diff --git a/packages/multimedia/ffmpeg/patches/0.10.7/ffmpeg-0.10.6-905.01-VFP_acceleration.patch b/packages/multimedia/ffmpeg/patches/0.10.7/ffmpeg-0.10.6-905.01-VFP_acceleration.patch new file mode 100644 index 0000000000..48da429b70 --- /dev/null +++ b/packages/multimedia/ffmpeg/patches/0.10.7/ffmpeg-0.10.6-905.01-VFP_acceleration.patch @@ -0,0 +1,4019 @@ +From d4b2cd80ab6f54590fecfbfbf0a414cec1cf1204 Mon Sep 17 00:00:00 2001 +From: Ben Avison +Date: Fri, 14 Jun 2013 16:07:53 +0100 +Subject: [PATCH 50/55] Add VFP-accelerated version of synth_filter_float(), + used by DTS Coherent Acoustics decoder + +--- + libavcodec/arm/Makefile | 1 + + libavcodec/arm/fft_init_arm.c | 8 ++ + libavcodec/arm/synth_filter_vfp.S | 206 +++++++++++++++++++++++++++ + 3 files changed, 215 insertions(+) + create mode 100644 libavcodec/arm/synth_filter_vfp.S + +diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile +index 52709b6..b8b4df2 100644 +--- a/libavcodec/arm/Makefile ++++ b/libavcodec/arm/Makefile +@@ -37,6 +37,7 @@ OBJS-$(HAVE_ARMV6) += arm/dsputil_init_armv6.o \ + $(ARMV6-OBJS-yes) + + VFP-OBJS-$(HAVE_ARMV6) += arm/fmtconvert_vfp.o \ ++ arm/synth_filter_vfp.o + + OBJS-$(HAVE_ARMVFP) += arm/dsputil_vfp.o \ + arm/dsputil_init_vfp.o \ +diff --git a/libavcodec/arm/fft_init_arm.c b/libavcodec/arm/fft_init_arm.c +index cdb64e7..605b3dd 100644 +--- a/libavcodec/arm/fft_init_arm.c ++++ b/libavcodec/arm/fft_init_arm.c +@@ -31,6 +31,12 @@ + + void ff_rdft_calc_neon(struct RDFTContext *s, FFTSample *z); + ++void ff_synth_filter_float_vfp(FFTContext *imdct, ++ float *synth_buf_ptr, int *synth_buf_offset, ++ float synth_buf2[32], const float window[512], ++ float out[32], const float in[32], ++ float scale); ++ + void ff_synth_filter_float_neon(FFTContext *imdct, + float *synth_buf_ptr, int *synth_buf_offset, + float synth_buf2[32], const float window[512], +@@ -62,6 +68,8 @@ av_cold void ff_rdft_init_arm(RDFTContext *s) + #if CONFIG_DCA_DECODER + av_cold void ff_synth_filter_init_arm(SynthFilterContext *s) + { ++ if (HAVE_ARMVFP) ++ s->synth_filter_float = ff_synth_filter_float_vfp; + if (HAVE_NEON) + s->synth_filter_float = ff_synth_filter_float_neon; + } +diff --git a/libavcodec/arm/synth_filter_vfp.S b/libavcodec/arm/synth_filter_vfp.S +new file mode 100644 +index 0000000..8c54267 +--- /dev/null ++++ b/libavcodec/arm/synth_filter_vfp.S +@@ -0,0 +1,206 @@ ++/* ++ * Copyright (c) 2013 RISC OS Open Ltd ++ * ++ * 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 ++ * ++ * Author: Ben Avison ++ */ ++ ++#include "libavcodec/arm/asm.S" ++ ++IMDCT .req r0 ++ORIG_P_SB .req r1 ++P_SB_OFF .req r2 ++I .req r0 ++P_SB2_UP .req r1 ++OLDFPSCR .req r2 ++P_SB2_DN .req r3 ++P_WIN_DN .req r4 ++P_OUT_DN .req r5 ++P_SB .req r6 ++J_WRAP .req r7 ++P_WIN_UP .req r12 ++P_OUT_UP .req r14 ++ ++SCALE .req s0 ++SBUF_DAT_REV0 .req s4 ++SBUF_DAT_REV1 .req s5 ++SBUF_DAT_REV2 .req s6 ++SBUF_DAT_REV3 .req s7 ++VA0 .req s8 ++VA3 .req s11 ++VB0 .req s12 ++VB3 .req s15 ++VC0 .req s8 ++VC3 .req s11 ++VD0 .req s12 ++VD3 .req s15 ++SBUF_DAT0 .req s16 ++SBUF_DAT1 .req s17 ++SBUF_DAT2 .req s18 ++SBUF_DAT3 .req s19 ++SBUF_DAT_ALT0 .req s20 ++SBUF_DAT_ALT1 .req s21 ++SBUF_DAT_ALT2 .req s22 ++SBUF_DAT_ALT3 .req s23 ++WIN_DN_DAT0 .req s24 ++WIN_UP_DAT0 .req s28 ++ ++ ++.macro inner_loop half, tail, head ++ .if (OFFSET & (64*4)) == 0 @ even numbered call ++ SBUF_DAT_THIS0 .req SBUF_DAT0 ++ SBUF_DAT_THIS1 .req SBUF_DAT1 ++ SBUF_DAT_THIS2 .req SBUF_DAT2 ++ SBUF_DAT_THIS3 .req SBUF_DAT3 ++ .ifnc "\head","" ++ vldr d8, [P_SB, #OFFSET] @ d8 = SBUF_DAT ++ vldr d9, [P_SB, #OFFSET+8] ++ .endif ++ .else ++ SBUF_DAT_THIS0 .req SBUF_DAT_ALT0 ++ SBUF_DAT_THIS1 .req SBUF_DAT_ALT1 ++ SBUF_DAT_THIS2 .req SBUF_DAT_ALT2 ++ SBUF_DAT_THIS3 .req SBUF_DAT_ALT3 ++ .ifnc "\head","" ++ vldr d10, [P_SB, #OFFSET] @ d10 = SBUF_DAT_ALT ++ vldr d11, [P_SB, #OFFSET+8] ++ .endif ++ .endif ++ .ifnc "\tail","" ++ .ifc "\half","ab" ++ vmls.f VA0, SBUF_DAT_REV0, WIN_DN_DAT0 @ all operands treated as vectors ++ .else ++ vmla.f VD0, SBUF_DAT_REV0, WIN_DN_DAT0 @ all operands treated as vectors ++ .endif ++ .endif ++ .ifnc "\head","" ++ vldr d14, [P_WIN_UP, #OFFSET] @ d14 = WIN_UP_DAT ++ vldr d15, [P_WIN_UP, #OFFSET+8] ++ vldr d12, [P_WIN_DN, #OFFSET] @ d12 = WIN_DN_DAT ++ vldr d13, [P_WIN_DN, #OFFSET+8] ++ vmov SBUF_DAT_REV3, SBUF_DAT_THIS0 ++ vmov SBUF_DAT_REV2, SBUF_DAT_THIS1 ++ vmov SBUF_DAT_REV1, SBUF_DAT_THIS2 ++ vmov SBUF_DAT_REV0, SBUF_DAT_THIS3 ++ .ifc "\half","ab" ++ vmla.f VB0, SBUF_DAT_THIS0, WIN_UP_DAT0 ++ .else ++ vmla.f VC0, SBUF_DAT_THIS0, WIN_UP_DAT0 ++ .endif ++ teq J_WRAP, #J ++ bne 2f @ strongly predictable, so better than cond exec in this case ++ sub P_SB, P_SB, #512*4 ++2: ++ .set J, J - 64 ++ .set OFFSET, OFFSET + 64*4 ++ .endif ++ .unreq SBUF_DAT_THIS0 ++ .unreq SBUF_DAT_THIS1 ++ .unreq SBUF_DAT_THIS2 ++ .unreq SBUF_DAT_THIS3 ++.endm ++ ++ ++/* void ff_synth_filter_float_vfp(FFTContext *imdct, ++ * float *synth_buf_ptr, int *synth_buf_offset, ++ * float synth_buf2[32], const float window[512], ++ * float out[32], const float in[32], float scale) ++ */ ++function ff_synth_filter_float_vfp, export=1 ++ push {r3-r7,lr} ++ vpush {s16-s31} ++ ldr lr, [P_SB_OFF] ++ add a2, ORIG_P_SB, lr, LSL #2 @ calculate synth_buf to pass to imdct_half ++ mov P_SB, a2 @ and keep a copy for ourselves ++ bic J_WRAP, lr, #63 @ mangled to make testing for wrap easier in inner loop ++ sub lr, lr, #32 ++ and lr, lr, #512-32 ++ str lr, [P_SB_OFF] @ rotate offset, modulo buffer size, ready for next call ++ ldr a3, [sp, #(16+6+2)*4] @ fetch in from stack, to pass to imdct_half ++VFP vmov s16, SCALE @ imdct_half is free to corrupt s0, but it contains one of our arguments in hardfp case ++ bl ff_imdct_half_c ++VFP vmov SCALE, s16 ++ ++ vmrs OLDFPSCR, FPSCR ++ ldr lr, =0x03030000 @ RunFast mode, short vectors of length 4, stride 1 ++ vmsr FPSCR, lr ++ ldr P_SB2_DN, [sp, #16*4] ++ ldr P_WIN_DN, [sp, #(16+6+0)*4] ++ ldr P_OUT_DN, [sp, #(16+6+1)*4] ++NOVFP vldr SCALE, [sp, #(16+6+3)*4] ++ ++#define IMM_OFF_SKEW 956 /* also valid immediate constant when you add 16*4 */ ++ add P_SB, P_SB, #IMM_OFF_SKEW @ so we can use -ve offsets to use full immediate offset range ++ add P_SB2_UP, P_SB2_DN, #16*4 ++ add P_WIN_UP, P_WIN_DN, #16*4+IMM_OFF_SKEW ++ add P_OUT_UP, P_OUT_DN, #16*4 ++ add P_SB2_DN, P_SB2_DN, #16*4 ++ add P_WIN_DN, P_WIN_DN, #12*4+IMM_OFF_SKEW ++ add P_OUT_DN, P_OUT_DN, #16*4 ++ mov I, #4 ++1: ++ vldmia P_SB2_UP!, {VB0-VB3} ++ vldmdb P_SB2_DN!, {VA0-VA3} ++ .set J, 512 - 64 ++ .set OFFSET, -IMM_OFF_SKEW ++ inner_loop ab,, head ++ .rept 7 ++ inner_loop ab, tail, head ++ .endr ++ inner_loop ab, tail ++ add P_WIN_UP, P_WIN_UP, #4*4 ++ sub P_WIN_DN, P_WIN_DN, #4*4 ++ vmul.f VB0, VB0, SCALE @ SCALE treated as scalar ++ add P_SB, P_SB, #(512+4)*4 ++ subs I, I, #1 ++ vmul.f VA0, VA0, SCALE ++ vstmia P_OUT_UP!, {VB0-VB3} ++ vstmdb P_OUT_DN!, {VA0-VA3} ++ bne 1b ++ ++ add P_SB2_DN, P_SB2_DN, #(16+28-12)*4 ++ sub P_SB2_UP, P_SB2_UP, #(16+16)*4 ++ add P_WIN_DN, P_WIN_DN, #(32+16+28-12)*4 ++ mov I, #4 ++1: ++ vldr.d d4, zero @ d4 = VC0 ++ vldr.d d5, zero ++ vldr.d d6, zero @ d6 = VD0 ++ vldr.d d7, zero ++ .set J, 512 - 64 ++ .set OFFSET, -IMM_OFF_SKEW ++ inner_loop cd,, head ++ .rept 7 ++ inner_loop cd, tail, head ++ .endr ++ inner_loop cd, tail ++ add P_WIN_UP, P_WIN_UP, #4*4 ++ sub P_WIN_DN, P_WIN_DN, #4*4 ++ add P_SB, P_SB, #(512+4)*4 ++ subs I, I, #1 ++ vstmia P_SB2_UP!, {VC0-VC3} ++ vstmdb P_SB2_DN!, {VD0-VD3} ++ bne 1b ++ ++ vmsr FPSCR, OLDFPSCR ++ vpop {s16-s31} ++ pop {r3-r7,pc} ++endfunc ++ ++ .align 3 ++zero: .word 0, 0 +-- +1.8.1.6 + + +From d32ba2d8b8aacc9efb8c8a80152ae0684600e874 Mon Sep 17 00:00:00 2001 +From: Ben Avison +Date: Tue, 25 Jun 2013 17:22:50 +0100 +Subject: [PATCH 51/55] 1st version of ff_int32_to_float_fmul_scalar_vfp + +--- + libavcodec/arm/fmtconvert_init_arm.c | 10 +++++-- + libavcodec/arm/fmtconvert_vfp.S | 38 +++++++++++++++++++++++++ + 2 files changed, 46 insertions(+), 2 deletions(-) + +diff --git a/libavcodec/arm/fmtconvert_init_arm.c b/libavcodec/arm/fmtconvert_init_arm.c +index 4b6e393..6eb6cd4 100644 +--- a/libavcodec/arm/fmtconvert_init_arm.c ++++ b/libavcodec/arm/fmtconvert_init_arm.c +@@ -29,12 +29,18 @@ void ff_int32_to_float_fmul_scalar_neon(float *dst, const int *src, + void ff_float_to_int16_neon(int16_t *dst, const float *src, long len); + void ff_float_to_int16_interleave_neon(int16_t *, const float **, long, int); + ++void ff_int32_to_float_fmul_scalar_vfp(float *dst, const int *src, ++ float mul, int len); ++ + void ff_float_to_int16_vfp(int16_t *dst, const float *src, long len); + + void ff_fmt_convert_init_arm(FmtConvertContext *c, AVCodecContext *avctx) + { +- if (HAVE_ARMVFP && HAVE_ARMV6) { +- c->float_to_int16 = ff_float_to_int16_vfp; ++ if (HAVE_ARMVFP) { ++ c->int32_to_float_fmul_scalar = ff_int32_to_float_fmul_scalar_vfp; ++ if (HAVE_ARMV6) { ++ c->float_to_int16 = ff_float_to_int16_vfp; ++ } + } + + if (HAVE_NEON) { +diff --git a/libavcodec/arm/fmtconvert_vfp.S b/libavcodec/arm/fmtconvert_vfp.S +index 7e2eb83..531a8ad 100644 +--- a/libavcodec/arm/fmtconvert_vfp.S ++++ b/libavcodec/arm/fmtconvert_vfp.S +@@ -1,5 +1,6 @@ + /* + * Copyright (c) 2008 Siarhei Siamashka ++ * Copyright (c) 2013 RISC OS Open Ltd + * + * This file is part of FFmpeg. + * +@@ -76,3 +77,40 @@ function ff_float_to_int16_vfp, export=1 + vpop {d8-d11} + pop {r4-r8,pc} + endfunc ++ ++/** ++ * ARM VFP optimised int32 to float conversion. ++ * Assume len is a multiple of 8, destination buffer is at least 4 bytes aligned ++ * (16 bytes alignment is best for BCM2835), little-endian. ++ */ ++@ void ff_int32_to_float_fmul_scalar_vfp(float *dst, const int *src, float mul, int len) ++function ff_int32_to_float_fmul_scalar_vfp, export=1 ++VFP tmp .req a4 ++VFP len .req a3 ++NOVFP tmp .req a3 ++NOVFP len .req a4 ++NOVFP vmov s0, a3 ++ ldr tmp, =0x03070000 @ RunFast mode, short vectors of length 8, stride 1 ++ vmrs ip, FPSCR ++ vmsr FPSCR, tmp ++1: ++ vldmia a2!, {s8-s15} ++ vcvt.f32.s32 s8, s8 ++ vcvt.f32.s32 s9, s9 ++ vcvt.f32.s32 s10, s10 ++ vcvt.f32.s32 s11, s11 ++ vcvt.f32.s32 s12, s12 ++ vcvt.f32.s32 s13, s13 ++ vcvt.f32.s32 s14, s14 ++ vcvt.f32.s32 s15, s15 ++ vmul.f32 s8, s8, s0 ++ subs len, len, #8 ++ vstmia a1!, {s8-s11} ++ vstmia a1!, {s12-s15} ++ bne 1b ++ ++ vmsr FPSCR, ip ++ bx lr ++endfunc ++ .unreq tmp ++ .unreq len +-- +1.8.1.6 + + +From e8ba866ef0bb68e43da157683d398ab7cce817a6 Mon Sep 17 00:00:00 2001 +From: Ben Avison +Date: Wed, 26 Jun 2013 00:49:15 +0100 +Subject: [PATCH 52/55] 2nd version of fmul_scalar + +--- + libavcodec/arm/fmtconvert_init_arm.c | 5 + + libavcodec/arm/fmtconvert_vfp.S | 162 ++ + libavcodec/dcadec.c | 2533 +++++++++++++++++++++++ + libavcodec/fmtconvert.c | 7 + + libavcodec/fmtconvert.h | 14 + + 5 files changed, 2721 insertions(+) + create mode 100644 libavcodec/dcadec.c + +diff --git a/libavcodec/arm/fmtconvert_init_arm.c b/libavcodec/arm/fmtconvert_init_arm.c +index 6eb6cd4..91c652e 100644 +--- a/libavcodec/arm/fmtconvert_init_arm.c ++++ b/libavcodec/arm/fmtconvert_init_arm.c +@@ -31,6 +31,8 @@ void ff_int32_to_float_fmul_scalar_neon(float *dst, const int *src, + + void ff_int32_to_float_fmul_scalar_vfp(float *dst, const int *src, + float mul, int len); ++void ff_int32_to_float_fmul_scalar_array_vfp(FmtConvertContext *c, float *dst, const int *src, ++ float *mul, int len); + + void ff_float_to_int16_vfp(int16_t *dst, const float *src, long len); + +@@ -38,6 +40,9 @@ void ff_fmt_convert_init_arm(FmtConvertContext *c, AVCodecContext *avctx) + { + if (HAVE_ARMVFP) { + c->int32_to_float_fmul_scalar = ff_int32_to_float_fmul_scalar_vfp; ++ if (!HAVE_NEON) { ++ c->int32_to_float_fmul_scalar_array = ff_int32_to_float_fmul_scalar_array_vfp; ++ } + if (HAVE_ARMV6) { + c->float_to_int16 = ff_float_to_int16_vfp; + } +diff --git a/libavcodec/arm/fmtconvert_vfp.S b/libavcodec/arm/fmtconvert_vfp.S +index 531a8ad..13f0bc0 100644 +--- a/libavcodec/arm/fmtconvert_vfp.S ++++ b/libavcodec/arm/fmtconvert_vfp.S +@@ -83,6 +83,168 @@ endfunc + * Assume len is a multiple of 8, destination buffer is at least 4 bytes aligned + * (16 bytes alignment is best for BCM2835), little-endian. + */ ++@ void ff_int32_to_float_fmul_scalar_array_vfp(if (FmtConvertContext *c, float *dst, const int *src, float *mul, int len) ++function ff_int32_to_float_fmul_scalar_array_vfp, export=1 ++ push {lr} ++ ldr a1, [sp, #4] ++ subs lr, a1, #3*8 ++ bcc 50f @ too short to pipeline ++ @ Now need to find (len / 8) % 3. The approximation ++ @ x / 24 = (x * 0xAB) >> 12 ++ @ is good for x < 4096, which is true for both AC3 and DCA. ++ mov a1, #0xAB ++ ldr ip, =0x03070000 @ RunFast mode, short vectors of length 8, stride 1 ++ mul a1, lr, a1 ++ vpush {s16-s31} ++ mov a1, a1, lsr #12 ++ add a1, a1, a1, lsl #1 ++ rsb a1, a1, lr, lsr #3 ++ cmp a1, #1 ++ vmrs a1, FPSCR ++ vmsr FPSCR, ip ++ beq 11f ++ blo 10f ++ @ Array is (2 + multiple of 3) x 8 floats long ++ @ drop through... ++ vldmia a3!, {s16-s23} ++ vldmia a4!, {s2,s3} ++ vldmia a3!, {s24-s31} ++ vcvt.f32.s32 s16, s16 ++ vcvt.f32.s32 s17, s17 ++ vcvt.f32.s32 s18, s18 ++ vcvt.f32.s32 s19, s19 ++ vcvt.f32.s32 s20, s20 ++ vcvt.f32.s32 s21, s21 ++ vcvt.f32.s32 s22, s22 ++ vcvt.f32.s32 s23, s23 ++ vmul.f32 s16, s16, s2 ++ @ drop through... ++3: ++ vldmia a3!, {s8-s15} ++ vldmia a4!, {s1} ++ vcvt.f32.s32 s24, s24 ++ vcvt.f32.s32 s25, s25 ++ vcvt.f32.s32 s26, s26 ++ vcvt.f32.s32 s27, s27 ++ vcvt.f32.s32 s28, s28 ++ vcvt.f32.s32 s29, s29 ++ vcvt.f32.s32 s30, s30 ++ vcvt.f32.s32 s31, s31 ++ vmul.f32 s24, s24, s3 ++ vstmia a2!, {s16-s19} ++ vstmia a2!, {s20-s23} ++2: ++ vldmia a3!, {s16-s23} ++ vldmia a4!, {s2} ++ vcvt.f32.s32 s8, s8 ++ vcvt.f32.s32 s9, s9 ++ vcvt.f32.s32 s10, s10 ++ vcvt.f32.s32 s11, s11 ++ vcvt.f32.s32 s12, s12 ++ vcvt.f32.s32 s13, s13 ++ vcvt.f32.s32 s14, s14 ++ vcvt.f32.s32 s15, s15 ++ vmul.f32 s8, s8, s1 ++ vstmia a2!, {s24-s27} ++ vstmia a2!, {s28-s31} ++1: ++ vldmia a3!, {s24-s31} ++ vldmia a4!, {s3} ++ vcvt.f32.s32 s16, s16 ++ vcvt.f32.s32 s17, s17 ++ vcvt.f32.s32 s18, s18 ++ vcvt.f32.s32 s19, s19 ++ vcvt.f32.s32 s20, s20 ++ vcvt.f32.s32 s21, s21 ++ vcvt.f32.s32 s22, s22 ++ vcvt.f32.s32 s23, s23 ++ vmul.f32 s16, s16, s2 ++ vstmia a2!, {s8-s11} ++ vstmia a2!, {s12-s15} ++ ++ subs lr, lr, #8*3 ++ bpl 3b ++ ++ vcvt.f32.s32 s24, s24 ++ vcvt.f32.s32 s25, s25 ++ vcvt.f32.s32 s26, s26 ++ vcvt.f32.s32 s27, s27 ++ vcvt.f32.s32 s28, s28 ++ vcvt.f32.s32 s29, s29 ++ vcvt.f32.s32 s30, s30 ++ vcvt.f32.s32 s31, s31 ++ vmul.f32 s24, s24, s3 ++ vstmia a2!, {s16-s19} ++ vstmia a2!, {s20-s23} ++ vstmia a2!, {s24-s27} ++ vstmia a2!, {s28-s31} ++ ++ vmsr FPSCR, a1 ++ vpop {s16-s31} ++ pop {pc} ++ ++10: @ Array is (multiple of 3) x 8 floats long ++ vldmia a3!, {s8-s15} ++ vldmia a4!, {s1,s2} ++ vldmia a3!, {s16-s23} ++ vcvt.f32.s32 s8, s8 ++ vcvt.f32.s32 s9, s9 ++ vcvt.f32.s32 s10, s10 ++ vcvt.f32.s32 s11, s11 ++ vcvt.f32.s32 s12, s12 ++ vcvt.f32.s32 s13, s13 ++ vcvt.f32.s32 s14, s14 ++ vcvt.f32.s32 s15, s15 ++ vmul.f32 s8, s8, s1 ++ b 1b ++ ++11: @ Array is (1 + multiple of 3) x 8 floats long ++ vldmia a3!, {s24-s31} ++ vldmia a4!, {s3} ++ vldmia a3!, {s8-s15} ++ vldmia a4!, {s1} ++ vcvt.f32.s32 s24, s24 ++ vcvt.f32.s32 s25, s25 ++ vcvt.f32.s32 s26, s26 ++ vcvt.f32.s32 s27, s27 ++ vcvt.f32.s32 s28, s28 ++ vcvt.f32.s32 s29, s29 ++ vcvt.f32.s32 s30, s30 ++ vcvt.f32.s32 s31, s31 ++ vmul.f32 s24, s24, s3 ++ b 2b ++ ++50: ++ ldr lr, =0x03070000 @ RunFast mode, short vectors of length 8, stride 1 ++ vmrs ip, FPSCR ++ vmsr FPSCR, lr ++51: ++ vldmia a3!, {s8-s15} ++ vldmia a4!, {s0} ++ vcvt.f32.s32 s8, s8 ++ vcvt.f32.s32 s9, s9 ++ vcvt.f32.s32 s10, s10 ++ vcvt.f32.s32 s11, s11 ++ vcvt.f32.s32 s12, s12 ++ vcvt.f32.s32 s13, s13 ++ vcvt.f32.s32 s14, s14 ++ vcvt.f32.s32 s15, s15 ++ vmul.f32 s8, s8, s0 ++ subs a1, a1, #8 ++ vstmia a2!, {s8-s11} ++ vstmia a2!, {s12-s15} ++ bne 51b ++ ++ vmsr FPSCR, ip ++ pop {pc} ++endfunc ++ ++/** ++ * ARM VFP optimised int32 to float conversion. ++ * Assume len is a multiple of 8, destination buffer is at least 4 bytes aligned ++ * (16 bytes alignment is best for BCM2835), little-endian. ++ * TODO: could be further optimised by unrolling and interleaving, as above ++ */ + @ void ff_int32_to_float_fmul_scalar_vfp(float *dst, const int *src, float mul, int len) + function ff_int32_to_float_fmul_scalar_vfp, export=1 + VFP tmp .req a4 +diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c +new file mode 100644 +index 0000000..fe568ee +--- /dev/null ++++ b/libavcodec/dcadec.c +@@ -0,0 +1,2533 @@ ++/* ++ * DCA compatible decoder ++ * Copyright (C) 2004 Gildas Bazin ++ * Copyright (C) 2004 Benjamin Zores ++ * Copyright (C) 2006 Benjamin Larsson ++ * Copyright (C) 2007 Konstantin Shishkov ++ * ++ * 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 ++#include ++#include ++ ++#include "libavutil/channel_layout.h" ++#include "libavutil/common.h" ++#include "libavutil/float_dsp.h" ++#include "libavutil/internal.h" ++#include "libavutil/intreadwrite.h" ++#include "libavutil/mathematics.h" ++#include "libavutil/samplefmt.h" ++#include "avcodec.h" ++#include "fft.h" ++#include "get_bits.h" ++#include "put_bits.h" ++#include "dcadata.h" ++#include "dcahuff.h" ++#include "dca.h" ++#include "dca_parser.h" ++#include "mathops.h" ++#include "synth_filter.h" ++#include "dcadsp.h" ++#include "fmtconvert.h" ++#include "internal.h" ++ ++#if ARCH_ARM ++# include "arm/dca.h" ++#endif ++ ++//#define TRACE ++ ++#define DCA_PRIM_CHANNELS_MAX (7) ++#define DCA_SUBBANDS (64) ++#define DCA_ABITS_MAX (32) /* Should be 28 */ ++#define DCA_SUBSUBFRAMES_MAX (4) ++#define DCA_SUBFRAMES_MAX (16) ++#define DCA_BLOCKS_MAX (16) ++#define DCA_LFE_MAX (3) ++#define DCA_CHSETS_MAX (4) ++#define DCA_CHSET_CHANS_MAX (8) ++ ++enum DCAMode { ++ DCA_MONO = 0, ++ DCA_CHANNEL, ++ DCA_STEREO, ++ DCA_STEREO_SUMDIFF, ++ DCA_STEREO_TOTAL, ++ DCA_3F, ++ DCA_2F1R, ++ DCA_3F1R, ++ DCA_2F2R, ++ DCA_3F2R, ++ DCA_4F2R ++}; ++ ++/* these are unconfirmed but should be mostly correct */ ++enum DCAExSSSpeakerMask { ++ DCA_EXSS_FRONT_CENTER = 0x0001, ++ DCA_EXSS_FRONT_LEFT_RIGHT = 0x0002, ++ DCA_EXSS_SIDE_REAR_LEFT_RIGHT = 0x0004, ++ DCA_EXSS_LFE = 0x0008, ++ DCA_EXSS_REAR_CENTER = 0x0010, ++ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT = 0x0020, ++ DCA_EXSS_REAR_LEFT_RIGHT = 0x0040, ++ DCA_EXSS_FRONT_HIGH_CENTER = 0x0080, ++ DCA_EXSS_OVERHEAD = 0x0100, ++ DCA_EXSS_CENTER_LEFT_RIGHT = 0x0200, ++ DCA_EXSS_WIDE_LEFT_RIGHT = 0x0400, ++ DCA_EXSS_SIDE_LEFT_RIGHT = 0x0800, ++ DCA_EXSS_LFE2 = 0x1000, ++ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT = 0x2000, ++ DCA_EXSS_REAR_HIGH_CENTER = 0x4000, ++ DCA_EXSS_REAR_HIGH_LEFT_RIGHT = 0x8000, ++}; ++ ++enum DCAXxchSpeakerMask { ++ DCA_XXCH_FRONT_CENTER = 0x0000001, ++ DCA_XXCH_FRONT_LEFT = 0x0000002, ++ DCA_XXCH_FRONT_RIGHT = 0x0000004, ++ DCA_XXCH_SIDE_REAR_LEFT = 0x0000008, ++ DCA_XXCH_SIDE_REAR_RIGHT = 0x0000010, ++ DCA_XXCH_LFE1 = 0x0000020, ++ DCA_XXCH_REAR_CENTER = 0x0000040, ++ DCA_XXCH_SURROUND_REAR_LEFT = 0x0000080, ++ DCA_XXCH_SURROUND_REAR_RIGHT = 0x0000100, ++ DCA_XXCH_SIDE_SURROUND_LEFT = 0x0000200, ++ DCA_XXCH_SIDE_SURROUND_RIGHT = 0x0000400, ++ DCA_XXCH_FRONT_CENTER_LEFT = 0x0000800, ++ DCA_XXCH_FRONT_CENTER_RIGHT = 0x0001000, ++ DCA_XXCH_FRONT_HIGH_LEFT = 0x0002000, ++ DCA_XXCH_FRONT_HIGH_CENTER = 0x0004000, ++ DCA_XXCH_FRONT_HIGH_RIGHT = 0x0008000, ++ DCA_XXCH_LFE2 = 0x0010000, ++ DCA_XXCH_SIDE_FRONT_LEFT = 0x0020000, ++ DCA_XXCH_SIDE_FRONT_RIGHT = 0x0040000, ++ DCA_XXCH_OVERHEAD = 0x0080000, ++ DCA_XXCH_SIDE_HIGH_LEFT = 0x0100000, ++ DCA_XXCH_SIDE_HIGH_RIGHT = 0x0200000, ++ DCA_XXCH_REAR_HIGH_CENTER = 0x0400000, ++ DCA_XXCH_REAR_HIGH_LEFT = 0x0800000, ++ DCA_XXCH_REAR_HIGH_RIGHT = 0x1000000, ++ DCA_XXCH_REAR_LOW_CENTER = 0x2000000, ++ DCA_XXCH_REAR_LOW_LEFT = 0x4000000, ++ DCA_XXCH_REAR_LOW_RIGHT = 0x8000000, ++}; ++ ++static const uint32_t map_xxch_to_native[28] = { ++ AV_CH_FRONT_CENTER, ++ AV_CH_FRONT_LEFT, ++ AV_CH_FRONT_RIGHT, ++ AV_CH_SIDE_LEFT, ++ AV_CH_SIDE_RIGHT, ++ AV_CH_LOW_FREQUENCY, ++ AV_CH_BACK_CENTER, ++ AV_CH_BACK_LEFT, ++ AV_CH_BACK_RIGHT, ++ AV_CH_SIDE_LEFT, /* side surround left -- dup sur side L */ ++ AV_CH_SIDE_RIGHT, /* side surround right -- dup sur side R */ ++ AV_CH_FRONT_LEFT_OF_CENTER, ++ AV_CH_FRONT_RIGHT_OF_CENTER, ++ AV_CH_TOP_FRONT_LEFT, ++ AV_CH_TOP_FRONT_CENTER, ++ AV_CH_TOP_FRONT_RIGHT, ++ AV_CH_LOW_FREQUENCY, /* lfe2 -- duplicate lfe1 position */ ++ AV_CH_FRONT_LEFT_OF_CENTER, /* side front left -- dup front cntr L */ ++ AV_CH_FRONT_RIGHT_OF_CENTER,/* side front right -- dup front cntr R */ ++ AV_CH_TOP_CENTER, /* overhead */ ++ AV_CH_TOP_FRONT_LEFT, /* side high left -- dup */ ++ AV_CH_TOP_FRONT_RIGHT, /* side high right -- dup */ ++ AV_CH_TOP_BACK_CENTER, ++ AV_CH_TOP_BACK_LEFT, ++ AV_CH_TOP_BACK_RIGHT, ++ AV_CH_BACK_CENTER, /* rear low center -- dup */ ++ AV_CH_BACK_LEFT, /* rear low left -- dup */ ++ AV_CH_BACK_RIGHT /* read low right -- dup */ ++}; ++ ++enum DCAExtensionMask { ++ DCA_EXT_CORE = 0x001, ///< core in core substream ++ DCA_EXT_XXCH = 0x002, ///< XXCh channels extension in core substream ++ DCA_EXT_X96 = 0x004, ///< 96/24 extension in core substream ++ DCA_EXT_XCH = 0x008, ///< XCh channel extension in core substream ++ DCA_EXT_EXSS_CORE = 0x010, ///< core in ExSS (extension substream) ++ DCA_EXT_EXSS_XBR = 0x020, ///< extended bitrate extension in ExSS ++ DCA_EXT_EXSS_XXCH = 0x040, ///< XXCh channels extension in ExSS ++ DCA_EXT_EXSS_X96 = 0x080, ///< 96/24 extension in ExSS ++ DCA_EXT_EXSS_LBR = 0x100, ///< low bitrate component in ExSS ++ DCA_EXT_EXSS_XLL = 0x200, ///< lossless extension in ExSS ++}; ++ ++/* -1 are reserved or unknown */ ++static const int dca_ext_audio_descr_mask[] = { ++ DCA_EXT_XCH, ++ -1, ++ DCA_EXT_X96, ++ DCA_EXT_XCH | DCA_EXT_X96, ++ -1, ++ -1, ++ DCA_EXT_XXCH, ++ -1, ++}; ++ ++/* extensions that reside in core substream */ ++#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96) ++ ++/* Tables for mapping dts channel configurations to libavcodec multichannel api. ++ * Some compromises have been made for special configurations. Most configurations ++ * are never used so complete accuracy is not needed. ++ * ++ * L = left, R = right, C = center, S = surround, F = front, R = rear, T = total, OV = overhead. ++ * S -> side, when both rear and back are configured move one of them to the side channel ++ * OV -> center back ++ * All 2 channel configurations -> AV_CH_LAYOUT_STEREO ++ */ ++static const uint64_t dca_core_channel_layout[] = { ++ AV_CH_FRONT_CENTER, ///< 1, A ++ AV_CH_LAYOUT_STEREO, ///< 2, A + B (dual mono) ++ AV_CH_LAYOUT_STEREO, ///< 2, L + R (stereo) ++ AV_CH_LAYOUT_STEREO, ///< 2, (L + R) + (L - R) (sum-difference) ++ AV_CH_LAYOUT_STEREO, ///< 2, LT + RT (left and right total) ++ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER, ///< 3, C + L + R ++ AV_CH_LAYOUT_STEREO | AV_CH_BACK_CENTER, ///< 3, L + R + S ++ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 4, C + L + R + S ++ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 4, L + R + SL + SR ++ ++ AV_CH_LAYOUT_STEREO | AV_CH_FRONT_CENTER | AV_CH_SIDE_LEFT | ++ AV_CH_SIDE_RIGHT, ///< 5, C + L + R + SL + SR ++ ++ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | ++ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER, ///< 6, CL + CR + L + R + SL + SR ++ ++ AV_CH_LAYOUT_STEREO | AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | ++ AV_CH_FRONT_CENTER | AV_CH_BACK_CENTER, ///< 6, C + L + R + LR + RR + OV ++ ++ AV_CH_FRONT_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | ++ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_BACK_CENTER | ++ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 6, CF + CR + LF + RF + LR + RR ++ ++ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | ++ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO | ++ AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT, ///< 7, CL + C + CR + L + R + SL + SR ++ ++ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER | ++ AV_CH_LAYOUT_STEREO | AV_CH_SIDE_LEFT | AV_CH_SIDE_RIGHT | ++ AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT, ///< 8, CL + CR + L + R + SL1 + SL2 + SR1 + SR2 ++ ++ AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_CENTER | ++ AV_CH_FRONT_RIGHT_OF_CENTER | AV_CH_LAYOUT_STEREO | ++ AV_CH_SIDE_LEFT | AV_CH_BACK_CENTER | AV_CH_SIDE_RIGHT, ///< 8, CL + C + CR + L + R + SL + S + SR ++}; ++ ++static const int8_t dca_lfe_index[] = { ++ 1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3 ++}; ++ ++static const int8_t dca_channel_reorder_lfe[][9] = { ++ { 