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-rw-r--r--libswresample/Makefile15
-rw-r--r--libswresample/arm/Makefile2
-rw-r--r--libswresample/arm/audio_convert_init.c65
-rw-r--r--libswresample/arm/audio_convert_neon.S363
-rw-r--r--libswresample/audioconvert.c208
-rw-r--r--libswresample/audioconvert.h77
-rw-r--r--libswresample/dither.c87
-rw-r--r--libswresample/libswresample.v4
-rw-r--r--libswresample/rematrix.c497
-rw-r--r--libswresample/rematrix_template.c74
-rw-r--r--libswresample/resample.c459
-rw-r--r--libswresample/resample_template.c139
-rw-r--r--libswresample/swresample-test.c410
-rw-r--r--libswresample/swresample.c795
-rw-r--r--libswresample/swresample.h220
-rw-r--r--libswresample/swresample_internal.h142
-rw-r--r--libswresample/version.h45
-rw-r--r--libswresample/x86/Makefile3
-rw-r--r--libswresample/x86/audio_convert.asm462
-rw-r--r--libswresample/x86/rematrix.asm252
-rw-r--r--libswresample/x86/resample_mmx.h70
-rw-r--r--libswresample/x86/swresample_x86.c195
22 files changed, 4584 insertions, 0 deletions
diff --git a/libswresample/Makefile b/libswresample/Makefile
new file mode 100644
index 0000000000..d85ed8be88
--- /dev/null
+++ b/libswresample/Makefile
@@ -0,0 +1,15 @@
+include $(SUBDIR)../config.mak
+
+NAME = swresample
+FFLIBS = avutil
+
+HEADERS = swresample.h \
+ version.h \
+
+OBJS = audioconvert.o \
+ dither.o \
+ rematrix.o \
+ resample.o \
+ swresample.o \
+
+TESTPROGS = swresample
diff --git a/libswresample/arm/Makefile b/libswresample/arm/Makefile
new file mode 100644
index 0000000000..55683cbfae
--- /dev/null
+++ b/libswresample/arm/Makefile
@@ -0,0 +1,2 @@
+OBJS += arm/audio_convert_init.o
+NEON-OBJS += arm/audio_convert_neon.o
diff --git a/libswresample/arm/audio_convert_init.c b/libswresample/arm/audio_convert_init.c
new file mode 100644
index 0000000000..9fdb174608
--- /dev/null
+++ b/libswresample/arm/audio_convert_init.c
@@ -0,0 +1,65 @@
+/*
+ * 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;
+ }
+}
diff --git a/libswresample/arm/audio_convert_neon.S b/libswresample/arm/audio_convert_neon.S
new file mode 100644
index 0000000000..471a2d864f
--- /dev/null
+++ b/libswresample/arm/audio_convert_neon.S
@@ -0,0 +1,363 @@
+/*
+ * 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/libswresample/audioconvert.c b/libswresample/audioconvert.c
new file mode 100644
index 0000000000..d2e3722a40
--- /dev/null
+++ b/libswresample/audioconvert.c
@@ -0,0 +1,208 @@
+/*
+ * 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;
+
+ av_assert0(ctx->channels == out->ch_count);
+
+ //FIXME optimize common cases
+
+ if(ctx->simd_f && !ctx->ch_map){
+ 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/libswresample/audioconvert.h b/libswresample/audioconvert.h
new file mode 100644
index 0000000000..9a234d4b93
--- /dev/null
+++ b/libswresample/audioconvert.h
@@ -0,0 +1,77 @@
+/*
+ * 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"
+#include "libavutil/audioconvert.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;
+ 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/libswresample/dither.c b/libswresample/dither.c
new file mode 100644
index 0000000000..79113f4c23
--- /dev/null
+++ b/libswresample/dither.c
@@ -0,0 +1,87 @@
+/*
+ * 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 "libavutil/avassert.h"
+#include "swresample_internal.h"
+
+void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt) {
+ double scale = 0;
+#define TMP_EXTRA 2
+ double *tmp = av_malloc((len + TMP_EXTRA) * sizeof(double));
+ int i;
+
+ 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;
+
+ 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;
+ case SWR_DITHER_TRIANGULAR :
+ case SWR_DITHER_TRIANGULAR_HIGHPASS :
+ v = ((double)seed) / UINT_MAX;
+ seed = seed*1664525 + 1013904223;
+ v-= ((double)seed) / UINT_MAX;
+ break;
+ default: av_assert0(0);
+ }
+ tmp[i] = v;
+ }
+
+ for(i=0; i<len; i++){
+ double v;
+
+ switch(s->dither_method){
+ case SWR_DITHER_RECTANGULAR:
+ case SWR_DITHER_TRIANGULAR :
+ v = tmp[i];
+ break;
+ case SWR_DITHER_TRIANGULAR_HIGHPASS :
+ v = (- tmp[i] + 2*tmp[i+1] - tmp[i+2]) / sqrt(6);
+ break;
+ default: av_assert0(0);
+ }
+
+ v*= scale;
+
+ switch(in_fmt){
+ case AV_SAMPLE_FMT_S16: ((int16_t*)dst)[i] = v; break;
+ case AV_SAMPLE_FMT_S32: ((int32_t*)dst)[i] = v; break;
+ case AV_SAMPLE_FMT_FLT: ((float *)dst)[i] = v; break;
+ case AV_SAMPLE_FMT_DBL: ((double *)dst)[i] = v; break;
