssse3; - } - } - if(mm_flags & AV_CPU_FLAG_SSE4) { - if(channels == 6) { - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_6ch_float_to_float_a_sse4; - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_6ch_int32_to_float_a_sse4; - if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = ff_pack_6ch_float_to_int32_a_sse4; - } - } - if(HAVE_AVX_EXTERNAL && mm_flags & AV_CPU_FLAG_AVX) { - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32 || out_fmt == AV_SAMPLE_FMT_FLTP && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_int32_to_float_a_avx; - if(channels == 6) { - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_FLTP || out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_6ch_float_to_float_a_avx; - if( out_fmt == AV_SAMPLE_FMT_FLT && in_fmt == AV_SAMPLE_FMT_S32P) - ac->simd_f = ff_pack_6ch_int32_to_float_a_avx; - if( out_fmt == AV_SAMPLE_FMT_S32 && in_fmt == AV_SAMPLE_FMT_FLTP) - ac->simd_f = ff_pack_6ch_float_to_int32_a_avx; - } - } -} - -#define D(type, simd) \ -mix_1_1_func_type ff_mix_1_1_a_## type ## _ ## simd;\ -mix_2_1_func_type ff_mix_2_1_a_## type ## _ ## simd; - -D(float, sse) -D(float, avx) -D(int16, mmx) -D(int16, sse2) - - -av_cold void swri_rematrix_init_x86(struct SwrContext *s){ - int mm_flags = av_get_cpu_flags(); - int nb_in = av_get_channel_layout_nb_channels(s->in_ch_layout); - int nb_out = av_get_channel_layout_nb_channels(s->out_ch_layout); - int num = nb_in * nb_out; - int i,j; - - s->mix_1_1_simd = NULL; - s->mix_2_1_simd = NULL; - - if (s->midbuf.fmt == AV_SAMPLE_FMT_S16P){ - if(mm_flags & AV_CPU_FLAG_MMX) { - s->mix_1_1_simd = ff_mix_1_1_a_int16_mmx; - s->mix_2_1_simd = ff_mix_2_1_a_int16_mmx; - } - if(mm_flags & AV_CPU_FLAG_SSE2) { - s->mix_1_1_simd = ff_mix_1_1_a_int16_sse2; - s->mix_2_1_simd = ff_mix_2_1_a_int16_sse2; - } - s->native_simd_matrix = av_mallocz(2 * num * sizeof(int16_t)); - for(i=0; i<nb_out; i++){ - int sh = 0; - for(j=0; j<nb_in; j++) - sh = FFMAX(sh, FFABS(((int*)s->native_matrix)[i * nb_in + j])); - sh = FFMAX(av_log2(sh) - 14, 0); - for(j=0; j<nb_in; j++) { - ((int16_t*)s->native_simd_matrix)[2*(i * nb_in + j)+1] = 15 - sh; - ((int16_t*)s->native_simd_matrix)[2*(i * nb_in + j)] = - ((((int*)s->native_matrix)[i * nb_in + j]) + (1<<sh>>1)) >> sh; - } - } - } else if(s->midbuf.fmt == AV_SAMPLE_FMT_FLTP){ - if(mm_flags & AV_CPU_FLAG_SSE) { - s->mix_1_1_simd = ff_mix_1_1_a_float_sse; - s->mix_2_1_simd = ff_mix_2_1_a_float_sse; - } - if(HAVE_AVX_EXTERNAL && mm_flags & AV_CPU_FLAG_AVX) { - s->mix_1_1_simd = ff_mix_1_1_a_float_avx; - s->mix_2_1_simd = ff_mix_2_1_a_float_avx; - } - s->native_simd_matrix = av_mallocz(num * sizeof(float)); - memcpy(s->native_simd_matrix, s->native_matrix, num * sizeof(float)); - } -} |