0, -1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, -1, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 4, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, 4, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 4, 5, -1, -1, -1, -1}, ++ { 3, 4, 0, 1, 5, 6, -1, -1, -1}, ++ { 2, 0, 1, 4, 5, 6, -1, -1, -1}, ++ { 0, 6, 4, 5, 2, 3, -1, -1, -1}, ++ { 4, 2, 5, 0, 1, 6, 7, -1, -1}, ++ { 5, 6, 0, 1, 7, 3, 8, 4, -1}, ++ { 4, 2, 5, 0, 1, 6, 8, 7, -1}, ++}; ++ ++static const int8_t dca_channel_reorder_lfe_xch[][9] = { ++ { 0, 2, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, -1, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 4, -1, -1, -1, -1, -1}, ++ { 0, 1, 3, 4, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 4, 5, -1, -1, -1, -1}, ++ { 0, 1, 4, 5, 3, -1, -1, -1, -1}, ++ { 2, 0, 1, 5, 6, 4, -1, -1, -1}, ++ { 3, 4, 0, 1, 6, 7, 5, -1, -1}, ++ { 2, 0, 1, 4, 5, 6, 7, -1, -1}, ++ { 0, 6, 4, 5, 2, 3, 7, -1, -1}, ++ { 4, 2, 5, 0, 1, 7, 8, 6, -1}, ++ { 5, 6, 0, 1, 8, 3, 9, 4, 7}, ++ { 4, 2, 5, 0, 1, 6, 9, 8, 7}, ++}; ++ ++static const int8_t dca_channel_reorder_nolfe[][9] = { ++ { 0, -1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, -1, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 3, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, 3, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 3, 4, -1, -1, -1, -1}, ++ { 2, 3, 0, 1, 4, 5, -1, -1, -1}, ++ { 2, 0, 1, 3, 4, 5, -1, -1, -1}, ++ { 0, 5, 3, 4, 1, 2, -1, -1, -1}, ++ { 3, 2, 4, 0, 1, 5, 6, -1, -1}, ++ { 4, 5, 0, 1, 6, 2, 7, 3, -1}, ++ { 3, 2, 4, 0, 1, 5, 7, 6, -1}, ++}; ++ ++static const int8_t dca_channel_reorder_nolfe_xch[][9] = { ++ { 0, 1, -1, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, -1, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, -1, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 3, -1, -1, -1, -1, -1}, ++ { 0, 1, 2, 3, -1, -1, -1, -1, -1}, ++ { 2, 0, 1, 3, 4, -1, -1, -1, -1}, ++ { 0, 1, 3, 4, 2, -1, -1, -1, -1}, ++ { 2, 0, 1, 4, 5, 3, -1, -1, -1}, ++ { 2, 3, 0, 1, 5, 6, 4, -1, -1}, ++ { 2, 0, 1, 3, 4, 5, 6, -1, -1}, ++ { 0, 5, 3, 4, 1, 2, 6, -1, -1}, ++ { 3, 2, 4, 0, 1, 6, 7, 5, -1}, ++ { 4, 5, 0, 1, 7, 2, 8, 3, 6}, ++ { 3, 2, 4, 0, 1, 5, 8, 7, 6}, ++}; ++ ++#define DCA_DOLBY 101 /* FIXME */ ++ ++#define DCA_CHANNEL_BITS 6 ++#define DCA_CHANNEL_MASK 0x3F ++ ++#define DCA_LFE 0x80 ++ ++#define HEADER_SIZE 14 ++ ++#define DCA_MAX_FRAME_SIZE 16384 ++#define DCA_MAX_EXSS_HEADER_SIZE 4096 ++ ++#define DCA_BUFFER_PADDING_SIZE 1024 ++ ++/** Bit allocation */ ++typedef struct { ++ int offset; ///< code values offset ++ int maxbits[8]; ///< max bits in VLC ++ int wrap; ///< wrap for get_vlc2() ++ VLC vlc[8]; ///< actual codes ++} BitAlloc; ++ ++static BitAlloc dca_bitalloc_index; ///< indexes for samples VLC select ++static BitAlloc dca_tmode; ///< transition mode VLCs ++static BitAlloc dca_scalefactor; ///< scalefactor VLCs ++static BitAlloc dca_smpl_bitalloc[11]; ///< samples VLCs ++ ++static av_always_inline int get_bitalloc(GetBitContext *gb, BitAlloc *ba, ++ int idx) ++{ ++ return get_vlc2(gb, ba->vlc[idx].table, ba->vlc[idx].bits, ba->wrap) + ++ ba->offset; ++} ++ ++typedef struct { ++ AVCodecContext *avctx; ++ /* Frame header */ ++ int frame_type; ///< type of the current frame ++ int samples_deficit; ///< deficit sample count ++ int crc_present; ///< crc is present in the bitstream ++ int sample_blocks; ///< number of PCM sample blocks ++ int frame_size; ///< primary frame byte size ++ int amode; ///< audio channels arrangement ++ int sample_rate; ///< audio sampling rate ++ int bit_rate; ///< transmission bit rate ++ int bit_rate_index; ///< transmission bit rate index ++ ++ int downmix; ///< embedded downmix enabled ++ int dynrange; ///< embedded dynamic range flag ++ int timestamp; ///< embedded time stamp flag ++ int aux_data; ///< auxiliary data flag ++ int hdcd; ///< source material is mastered in HDCD ++ int ext_descr; ///< extension audio descriptor flag ++ int ext_coding; ///< extended coding flag ++ int aspf; ///< audio sync word insertion flag ++ int lfe; ///< low frequency effects flag ++ int predictor_history; ///< predictor history flag ++ int header_crc; ///< header crc check bytes ++ int multirate_inter; ///< multirate interpolator switch ++ int version; ///< encoder software revision ++ int copy_history; ///< copy history ++ int source_pcm_res; ///< source pcm resolution ++ int front_sum; ///< front sum/difference flag ++ int surround_sum; ///< surround sum/difference flag ++ int dialog_norm; ///< dialog normalisation parameter ++ ++ /* Primary audio coding header */ ++ int subframes; ///< number of subframes ++ int total_channels; ///< number of channels including extensions ++ int prim_channels; ///< number of primary audio channels ++ int subband_activity[DCA_PRIM_CHANNELS_MAX]; ///< subband activity count ++ int vq_start_subband[DCA_PRIM_CHANNELS_MAX]; ///< high frequency vq start subband ++ int joint_intensity[DCA_PRIM_CHANNELS_MAX]; ///< joint intensity coding index ++ int transient_huffman[DCA_PRIM_CHANNELS_MAX]; ///< transient mode code book ++ int scalefactor_huffman[DCA_PRIM_CHANNELS_MAX]; ///< scale factor code book ++ int bitalloc_huffman[DCA_PRIM_CHANNELS_MAX]; ///< bit allocation quantizer select ++ int quant_index_huffman[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< quantization index codebook select ++ float scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment ++ ++ /* Primary audio coding side information */ ++ int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes ++ int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count ++ int prediction_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not) ++ int prediction_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< prediction VQ coefs ++ int bitalloc[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< bit allocation index ++ int transition_mode[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< transition mode (transients) ++ int scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][2]; ///< scale factors (2 if transient) ++ int joint_huff[DCA_PRIM_CHANNELS_MAX]; ///< joint subband scale factors codebook ++ int joint_scale_factor[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< joint subband scale factors ++ int downmix_coef[DCA_PRIM_CHANNELS_MAX][2]; ///< stereo downmix coefficients ++ int dynrange_coef; ///< dynamic range coefficient ++ ++ int high_freq_vq[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS]; ///< VQ encoded high frequency subbands ++ ++ float lfe_data[2 * DCA_LFE_MAX * (DCA_BLOCKS_MAX + 4)]; ///< Low frequency effect data ++ int lfe_scale_factor; ++ ++ /* Subband samples history (for ADPCM) */ ++ DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4]; ++ DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512]; ++ DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32]; ++ int hist_index[DCA_PRIM_CHANNELS_MAX]; ++ DECLARE_ALIGNED(32, float, raXin)[32]; ++ ++ int output; ///< type of output ++ ++ DECLARE_ALIGNED(32, float, subband_samples)[DCA_BLOCKS_MAX][DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][8]; ++ float *samples_chanptr[DCA_PRIM_CHANNELS_MAX + 1]; ++ float *extra_channels[DCA_PRIM_CHANNELS_MAX + 1]; ++ uint8_t *extra_channels_buffer; ++ unsigned int extra_channels_buffer_size; ++ ++ uint8_t dca_buffer[DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE + DCA_BUFFER_PADDING_SIZE]; ++ int dca_buffer_size; ///< how much data is in the dca_buffer ++ ++ const int8_t *channel_order_tab; ///< channel reordering table, lfe and non lfe ++ GetBitContext gb; ++ /* Current position in DCA frame */ ++ int current_subframe; ++ int current_subsubframe; ++ ++ int core_ext_mask; ///< present extensions in the core substream ++ ++ /* XCh extension information */ ++ int xch_present; ///< XCh extension present and valid ++ int xch_base_channel; ///< index of first (only) channel containing XCH data ++ ++ /* XXCH extension information */ ++ int xxch_chset; ++ int xxch_nbits_spk_mask; ++ uint32_t xxch_core_spkmask; ++ uint32_t xxch_spk_masks[4]; /* speaker masks, last element is core mask */ ++ int xxch_chset_nch[4]; ++ float