+ default: av_assert0(0);
+ }
+ }
+
+ av_free(tmp);
+}
diff --git a/libswresample/libswresample.v b/libswresample/libswresample.v
new file mode 100644
index 0000000000..9b797bde77
--- /dev/null
+++ b/libswresample/libswresample.v
@@ -0,0 +1,4 @@
+LIBSWRESAMPLE_$MAJOR {
+ global: swr_*; ff_*; swresample_*;
+ local: *;
+};
diff --git a/libswresample/rematrix.c b/libswresample/rematrix.c
new file mode 100644
index 0000000000..72da438e6c
--- /dev/null
+++ b/libswresample/rematrix.c
@@ -0,0 +1,497 @@
+/*
+ * 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/audioconvert.h"
+#include "libavutil/avassert.h"
+
+#define ONE (1.0)
+#define R(x) x
+#define SAMPLE float
+#define COEFF float
+#define INTER float
+#define RENAME(x) x ## _float
+#include "rematrix_template.c"
+#undef SAMPLE
+#undef RENAME
+#undef R
+#undef ONE
+#undef COEFF
+#undef INTER
+
+#define ONE (1.0)
+#define R(x) x
+#define SAMPLE double
+#define COEFF double
+#define INTER double
+#define RENAME(x) x ## _double
+#include "rematrix_template.c"
+#undef SAMPLE
+#undef RENAME
+#undef R
+#undef ONE
+#undef COEFF
+#undef INTER
+
+#define ONE (-32768)
+#define R(x) (((x) + 16384)>>15)
+#define SAMPLE int16_t
+#define COEFF int
+#define INTER int
+#define RENAME(x) x ## _s16
+#include "rematrix_template.c"
+
+
+#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 & (1LL<<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/libswresample/rematrix_template.c b/libswresample/rematrix_template.c
new file mode 100644
index 0000000000..627bf1f6fc
--- /dev/null
+++ b/libswresample/rematrix_template.c
@@ -0,0 +1,74 @@
+/*
+ * 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
+ */
+
+typedef void (RENAME(mix_any_func_type))(SAMPLE **out, const SAMPLE **in1, COEFF *coeffp, int len);
+
+static void RENAME(sum2)(SAMPLE *out, const SAMPLE *in1, const SAMPLE *in2, COEFF *coeffp, int index1, int index2, int 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, int index, int 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, int 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, int 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;
+}
diff --git a/libswresample/resample.c b/libswresample/resample.c
new file mode 100644
index 0000000000..bf412dd268
--- /dev/null
+++ b/libswresample/resample.c
@@ -0,0 +1,459 @@
+/*
+ * 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_clip(lrintf(tab[i] * scale / norm), INT32_MIN, INT32_MAX);
+ 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;
+}
+
+ResampleContext *swri_resample_init(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 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_mallocz(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;
+}
+
+void swri_resample_free(ResampleContext **c){
+ if(!*c)
+ return;
+ av_freep(&(*c)->filter_bank);
+ av_freep(c);
+}
+
+int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance){
+ ResampleContext *c;
+ 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;
+ }
+ c= s->resample;
+ c->compensation_distance= compensation_distance;
+ if (compensation_distance)
+ c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
+ else
+ c->dst_incr = c->ideal_dst_incr;
+ return 0;
+}
+
+#define RENAME(N) N ## _int16
+#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
+#include "resample_template.c"
+
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+
+#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
+#include "resample_template.c"
+
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+
+#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
+#include "resample_template.c"
+
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+
+#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
+#include "resample_template.c"
+
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+// 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 COMMON_CORE COMMON_CORE_INT16_MMX2
+#define RENAME(N) N ## _int16_mmx2
+#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
+#include "resample_template.c"
+
+#undef COMMON_CORE
+#undef RENAME
+#undef FELEM
+#undef FELEM2
+#undef DELEM
+#undef FELEML
+#undef OUT
+#undef FELEM_MIN
+#undef FELEM_MAX
+#undef FILTER_SHIFT
+
+#if HAVE_SSSE3_INLINE
+#define COMMON_CORE COMMON_CORE_INT16_SSSE3
+#define RENAME(N) N ## _int16_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
+#include "resample_template.c"
+#endif
+#endif // HAVE_MMXEXT_INLINE