xxch_dmix_sf[DCA_CHSETS_MAX]; ++ ++ uint32_t xxch_dmix_embedded; /* lower layer has mix pre-embedded, per chset */ ++ float xxch_dmix_coeff[DCA_PRIM_CHANNELS_MAX][32]; /* worst case sizing */ ++ ++ int8_t xxch_order_tab[32]; ++ int8_t lfe_index; ++ ++ /* ExSS header parser */ ++ int static_fields; ///< static fields present ++ int mix_metadata; ///< mixing metadata present ++ int num_mix_configs; ///< number of mix out configurations ++ int mix_config_num_ch[4]; ///< number of channels in each mix out configuration ++ ++ int profile; ++ ++ int debug_flag; ///< used for suppressing repeated error messages output ++ AVFloatDSPContext fdsp; ++ FFTContext imdct; ++ SynthFilterContext synth; ++ DCADSPContext dcadsp; ++ FmtConvertContext fmt_conv; ++} DCAContext; ++ ++static const uint16_t dca_vlc_offs[] = { ++ 0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364, ++ 5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508, ++ 5572, 5604, 5668, 5796, 5860, 5892, 6412, 6668, 6796, 7308, 7564, ++ 7820, 8076, 8620, 9132, 9388, 9910, 10166, 10680, 11196, 11726, 12240, ++ 12752, 13298, 13810, 14326, 14840, 15500, 16022, 16540, 17158, 17678, 18264, ++ 18796, 19352, 19926, 20468, 21472, 22398, 23014, 23622, ++}; ++ ++static av_cold void dca_init_vlcs(void) ++{ ++ static int vlcs_initialized = 0; ++ int i, j, c = 14; ++ static VLC_TYPE dca_table[23622][2]; ++ ++ if (vlcs_initialized) ++ return; ++ ++ dca_bitalloc_index.offset = 1; ++ dca_bitalloc_index.wrap = 2; ++ for (i = 0; i < 5; i++) { ++ dca_bitalloc_index.vlc[i].table = &dca_table[dca_vlc_offs[i]]; ++ dca_bitalloc_index.vlc[i].table_allocated = dca_vlc_offs[i + 1] - dca_vlc_offs[i]; ++ init_vlc(&dca_bitalloc_index.vlc[i], bitalloc_12_vlc_bits[i], 12, ++ bitalloc_12_bits[i], 1, 1, ++ bitalloc_12_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); ++ } ++ dca_scalefactor.offset = -64; ++ dca_scalefactor.wrap = 2; ++ for (i = 0; i < 5; i++) { ++ dca_scalefactor.vlc[i].table = &dca_table[dca_vlc_offs[i + 5]]; ++ dca_scalefactor.vlc[i].table_allocated = dca_vlc_offs[i + 6] - dca_vlc_offs[i + 5]; ++ init_vlc(&dca_scalefactor.vlc[i], SCALES_VLC_BITS, 129, ++ scales_bits[i], 1, 1, ++ scales_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); ++ } ++ dca_tmode.offset = 0; ++ dca_tmode.wrap = 1; ++ for (i = 0; i < 4; i++) { ++ dca_tmode.vlc[i].table = &dca_table[dca_vlc_offs[i + 10]]; ++ dca_tmode.vlc[i].table_allocated = dca_vlc_offs[i + 11] - dca_vlc_offs[i + 10]; ++ init_vlc(&dca_tmode.vlc[i], tmode_vlc_bits[i], 4, ++ tmode_bits[i], 1, 1, ++ tmode_codes[i], 2, 2, INIT_VLC_USE_NEW_STATIC); ++ } ++ ++ for (i = 0; i < 10; i++) ++ for (j = 0; j < 7; j++) { ++ if (!bitalloc_codes[i][j]) ++ break; ++ dca_smpl_bitalloc[i + 1].offset = bitalloc_offsets[i]; ++ dca_smpl_bitalloc[i + 1].wrap = 1 + (j > 4); ++ dca_smpl_bitalloc[i + 1].vlc[j].table = &dca_table[dca_vlc_offs[c]]; ++ dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = dca_vlc_offs[c + 1] - dca_vlc_offs[c]; ++ ++ init_vlc(&dca_smpl_bitalloc[i + 1].vlc[j], bitalloc_maxbits[i][j], ++ bitalloc_sizes[i], ++ bitalloc_bits[i][j], 1, 1, ++ bitalloc_codes[i][j], 2, 2, INIT_VLC_USE_NEW_STATIC); ++ c++; ++ } ++ vlcs_initialized = 1; ++} ++ ++static inline void get_array(GetBitContext *gb, int *dst, int len, int bits) ++{ ++ while (len--) ++ *dst++ = get_bits(gb, bits); ++} ++ ++static inline int dca_xxch2index(DCAContext *s, int xxch_ch) ++{ ++ int i, base, mask; ++ ++ /* locate channel set containing the channel */ ++ for (i = -1, base = 0, mask = (s->xxch_core_spkmask & ~DCA_XXCH_LFE1); ++ i <= s->xxch_chset && !(mask & xxch_ch); mask = s->xxch_spk_masks[++i]) ++ base += av_popcount(mask); ++ ++ return base + av_popcount(mask & (xxch_ch - 1)); ++} ++ ++static int dca_parse_audio_coding_header(DCAContext *s, int base_channel, ++ int xxch) ++{ ++ int i, j; ++ static const float adj_table[4] = { 1.0, 1.1250, 1.2500, 1.4375 }; ++ static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 }; ++ static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 }; ++ int hdr_pos = 0, hdr_size = 0; ++ float sign, mag, scale_factor; ++ int this_chans, acc_mask; ++ int embedded_downmix; ++ int nchans, mask[8]; ++ int coeff, ichan; ++ ++ /* xxch has arbitrary sized audio coding headers */ ++ if (xxch) { ++ hdr_pos = get_bits_count(&s->gb); ++ hdr_size = get_bits(&s->gb, 7) + 1; ++ } ++ ++ nchans = get_bits(&s->gb, 3) + 1; ++ s->total_channels = nchans + base_channel; ++ s->prim_channels = s->total_channels; ++ ++ /* obtain speaker layout mask & downmix coefficients for XXCH */ ++ if (xxch) { ++ acc_mask = s->xxch_core_spkmask; ++ ++ this_chans = get_bits(&s->gb, s->xxch_nbits_spk_mask - 6) << 6; ++ s->xxch_spk_masks[s->xxch_chset] = this_chans; ++ s->xxch_chset_nch[s->xxch_chset] = nchans; ++ ++ for (i = 0; i <= s->xxch_chset; i++) ++ acc_mask |= s->xxch_spk_masks[i]; ++ ++ /* check for downmixing information */ ++ if (get_bits1(&s->gb)) { ++ embedded_downmix = get_bits1(&s->gb); ++ scale_factor = ++ 1.0f / dca_downmix_scale_factors[(get_bits(&s->gb, 6) - 1) << 2]; ++ ++ s->xxch_dmix_sf[s->xxch_chset] = scale_factor; ++ ++ for (i = base_channel; i < s->prim_channels; i++) { ++ mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask); ++ } ++ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ memset(s->xxch_dmix_coeff[j], 0, sizeof(s->xxch_dmix_coeff[0])); ++ s->xxch_dmix_embedded |= (embedded_downmix << j); ++ for (i = 0; i < s->xxch_nbits_spk_mask; i++) { ++ if (mask[j] & (1 << i)) { ++ if ((1 << i) == DCA_XXCH_LFE1) { ++ av_log(s->avctx, AV_LOG_WARNING, ++ "DCA-XXCH: dmix to LFE1 not supported.\n"); ++ continue; ++ } ++ ++ coeff = get_bits(&s->gb, 7); ++ sign = (coeff & 64) ? 1.0 : -1.0; ++ mag = dca_downmix_scale_factors[((coeff & 63) - 1) << 2]; ++ ichan = dca_xxch2index(s, 1 << i); ++ s->xxch_dmix_coeff[j][ichan] = sign * mag; ++ } ++ } ++ } ++ } ++ } ++ ++ if (s->prim_channels > DCA_PRIM_CHANNELS_MAX) ++ s->prim_channels = DCA_PRIM_CHANNELS_MAX; ++ ++ ++ for (i = base_channel; i < s->prim_channels; i++) { ++ s->subband_activity[i] = get_bits(&s->gb, 5) + 2; ++ if (s->subband_activity[i] > DCA_SUBBANDS) ++ s->subband_activity[i] = DCA_SUBBANDS; ++ } ++ for (i = base_channel; i < s->prim_channels; i++) { ++ s->vq_start_subband[i] = get_bits(&s->gb, 5) + 1; ++ if (s->vq_start_subband[i] > DCA_SUBBANDS) ++ s->vq_start_subband[i] = DCA_SUBBANDS; ++ } ++ get_array(&s->gb, s->joint_intensity + base_channel, s->prim_channels - base_channel, 3); ++ get_array(&s->gb, s->transient_huffman + base_channel, s->prim_channels - base_channel, 2); ++ get_array(&s->gb, s->scalefactor_huffman + base_channel, s->prim_channels - base_channel, 3); ++ get_array(&s->gb, s->bitalloc_huffman + base_channel, s->prim_channels - base_channel, 3); ++ ++ /* Get codebooks quantization indexes */ ++ if (!base_channel) ++ memset(s->quant_index_huffman, 0, sizeof(s->quant_index_huffman)); ++ for (j = 1; j < 11; j++) ++ for (i = base_channel; i < s->prim_channels; i++) ++ s->quant_index_huffman[i][j] = get_bits(&s->gb, bitlen[j]); ++ ++ /* Get scale factor adjustment */ ++ for (j = 0; j < 11; j++) ++ for (i = base_channel; i < s->prim_channels; i++) ++ s->scalefactor_adj[i][j] = 1; ++ ++ for (j = 1; j < 11; j++) ++ for (i = base_channel; i < s->prim_channels; i++) ++ if (s->quant_index_huffman[i][j] < thr[j]) ++ s->scalefactor_adj[i][j] = adj_table[get_bits(&s->gb, 2)]; ++ ++ if (!xxch) { ++ if (s->crc_present) { ++ /* Audio header CRC check */ ++ get_bits(&s->gb, 16); ++ } ++ } else { ++ /* Skip to the end of the header, also ignore CRC if present */ ++ i = get_bits_count(&s->gb); ++ if (hdr_pos + 8 * hdr_size > i) ++ skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i); ++ } ++ ++ s->current_subframe = 0; ++ s->current_subsubframe = 0; ++ ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "subframes: %i\n", s->subframes); ++ av_log(s->avctx, AV_LOG_DEBUG, "prim channels: %i\n", s->prim_channels); ++ for (i = base_channel; i < s->prim_channels; i++) { ++ av_log(s->avctx, AV_LOG_DEBUG, "subband activity: %i\n", ++ s->subband_activity[i]); ++ av_log(s->avctx, AV_LOG_DEBUG, "vq start subband: %i\n", ++ s->vq_start_subband[i]); ++ av_log(s->avctx, AV_LOG_DEBUG, "joint intensity: %i\n", ++ s->joint_intensity[i]); ++ av_log(s->avctx, AV_LOG_DEBUG, "transient mode codebook: %i\n", ++ s->transient_huffman[i]); ++ av_log(s->avctx, AV_LOG_DEBUG, "scale factor codebook: %i\n", ++ s->scalefactor_huffman[i]); ++ av_log(s->avctx, AV_LOG_DEBUG, "bit allocation quantizer: %i\n", ++ s->bitalloc_huffman[i]); ++ av_log(s->avctx, AV_LOG_DEBUG, "quant index huff:"); ++ for (j = 0; j < 11; j++) ++ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->quant_index_huffman[i][j]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ av_log(s->avctx, AV_LOG_DEBUG, "scalefac adj:"); ++ for (j = 0; j < 11; j++) ++ av_log(s->avctx, AV_LOG_DEBUG, " %1.3f", s->scalefactor_adj[i][j]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++#endif ++ ++ return 0; ++} ++ ++static int dca_parse_frame_header(DCAContext *s) ++{ ++ init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); ++ ++ /* Sync code */ ++ skip_bits_long(&s->gb, 32); ++ ++ /* Frame header */ ++ s->frame_type = get_bits(&s->gb, 1); ++ s->samples_deficit = get_bits(&s->gb, 5) + 1; ++ s->crc_present = get_bits(&s->gb, 1); ++ s->sample_blocks = get_bits(&s->gb, 7) + 1; ++ s->frame_size = get_bits(&s->gb, 14) + 1; ++ if (s->frame_size < 95) ++ return AVERROR_INVALIDDATA; ++ s->amode = get_bits(&s->gb, 6); ++ s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)]; ++ if (!s->sample_rate) ++ return AVERROR_INVALIDDATA; ++ s->bit_rate_index = get_bits(&s->gb, 5); ++ s->bit_rate = dca_bit_rates[s->bit_rate_index]; ++ if (!s->bit_rate) ++ return AVERROR_INVALIDDATA; ++ ++ s->downmix = get_bits(&s->gb, 1); /* note: this is FixedBit == 0 */ ++ s->dynrange = get_bits(&s->gb, 1); ++ s->timestamp = get_bits(&s->gb, 1); ++ s->aux_data = get_bits(&s->gb, 1); ++ s->hdcd = get_bits(&s->gb, 1); ++ s->ext_descr = get_bits(&s->gb, 3); ++ s->ext_coding = get_bits(&s->gb, 1); ++ s->aspf = get_bits(&s->gb, 1); ++ s->lfe = get_bits(&s->gb, 2); ++ s->predictor_history = get_bits(&s->gb, 1); ++ ++ if (s->lfe == 3) { ++ s->lfe = 0; ++ av_log_ask_for_sample(s->avctx, "LFE is 3\n"); ++ return AVERROR_PATCHWELCOME; ++ } ++ ++ /* TODO: check CRC */ ++ if (s->crc_present) ++ s->header_crc = get_bits(&s->gb, 16); ++ ++ s->multirate_inter = get_bits(&s->gb, 1); ++ s->version = get_bits(&s->gb, 4); ++ s->copy_history = get_bits(&s->gb, 2); ++ s->source_pcm_res = get_bits(&s->gb, 3); ++ s->front_sum = get_bits(&s->gb, 1); ++ s->surround_sum = get_bits(&s->gb, 1); ++ s->dialog_norm = get_bits(&s->gb, 4); ++ ++ /* FIXME: channels mixing levels */ ++ s->output = s->amode; ++ if (s->lfe) ++ s->output |= DCA_LFE; ++ ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "frame type: %i\n", s->frame_type); ++ av_log(s->avctx, AV_LOG_DEBUG, "samples deficit: %i\n", s->samples_deficit); ++ av_log(s->avctx, AV_LOG_DEBUG, "crc present: %i\n", s->crc_present); ++ av_log(s->avctx, AV_LOG_DEBUG, "sample blocks: %i (%i samples)\n", ++ s->sample_blocks, s->sample_blocks * 32); ++ av_log(s->avctx, AV_LOG_DEBUG, "frame size: %i bytes\n", s->frame_size); ++ av_log(s->avctx, AV_LOG_DEBUG, "amode: %i (%i channels)\n", ++ s->amode, dca_channels[s->amode]); ++ av_log(s->avctx, AV_LOG_DEBUG, "sample rate: %i Hz\n", ++ s->sample_rate); ++ av_log(s->avctx, AV_LOG_DEBUG, "bit rate: %i bits/s\n", ++ s->bit_rate); ++ av_log(s->avctx, AV_LOG_DEBUG, "downmix: %i\n", s->downmix); ++ av_log(s->avctx, AV_LOG_DEBUG, "dynrange: %i\n", s->dynrange); ++ av_log(s->avctx, AV_LOG_DEBUG, "timestamp: %i\n", s->timestamp); ++ av_log(s->avctx, AV_LOG_DEBUG, "aux_data: %i\n", s->aux_data); ++ av_log(s->avctx, AV_LOG_DEBUG, "hdcd: %i\n", s->hdcd); ++ av_log(s->avctx, AV_LOG_DEBUG, "ext descr: %i\n", s->ext_descr); ++ av_log(s->avctx, AV_LOG_DEBUG, "ext coding: %i\n", s->ext_coding); ++ av_log(s->avctx, AV_LOG_DEBUG, "aspf: %i\n", s->aspf); ++ av_log(s->avctx, AV_LOG_DEBUG, "lfe: %i\n", s->lfe); ++ av_log(s->avctx, AV_LOG_DEBUG, "predictor history: %i\n", ++ s->predictor_history); ++ av_log(s->avctx, AV_LOG_DEBUG, "header crc: %i\n", s->header_crc); ++ av_log(s->avctx, AV_LOG_DEBUG, "multirate inter: %i\n", ++ s->multirate_inter); ++ av_log(s->avctx, AV_LOG_DEBUG, "version number: %i\n", s->version); ++ av_log(s->avctx, AV_LOG_DEBUG, "copy history: %i\n", s->copy_history); ++ av_log(s->avctx, AV_LOG_DEBUG, ++ "source pcm resolution: %i (%i bits/sample)\n", ++ s->source_pcm_res, dca_bits_per_sample[s->source_pcm_res]); ++ av_log(s->avctx, AV_LOG_DEBUG, "front sum: %i\n", s->front_sum); ++ av_log(s->avctx, AV_LOG_DEBUG, "surround sum: %i\n", s->surround_sum); ++ av_log(s->avctx, AV_LOG_DEBUG, "dialog norm: %i\n", s->dialog_norm); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++#endif ++ ++ /* Primary audio coding header */ ++ s->subframes = get_bits(&s->gb, 4) + 1; ++ ++ return dca_parse_audio_coding_header(s, 0, 0); ++} ++ ++ ++static inline int get_scale(GetBitContext *gb, int level, int value, int log2range) ++{ ++ if (level < 5) { ++ /* huffman encoded */ ++ value += get_bitalloc(gb, &dca_scalefactor, level); ++ value = av_clip(value, 0, (1 << log2range) - 1); ++ } else if (level < 8) { ++ if (level + 1 > log2range) { ++ skip_bits(gb, level + 1 - log2range); ++ value = get_bits(gb, log2range); ++ } else { ++ value = get_bits(gb, level + 1); ++ } ++ } ++ return value; ++} ++ ++static int dca_subframe_header(DCAContext *s, int base_channel, int block_index) ++{ ++ /* Primary audio coding side information */ ++ int j, k; ++ ++ if (get_bits_left(&s->gb) < 0) ++ return AVERROR_INVALIDDATA; ++ ++ if (!base_channel) { ++ s->subsubframes[s->current_subframe] = get_bits(&s->gb, 2) + 1; ++ s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3); ++ } ++ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ for (k = 0; k < s->subband_activity[j]; k++) ++ s->prediction_mode[j][k] = get_bits(&s->gb, 1); ++ } ++ ++ /* Get prediction codebook */ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ for (k = 0; k < s->subband_activity[j]; k++) { ++ if (s->prediction_mode[j][k] > 0) { ++ /* (Prediction coefficient VQ address) */ ++ s->prediction_vq[j][k] = get_bits(&s->gb, 12); ++ } ++ } ++ } ++ ++ /* Bit allocation index */ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ for (k = 0; k < s->vq_start_subband[j]; k++) { ++ if (s->bitalloc_huffman[j] == 6) ++ s->bitalloc[j][k] = get_bits(&s->gb, 5); ++ else if (s->bitalloc_huffman[j] == 5) ++ s->bitalloc[j][k] = get_bits(&s->gb, 4); ++ else if (s->bitalloc_huffman[j] == 7) { ++ av_log(s->avctx, AV_LOG_ERROR, ++ "Invalid bit allocation index\n"); ++ return AVERROR_INVALIDDATA; ++ } else { ++ s->bitalloc[j][k] = ++ get_bitalloc(&s->gb, &dca_bitalloc_index, s->bitalloc_huffman[j]); ++ } ++ ++ if (s->bitalloc[j][k] > 26) { ++ av_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n", ++ j, k, s->bitalloc[j][k]); ++ return AVERROR_INVALIDDATA; ++ } ++ } ++ } ++ ++ /* Transition mode */ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ for (k = 0; k < s->subband_activity[j]; k++) { ++ s->transition_mode[j][k] = 0; ++ if (s->subsubframes[s->current_subframe] > 1 && ++ k < s->vq_start_subband[j] && s->bitalloc[j][k] > 0) { ++ s->transition_mode[j][k] = ++ get_bitalloc(&s->gb, &dca_tmode, s->transient_huffman[j]); ++ } ++ } ++ } ++ ++ if (get_bits_left(&s->gb) < 0) ++ return AVERROR_INVALIDDATA; ++ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ const uint32_t *scale_table; ++ int scale_sum, log_size; ++ ++ memset(s->scale_factor[j], 0, ++ s->subband_activity[j] * sizeof(s->scale_factor[0][0][0]) * 2); ++ ++ if (s->scalefactor_huffman[j] == 6) { ++ scale_table = scale_factor_quant7; ++ log_size = 7; ++ } else { ++ scale_table = scale_factor_quant6; ++ log_size = 6; ++ } ++ ++ /* When huffman coded, only the difference is encoded */ ++ scale_sum = 0; ++ ++ for (k = 0; k < s->subband_activity[j]; k++) { ++ if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) { ++ scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size); ++ s->scale_factor[j][k][0] = scale_table[scale_sum]; ++ } ++ ++ if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) { ++ /* Get second scale factor */ ++ scale_sum = get_scale(&s->gb, s->scalefactor_huffman[j], scale_sum, log_size); ++ s->scale_factor[j][k][1] = scale_table[scale_sum]; ++ } ++ } ++ } ++ ++ /* Joint subband scale factor codebook select */ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ /* Transmitted only if joint subband coding enabled */ ++ if (s->joint_intensity[j] > 0) ++ s->joint_huff[j] = get_bits(&s->gb, 3); ++ } ++ ++ if (get_bits_left(&s->gb) < 0) ++ return AVERROR_INVALIDDATA; ++ ++ /* Scale factors for joint subband coding */ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ int source_channel; ++ ++ /* Transmitted only if joint subband coding enabled */ ++ if (s->joint_intensity[j] > 0) { ++ int scale = 0; ++ source_channel = s->joint_intensity[j] - 1; ++ ++ /* When huffman coded, only the difference is encoded ++ * (is this valid as well for joint scales ???) */ ++ ++ for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) { ++ scale = get_scale(&s->gb, s->joint_huff[j], 64 /* bias */, 7); ++ s->joint_scale_factor[j][k] = scale; /*joint_scale_table[scale]; */ ++ } ++ ++ if (!