+
+int swri_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;
+}
+
+int64_t swr_get_delay(struct SwrContext *s, int64_t base){
+ ResampleContext *c = s->resample;
+ if(c){
+ 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);
+ }else{
+ return (s->in_buffer_count*base + (s->in_sample_rate>>1))/ s->in_sample_rate;
+ }
+}
diff --git a/libswresample/resample_template.c b/libswresample/resample_template.c
new file mode 100644
index 0000000000..ad840702ce
--- /dev/null
+++ b/libswresample/resample_template.c
@@ -0,0 +1,139 @@
+/*
+ * 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>
+ */
+
+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;
+ }else if(compensation_distance == 0 && !c->linear && index >= 0){
+ for(dst_index=0; dst_index < dst_size; dst_index++){
+ FELEM *filter= ((FELEM*)c->filter_bank) + c->filter_alloc*(index & c->phase_mask);
+ int sample_index= index >> c->phase_shift;
+
+ 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++;
+ }
+ }
+ }else{
+ for(dst_index=0; dst_index < dst_size; dst_index++){
+ FELEM *filter= ((FELEM*)c->filter_bank) + c->filter_alloc*(index & c->phase_mask);
+ int sample_index= index >> c->phase_shift;
+ FELEM2 val=0;
+
+ 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)] * 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(index, 0) >> c->phase_shift;
+ if(index>=0) index &= c->phase_mask;
+
+ 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;
+ }
+#if 0
+ if(update_ctx && !c->compensation_distance){
+#undef rand
+ av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
+av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
+ }
+#endif
+
+ return dst_index;
+}
diff --git a/libswresample/swresample-test.c b/libswresample/swresample-test.c
new file mode 100644
index 0000000000..4a0728c8e7
--- /dev/null
+++ b/libswresample/swresample-test.c
@@ -0,0 +1,410 @@
+/*
+ * 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/common.h"
+#include "libavutil/audioconvert.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(lrint(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(swr_init( forw_ctx) < 0)
+ fprintf(stderr, "swr_init(->) failed\n");
+ if(swr_init(backw_ctx) < 0)
+ fprintf(stderr, "swr_init(<-) failed\n");
+ if(!forw_ctx)
+ fprintf(stderr, "Failed to init forw_cts\n");
+ if(!backw_ctx)
+ fprintf(stderr, "Failed to init backw_ctx\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", sqrt(sse/out_count), 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/libswresample/swresample.c b/libswresample/swresample.c
new file mode 100644
index 0000000000..5d02aafb49
--- /dev/null
+++ b/libswresample/swresample.c
@@ -0,0 +1,795 @@
+/*
+ * 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 "libavutil/opt.h"
+#include "swresample_internal.h"
+#include "audioconvert.h"
+#include "libavutil/avassert.h"
+#include "libavutil/audioconvert.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" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
+{"in_channel_count" , "Input Channel Count" , OFFSET( in.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
+{"och" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
+{"out_channel_count" , "Output Channel Count" , OFFSET(out.ch_count ), AV_OPT_TYPE_INT , {.i64=2 }, 0 , SWR_CH_MAX, PARAM},
+{"uch" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
+{"used_channel_count" , "Used Channel Count" , OFFSET(used_ch_count ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_CH_MAX, PARAM},
+{"isr" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
+{"in_sample_rate" , "Input Sample Rate" , OFFSET( in_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
+{"osr" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
+{"out_sample_rate" , "Output Sample Rate" , OFFSET(out_sample_rate), AV_OPT_TYPE_INT , {.i64=0 }, 0 , INT_MAX , PARAM},
+{"isf" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
+{"in_sample_fmt" , "Input Sample Format" , OFFSET( in_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
+{"osf" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
+{"out_sample_fmt" , "Output Sample Format" , OFFSET(out_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_NB-1+256, PARAM},
+{"tsf" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
+{"internal_sample_fmt" , "Internal Sample Format" , OFFSET(int_sample_fmt ), AV_OPT_TYPE_INT , {.i64=AV_SAMPLE_FMT_NONE }, -1 , AV_SAMPLE_FMT_FLTP, PARAM},