(s->debug_flag & 0x02)) { ++ av_log(s->avctx, AV_LOG_DEBUG, ++ "Joint stereo coding not supported\n"); ++ s->debug_flag |= 0x02; ++ } ++ } ++ } ++ ++ /* Stereo downmix coefficients */ ++ if (!base_channel && s->prim_channels > 2) { ++ if (s->downmix) { ++ for (j = base_channel; j < s->prim_channels; j++) { ++ s->downmix_coef[j][0] = get_bits(&s->gb, 7); ++ s->downmix_coef[j][1] = get_bits(&s->gb, 7); ++ } ++ } else { ++ int am = s->amode & DCA_CHANNEL_MASK; ++ if (am >= FF_ARRAY_ELEMS(dca_default_coeffs)) { ++ av_log(s->avctx, AV_LOG_ERROR, ++ "Invalid channel mode %d\n", am); ++ return AVERROR_INVALIDDATA; ++ } ++ for (j = base_channel; j < FFMIN(s->prim_channels, FF_ARRAY_ELEMS(dca_default_coeffs[am])); j++) { ++ s->downmix_coef[j][0] = dca_default_coeffs[am][j][0]; ++ s->downmix_coef[j][1] = dca_default_coeffs[am][j][1]; ++ } ++ } ++ } ++ ++ /* Dynamic range coefficient */ ++ if (!base_channel && s->dynrange) ++ s->dynrange_coef = get_bits(&s->gb, 8); ++ ++ /* Side information CRC check word */ ++ if (s->crc_present) { ++ get_bits(&s->gb, 16); ++ } ++ ++ /* ++ * Primary audio data arrays ++ */ ++ ++ /* VQ encoded high frequency subbands */ ++ for (j = base_channel; j < s->prim_channels; j++) ++ for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) ++ /* 1 vector -> 32 samples */ ++ s->high_freq_vq[j][k] = get_bits(&s->gb, 10); ++ ++ /* Low frequency effect data */ ++ if (!base_channel && s->lfe) { ++ int quant7; ++ /* LFE samples */ ++ int lfe_samples = 2 * s->lfe * (4 + block_index); ++ int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]); ++ float lfe_scale; ++ ++ for (j = lfe_samples; j < lfe_end_sample; j++) { ++ /* Signed 8 bits int */ ++ s->lfe_data[j] = get_sbits(&s->gb, 8); ++ } ++ ++ /* Scale factor index */ ++ quant7 = get_bits(&s->gb, 8); ++ if (quant7 > 127) { ++ av_log_ask_for_sample(s->avctx, "LFEScaleIndex larger than 127\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ s->lfe_scale_factor = scale_factor_quant7[quant7]; ++ ++ /* Quantization step size * scale factor */ ++ lfe_scale = 0.035 * s->lfe_scale_factor; ++ ++ for (j = lfe_samples; j < lfe_end_sample; j++) ++ s->lfe_data[j] *= lfe_scale; ++ } ++ ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "subsubframes: %i\n", ++ s->subsubframes[s->current_subframe]); ++ av_log(s->avctx, AV_LOG_DEBUG, "partial samples: %i\n", ++ s->partial_samples[s->current_subframe]); ++ ++ for (j = base_channel; j < s->prim_channels; j++) { ++ av_log(s->avctx, AV_LOG_DEBUG, "prediction mode:"); ++ for (k = 0; k < s->subband_activity[j]; k++) ++ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->prediction_mode[j][k]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++ for (j = base_channel; j < s->prim_channels; j++) { ++ for (k = 0; k < s->subband_activity[j]; k++) ++ av_log(s->avctx, AV_LOG_DEBUG, ++ "prediction coefs: %f, %f, %f, %f\n", ++ (float) adpcm_vb[s->prediction_vq[j][k]][0] / 8192, ++ (float) adpcm_vb[s->prediction_vq[j][k]][1] / 8192, ++ (float) adpcm_vb[s->prediction_vq[j][k]][2] / 8192, ++ (float) adpcm_vb[s->prediction_vq[j][k]][3] / 8192); ++ } ++ for (j = base_channel; j < s->prim_channels; j++) { ++ av_log(s->avctx, AV_LOG_DEBUG, "bitalloc index: "); ++ for (k = 0; k < s->vq_start_subband[j]; k++) ++ av_log(s->avctx, AV_LOG_DEBUG, "%2.2i ", s->bitalloc[j][k]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++ for (j = base_channel; j < s->prim_channels; j++) { ++ av_log(s->avctx, AV_LOG_DEBUG, "Transition mode:"); ++ for (k = 0; k < s->subband_activity[j]; k++) ++ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->transition_mode[j][k]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++ for (j = base_channel; j < s->prim_channels; j++) { ++ av_log(s->avctx, AV_LOG_DEBUG, "Scale factor:"); ++ for (k = 0; k < s->subband_activity[j]; k++) { ++ if (k >= s->vq_start_subband[j] || s->bitalloc[j][k] > 0) ++ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->scale_factor[j][k][0]); ++ if (k < s->vq_start_subband[j] && s->transition_mode[j][k]) ++ av_log(s->avctx, AV_LOG_DEBUG, " %i(t)", s->scale_factor[j][k][1]); ++ } ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++ for (j = base_channel; j < s->prim_channels; j++) { ++ if (s->joint_intensity[j] > 0) { ++ int source_channel = s->joint_intensity[j] - 1; ++ av_log(s->avctx, AV_LOG_DEBUG, "Joint scale factor index:\n"); ++ for (k = s->subband_activity[j]; k < s->subband_activity[source_channel]; k++) ++ av_log(s->avctx, AV_LOG_DEBUG, " %i", s->joint_scale_factor[j][k]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++ } ++ if (!base_channel && s->prim_channels > 2 && s->downmix) { ++ av_log(s->avctx, AV_LOG_DEBUG, "Downmix coeffs:\n"); ++ for (j = 0; j < s->prim_channels; j++) { ++ av_log(s->avctx, AV_LOG_DEBUG, "Channel 0, %d = %f\n", j, ++ dca_downmix_coeffs[s->downmix_coef[j][0]]); ++ av_log(s->avctx, AV_LOG_DEBUG, "Channel 1, %d = %f\n", j, ++ dca_downmix_coeffs[s->downmix_coef[j][1]]); ++ } ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++ for (j = base_channel; j < s->prim_channels; j++) ++ for (k = s->vq_start_subband[j]; k < s->subband_activity[j]; k++) ++ av_log(s->avctx, AV_LOG_DEBUG, "VQ index: %i\n", s->high_freq_vq[j][k]); ++ if (!base_channel && s->lfe) { ++ int lfe_samples = 2 * s->lfe * (4 + block_index); ++ int lfe_end_sample = 2 * s->lfe * (4 + block_index + s->subsubframes[s->current_subframe]); ++ ++ av_log(s->avctx, AV_LOG_DEBUG, "LFE samples:\n"); ++ for (j = lfe_samples; j < lfe_end_sample; j++) ++ av_log(s->avctx, AV_LOG_DEBUG, " %f", s->lfe_data[j]); ++ av_log(s->avctx, AV_LOG_DEBUG, "\n"); ++ } ++#endif ++ ++ return 0; ++} ++ ++static void qmf_32_subbands(DCAContext *s, int chans, ++ float samples_in[32][8], float *samples_out, ++ float scale) ++{ ++ const float *prCoeff; ++ int i; ++ ++ int sb_act = s->subband_activity[chans]; ++ int subindex; ++ ++ scale *= sqrt(1 / 8.0); ++ ++ /* Select filter */ ++ if (!s->multirate_inter) /* Non-perfect reconstruction */ ++ prCoeff = fir_32bands_nonperfect; ++ else /* Perfect reconstruction */ ++ prCoeff = fir_32bands_perfect; ++ ++ for (i = sb_act; i < 32; i++) ++ s->raXin[i] = 0.0; ++ ++ /* Reconstructed channel sample index */ ++ for (subindex = 0; subindex < 8; subindex++) { ++ /* Load in one sample from each subband and clear inactive subbands */ ++ for (i = 0; i < sb_act; i++) { ++ unsigned sign = (i - 1) & 2; ++ uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30; ++ AV_WN32A(&s->raXin[i], v); ++ } ++ ++ s->synth.synth_filter_float(&s->imdct, ++ s->subband_fir_hist[chans], ++ &s->hist_index[chans], ++ s->subband_fir_noidea[chans], prCoeff, ++ samples_out, s->raXin, scale); ++ samples_out += 32; ++ } ++} ++ ++static void lfe_interpolation_fir(DCAContext *s, int decimation_select, ++ int num_deci_sample, float *samples_in, ++ float *samples_out, float scale) ++{ ++ /* samples_in: An array holding decimated samples. ++ * Samples in current subframe starts from samples_in[0], ++ * while samples_in[-1], samples_in[-2], ..., stores samples ++ * from last subframe as history. ++ * ++ * samples_out: An array holding interpolated samples ++ */ ++ ++ int decifactor; ++ const float *prCoeff; ++ int