+{"icl" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
+{"in_channel_layout" , "Input Channel Layout" , OFFSET( in_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
+{"ocl" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
+{"out_channel_layout" , "Output Channel Layout" , OFFSET(out_ch_layout ), AV_OPT_TYPE_INT64, {.i64=0 }, 0 , INT64_MAX , PARAM, "channel_layout"},
+{"clev" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
+{"center_mix_level" , "Center Mix Level" , OFFSET(clev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
+{"slev" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
+{"surround_mix_level" , "Sourround Mix Level" , OFFSET(slev ), AV_OPT_TYPE_FLOAT, {.dbl=C_30DB }, -32 , 32 , PARAM},
+{"lfe_mix_level" , "LFE Mix Level" , OFFSET(lfe_mix_level ), AV_OPT_TYPE_FLOAT, {.dbl=0 }, -32 , 32 , PARAM},
+{"rmvol" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
+{"rematrix_volume" , "Rematrix Volume" , OFFSET(rematrix_volume), AV_OPT_TYPE_FLOAT, {.dbl=1.0 }, -1000 , 1000 , PARAM},
+{"flags" , NULL , OFFSET(flags ), AV_OPT_TYPE_FLAGS, {.i64=0 }, 0 , UINT_MAX , PARAM, "flags"},
+{"swr_flags" , NULL , 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" , "Dither Scale" , OFFSET(dither_scale ), AV_OPT_TYPE_FLOAT, {.dbl=1 }, 0 , INT_MAX , PARAM},
+{"dither_method" , "Dither Method" , OFFSET(dither_method ), AV_OPT_TYPE_INT , {.i64=0 }, 0 , SWR_DITHER_NB-1, PARAM, "dither_method"},
+{"rectangular" , "Rectangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_RECTANGULAR}, INT_MIN, INT_MAX , PARAM, "dither_method"},
+{"triangular" , "Triangular Dither" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR }, INT_MIN, INT_MAX , PARAM, "dither_method"},
+{"triangular_hp" , "Triangular Dither With High Pass" , 0 , AV_OPT_TYPE_CONST, {.i64=SWR_DITHER_TRIANGULAR_HIGHPASS }, INT_MIN, INT_MAX, PARAM, "dither_method"},
+{"filter_size" , "Resampling Filter Size" , OFFSET(filter_size) , AV_OPT_TYPE_INT , {.i64=16 }, 0 , INT_MAX , PARAM },
+{"phase_shift" , "Resampling Phase Shift" , OFFSET(phase_shift) , AV_OPT_TYPE_INT , {.i64=10 }, 0 , 30 , PARAM },
+{"linear_interp" , "Use Linear Interpolation" , OFFSET(linear_interp) , AV_OPT_TYPE_INT , {.i64=0 }, 0 , 1 , PARAM },
+{"cutoff" , "Cutoff Frequency Ratio" , OFFSET(cutoff) , AV_OPT_TYPE_DOUBLE,{.dbl=0.8 }, 0 , 1 , PARAM },
+{"min_comp" , "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" , "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" , "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" , "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 },
+{ "matrix_encoding" , "Matrixed Stereo Encoding" , OFFSET(matrix_encoding), AV_OPT_TYPE_INT ,{.i64 = AV_MATRIX_ENCODING_NONE}, AV_MATRIX_ENCODING_NONE, AV_MATRIX_ENCODING_NB-1, PARAM, "matrix_encoding" },
+ { "none", "None", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
+ { "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
+ { "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { .i64 = AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
+{ "filter_type" , "Filter Type" , OFFSET(filter_type) , AV_OPT_TYPE_INT , { .i64 = SWR_FILTER_TYPE_KAISER }, SWR_FILTER_TYPE_CUBIC, SWR_FILTER_TYPE_KAISER, PARAM, "filter_type" },
+ { "cubic" , "Cubic" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+ { "kaiser" , "Kaiser Windowed Sinc" , 0 , AV_OPT_TYPE_CONST, { .i64 = SWR_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
+{ "kaiser_beta" , "Kaiser Window Beta" ,OFFSET(kaiser_beta) , AV_OPT_TYPE_INT , {.i64=9 }, 2 , 16 , PARAM },
+
+{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->dither);
+ swri_audio_convert_free(&s-> in_convert);
+ swri_audio_convert_free(&s->out_convert);
+ swri_audio_convert_free(&s->full_convert);
+ swri_resample_free(&s->resample);
+ swri_rematrix_free(s);
+ }
+
+ av_freep(ss);
+}
+
+av_cold int swr_init(struct SwrContext *s){
+ 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->dither);
+ 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(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_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->out_sample_rate!=s->in_sample_rate || (s->flags & SWR_FLAG_RESAMPLE)){
+ s->resample = swri_resample_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);
+ }else
+ swri_resample_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;
+ }
+
+ 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;
+
+#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) {
+ av_log(s, AV_LOG_ERROR, "Rematrix is needed but there is not enough information to do it\n");
+ 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;
+
+ 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);
+
+
+ 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);
+ }
+
+ s->dither = s->preout;
+
+ if(s->rematrix || s->dither_method)
+ return swri_rematrix_init(s);
+
+ return 0;
+}
+
+static int 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= swri_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(in_count && !s->in_buffer_count){
+ s->in_buffer_index=0;
+ ret= swri_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=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=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=realloc_audio(&s->midbuf, out_count))<0)
+ return ret;
+ }else{
+ av_assert0(s->midbuf.ch_count == s->out.ch_count);
+ if((ret=realloc_audio(&s->midbuf, in_count))<0)
+ return ret;
+ }
+ if((ret=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){
+ if(s->dither_method){
+ int ch;
+ int dither_count= FFMAX(out_count, 1<<16);
+ av_assert0(preout != in);
+
+ if((ret=realloc_audio(&s->dither, dither_count))<0)
+ return ret;
+ if(ret)
+ for(ch=0; ch<s->dither.ch_count; ch++)
+ swri_get_dither(s, s->dither.ch[ch], s->dither.count, 12345678913579<<ch, s->out_sample_fmt, s->int_sample_fmt);
+ av_assert0(s->dither.ch_count == preout->ch_count);
+
+ if(s->dither_pos + out_count > s->dither.count)
+ s->dither_pos = 0;
+
+ for(ch=0; ch<preout->ch_count; ch++)
+ s->mix_2_1_f(preout->ch[ch], preout->ch[ch], s->dither.ch[ch] + s->dither.bps * s->dither_pos, s->native_one, 0, 0, out_count);
+
+ s->dither_pos += out_count;
+ }
+//FIXME packed doesnt need more than 1 chan here!
+ swri_audio_convert(s->out_convert, out, preout, 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;
+
+ if(s->drop_output > 0){
+ int ret;
+ AudioData tmp = s->out;
+ uint8_t *tmp_arg[SWR_CH_MAX];
+ tmp.count = 0;
+ tmp.data = NULL;
+ if((ret=realloc_audio(&tmp, s->drop_output))<0)
+ return ret;
+
+ reversefill_audiodata(&tmp, tmp_arg);
+ s->drop_output *= -1; //FIXME find a less hackish solution
+ ret = swr_convert(s, tmp_arg, -s->drop_output, in_arg, in_count); //FIXME optimize but this is as good as never called so maybe it doesnt matter
+ s->drop_output *= -1;
+ if(ret>0)
+ s->drop_output -= ret;
+
+ av_freep(&tmp.data);
+ if(s->drop_output || !out_arg)
+ return 0;
+ in_count = 0;
+ }
+
+ if(!in_arg){
+ if(s->in_buffer_count){
+ if (s->resample && !s->flushed) {
+ AudioData *a= &s->in_buffer;
+ int i, j, ret;
+ if((ret=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;
+ s->resample_in_constraint = 0;
+ s->flushed = 1;
+ }
+ }else{
+ 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=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;
+ AudioData silence = s->in;
+ uint8_t *tmp_arg[SWR_CH_MAX];
+
+ if(count <= 0)
+ return 0;
+
+ silence.count = 0;
+ silence.data = NULL;
+ if((ret=realloc_audio(&silence, count))<0)
+ return ret;
+
+ if(silence.planar) for(i=0; i<silence.ch_count; i++) {
+ memset(silence.ch[i], silence.bps==1 ? 0x80 : 0, count*silence.bps);
+ } else
+ memset(silence.ch[0], silence.bps==1 ? 0x80 : 0, count*silence.bps*silence.ch_count);
+
+ reversefill_audiodata(&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);
+ av_freep(&silence.data);
+ return ret;
+}
+
+int64_t swr_next_pts(struct SwrContext *s, int64_t pts){
+ if(pts == INT64_MIN)
+ return s->outpts;
+ 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;
+ double fdelta = delta /(double)(s->in_sample_rate * (int64_t)s->out_sample_rate);
+
+ if(fabs(fdelta) > s->min_compensation) {
+ if(!s->outpts || 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/libswresample/swresample.h b/libswresample/swresample.h
new file mode 100644
index 0000000000..ac43cd25c4
--- /dev/null
+++ b/libswresample/swresample.h
@@ -0,0 +1,220 @@
+/*
+ * 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
+ */
+
+/**
+ * @file
+ * libswresample public header
+ */
+
+#ifndef SWR_H
+#define SWR_H
+
+#include <inttypes.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_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
+ * timestampe 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 timstamp for the next input sample, INT64_MIN if unknown
+ * @returns 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.
+ *
+ * @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
diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h
new file mode 100644
index 0000000000..ab17602c64
--- /dev/null
+++ b/libswresample/swresample_internal.h
@@ -0,0 +1,142 @@
+/*
+ * 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
+ */
+
+#ifndef SWR_INTERNAL_H
+#define SWR_INTERNAL_H
+
+#include "swresample.h"
+#include "libavutil/audioconvert.h"
+
+#define SQRT3_2 1.22474487139158904909 /* sqrt(3/2) */
+
+typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, int index, int len);
+typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, int index1, int index2, int len);
+
+typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, int 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 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 SwrDitherType dither_method;
+ int dither_pos;
+ float dither_scale;
+ int filter_size; /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
+ int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
+ int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
+ double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
+ enum SwrFilterType filter_type; /**< resampling filter type */
+ int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
+
+ float min_compensation; ///< minimum below which no compensation will happen
+ float min_hard_compensation; ///< minimum below which no silence inject / sample drop will happen
+ float soft_compensation_duration; ///< duration over which soft compensation is applied
+ float max_soft_compensation; ///< maximum soft compensation in seconds over soft_compensation_duration
+
+ 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 dither; ///< noise used for dithering
+ 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
+ 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
+
+ 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 */
+};
+
+struct ResampleContext *swri_resample_init(struct ResampleContext *, int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff, enum AVSampleFormat, enum SwrFilterType, int kaiser_beta);
+void swri_resample_free(struct ResampleContext **c);
+int swri_multiple_resample(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
+void swri_resample_compensate(struct ResampleContext *c, int sample_delta, int compensation_distance);
+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);
+
+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 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/libswresample/version.h b/libswresample/version.h
new file mode 100644
index 0000000000..2c7b932818
--- /dev/null
+++ b/libswresample/version.h
@@ -0,0 +1,45 @@
+/*
+ * 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 16
+#define LIBSWRESAMPLE_VERSION_MICRO 100
+
+#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/libswresample/x86/Makefile b/libswresample/x86/Makefile
new file mode 100644
index 0000000000..e8feede6b6
--- /dev/null
+++ b/libswresample/x86/Makefile
@@ -0,0 +1,3 @@
+YASM-OBJS += x86/swresample_x86.o\
+ x86/audio_convert.o\
+ x86/rematrix.o\
diff --git a/libswresample/x86/audio_convert.asm b/libswresample/x86/audio_convert.asm
new file mode 100644
index 0000000000..29701a594c
--- /dev/null
+++ b/libswresample/x86/audio_convert.asm
@@ -0,0 +1,462 @@
+;******************************************************************************
+;* 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/x86inc.asm"
+%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/libswresample/x86/rematrix.asm b/libswresample/x86/rematrix.asm
new file mode 100644
index 0000000000..d63545ff5e
--- /dev/null
+++ b/libswresample/x86/rematrix.asm
@@ -0,0 +1,252 @@
+;******************************************************************************
+;* 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/x86inc.asm"
+%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/libswresample/x86/resample_mmx.h b/libswresample/x86/resample_mmx.h
new file mode 100644
index 0000000000..d96fd5a9d2
--- /dev/null
+++ b/libswresample/x86/resample_mmx.h
@@ -0,0 +1,70 @@
+/*
+ * 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/libswresample/x86/swresample_x86.c b/libswresample/x86/swresample_x86.c
new file mode 100644
index 0000000000..e18f0c57ba
--- /dev/null
+++ b/libswresample/x86/swresample_x86.c
@@ -0,0 +1,195 @@
+/*
+ * 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));
+ }
+}
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-06 08:16:17 +0000