deciindex; ++ ++ /* Select decimation filter */ ++ if (decimation_select == 1) { ++ decifactor = 64; ++ prCoeff = lfe_fir_128; ++ } else { ++ decifactor = 32; ++ prCoeff = lfe_fir_64; ++ } ++ /* Interpolation */ ++ for (deciindex = 0; deciindex < num_deci_sample; deciindex++) { ++ s->dcadsp.lfe_fir(samples_out, samples_in, prCoeff, decifactor, scale); ++ samples_in++; ++ samples_out += 2 * decifactor; ++ } ++} ++ ++/* downmixing routines */ ++#define MIX_REAR1(samples, s1, rs, coef) \ ++ samples[0][i] += samples[s1][i] * coef[rs][0]; \ ++ samples[1][i] += samples[s1][i] * coef[rs][1]; ++ ++#define MIX_REAR2(samples, s1, s2, rs, coef) \ ++ samples[0][i] += samples[s1][i] * coef[rs][0] + samples[s2][i] * coef[rs + 1][0]; \ ++ samples[1][i] += samples[s1][i] * coef[rs][1] + samples[s2][i] * coef[rs + 1][1]; ++ ++#define MIX_FRONT3(samples, coef) \ ++ t = samples[c][i]; \ ++ u = samples[l][i]; \ ++ v = samples[r][i]; \ ++ samples[0][i] = t * coef[0][0] + u * coef[1][0] + v * coef[2][0]; \ ++ samples[1][i] = t * coef[0][1] + u * coef[1][1] + v * coef[2][1]; ++ ++#define DOWNMIX_TO_STEREO(op1, op2) \ ++ for (i = 0; i < 256; i++) { \ ++ op1 \ ++ op2 \ ++ } ++ ++static void dca_downmix(float **samples, int srcfmt, ++ int downmix_coef[DCA_PRIM_CHANNELS_MAX][2], ++ const int8_t *channel_mapping) ++{ ++ int c, l, r, sl, sr, s; ++ int i; ++ float t, u, v; ++ float coef[DCA_PRIM_CHANNELS_MAX][2]; ++ ++ for (i = 0; i < DCA_PRIM_CHANNELS_MAX; i++) { ++ coef[i][0] = dca_downmix_coeffs[downmix_coef[i][0]]; ++ coef[i][1] = dca_downmix_coeffs[downmix_coef[i][1]]; ++ } ++ ++ switch (srcfmt) { ++ case DCA_MONO: ++ case DCA_CHANNEL: ++ case DCA_STEREO_TOTAL: ++ case DCA_STEREO_SUMDIFF: ++ case DCA_4F2R: ++ av_log(NULL, AV_LOG_ERROR, "Not implemented!\n"); ++ break; ++ case DCA_STEREO: ++ break; ++ case DCA_3F: ++ c = channel_mapping[0]; ++ l = channel_mapping[1]; ++ r = channel_mapping[2]; ++ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), ); ++ break; ++ case DCA_2F1R: ++ s = channel_mapping[2]; ++ DOWNMIX_TO_STEREO(MIX_REAR1(samples, s, 2, coef), ); ++ break; ++ case DCA_3F1R: ++ c = channel_mapping[0]; ++ l = channel_mapping[1]; ++ r = channel_mapping[2]; ++ s = channel_mapping[3]; ++ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), ++ MIX_REAR1(samples, s, 3, coef)); ++ break; ++ case DCA_2F2R: ++ sl = channel_mapping[2]; ++ sr = channel_mapping[3]; ++ DOWNMIX_TO_STEREO(MIX_REAR2(samples, sl, sr, 2, coef), ); ++ break; ++ case DCA_3F2R: ++ c = channel_mapping[0]; ++ l = channel_mapping[1]; ++ r = channel_mapping[2]; ++ sl = channel_mapping[3]; ++ sr = channel_mapping[4]; ++ DOWNMIX_TO_STEREO(MIX_FRONT3(samples, coef), ++ MIX_REAR2(samples, sl, sr, 3, coef)); ++ break; ++ } ++} ++ ++ ++#ifndef decode_blockcodes ++/* Very compact version of the block code decoder that does not use table ++ * look-up but is slightly slower */ ++static int decode_blockcode(int code, int levels, int *values) ++{ ++ int i; ++ int offset = (levels - 1) >> 1; ++ ++ for (i = 0; i < 4; i++) { ++ int div = FASTDIV(code, levels); ++ values[i] = code - offset - div * levels; ++ code = div; ++ } ++ ++ return code; ++} ++ ++static int decode_blockcodes(int code1, int code2, int levels, int *values) ++{ ++ return decode_blockcode(code1, levels, values) | ++ decode_blockcode(code2, levels, values + 4); ++} ++#endif ++ ++static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 }; ++static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 }; ++ ++#ifndef int8x8_fmul_int32 ++static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale) ++{ ++ float fscale = scale / 16.0; ++ int i; ++ for (i = 0; i < 8; i++) ++ dst[i] = src[i] * fscale; ++} ++#endif ++ ++static int dca_subsubframe(DCAContext *s, int base_channel, int block_index) ++{ ++ int k, l; ++ int subsubframe = s->current_subsubframe; ++ ++ const float *quant_step_table; ++ ++ /* FIXME */ ++ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index]; ++ LOCAL_ALIGNED_16(int, block, [8 * DCA_SUBBANDS]); ++ ++ /* ++ * Audio data ++ */ ++ ++ /* Select quantization step size table */ ++ if (s->bit_rate_index == 0x1f) ++ quant_step_table = lossless_quant_d; ++ else ++ quant_step_table = lossy_quant_d; ++ ++ for (k = base_channel; k < s->prim_channels; k++) { ++ float rscale[DCA_SUBBANDS]; ++ ++ if (get_bits_left(&s->gb) < 0) ++ return AVERROR_INVALIDDATA; ++ ++ for (l = 0; l < s->vq_start_subband[k]; l++) { ++ int m; ++ ++ /* Select the mid-tread linear quantizer */ ++ int abits = s->bitalloc[k][l]; ++ ++ float quant_step_size = quant_step_table[abits]; ++ ++ /* ++ * Determine quantization index code book and its type ++ */ ++ ++ /* Select quantization index code book */ ++ int sel = s->quant_index_huffman[k][abits]; ++ ++ /* ++ * Extract bits from the bit stream ++ */ ++ if (!abits) { ++ rscale[l] = 0; ++ memset(block + 8 * l, 0, 8 * sizeof(block[0])); ++ } else { ++ /* Deal with transients */ ++ int sfi = s->transition_mode[k][l] && subsubframe >= s->transition_mode[k][l]; ++ rscale[l] = quant_step_size * s->scale_factor[k][l][sfi] * ++ s->scalefactor_adj[k][sel]; ++ ++ if (abits >= 11 || !dca_smpl_bitalloc[abits].vlc[sel].table) { ++ if (abits <= 7) { ++ /* Block code */ ++ int block_code1, block_code2, size, levels, err; ++ ++ size = abits_sizes[abits - 1]; ++ levels = abits_levels[abits - 1]; ++ ++ block_code1 = get_bits(&s->gb, size); ++ block_code2 = get_bits(&s->gb, size); ++ err = decode_blockcodes(block_code1, block_code2, ++ levels, block + 8 * l); ++ if (err) { ++ av_log(s->avctx, AV_LOG_ERROR, ++ "ERROR: block code look-up failed\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ } else { ++ /* no coding */ ++ for (m = 0; m < 8; m++) ++ block[8 * l + m] = get_sbits(&s->gb, abits - 3); ++ } ++ } else { ++ /* Huffman coded */ ++ for (m = 0; m < 8; m++) ++ block[8 * l + m] = get_bitalloc(&s->gb, ++ &dca_smpl_bitalloc[abits], sel); ++ } ++ ++ } ++ } ++ ++ s->fmt_conv.int32_to_float_fmul_scalar_array(&s->fmt_conv, subband_samples[k][0], ++ block, rscale, 8 * s->vq_start_subband[k]); ++ ++ for (l = 0; l < s->vq_start_subband[k]; l++) { ++ int m; ++ /* ++ * Inverse ADPCM if in prediction mode ++ */ ++ if (s->prediction_mode[k][l]) { ++ int n; ++ for (m = 0; m < 8; m++) { ++ for (n = 1; n <= 4; n++) ++ if (m >= n) ++ subband_samples[k][l][m] += ++ (adpcm_vb[s->prediction_vq[k][l]][n - 1] * ++ subband_samples[k][l][m - n] / 8192); ++ else if (s->predictor_history) ++ subband_samples[k][l][m] += ++ (adpcm_vb[s->prediction_vq[k][l]][n - 1] * ++ s->subband_samples_hist[k][l][m - n + 4] / 8192); ++ } ++ } ++ } ++ ++ /* ++ * Decode VQ encoded high frequencies ++ */ ++ for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; l++) { ++ /* 1 vector -> 32 samples but we only need the 8 samples ++ * for this subsubframe. */ ++ int hfvq = s->high_freq_vq[k][l]; ++ ++ if (!s->debug_flag & 0x01) { ++ av_log(s->avctx, AV_LOG_DEBUG, ++ "Stream with high frequencies VQ coding\n"); ++ s->debug_flag |= 0x01; ++ } ++ ++ int8x8_fmul_int32(subband_samples[k][l], ++ &high_freq_vq[hfvq][subsubframe * 8], ++ s->scale_factor[k][l][0]); ++ } ++ } ++ ++ /* Check for DSYNC after subsubframe */ ++ if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) { ++ if (0xFFFF == get_bits(&s->gb, 16)) { /* 0xFFFF */ ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "Got subframe DSYNC\n"); ++#endif ++ } else { ++ av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n"); ++ } ++ } ++ ++ /* Backup predictor history for adpcm */ ++ for (k = base_channel; k < s->prim_channels; k++) ++ for (l = 0; l < s->vq_start_subband[k]; l++) ++ memcpy(s->subband_samples_hist[k][l], ++ &subband_samples[k][l][4], ++ 4 * sizeof(subband_samples[0][0][0])); ++ ++ return 0; ++} ++ ++static int dca_filter_channels(DCAContext *s, int block_index) ++{ ++ float (*subband_samples)[DCA_SUBBANDS][8] = s->subband_samples[block_index]; ++ int k; ++ ++ /* 32 subbands QMF */ ++ for (k = 0; k < s->prim_channels; k++) { ++/* static float pcm_to_double[8] = { 32768.0, 32768.0, 524288.0, 524288.0, ++ 0, 8388608.0, 8388608.0 };*/ ++ if (s->channel_order_tab[k] >= 0) ++ qmf_32_subbands(s, k, subband_samples[k], ++ s->samples_chanptr[s->channel_order_tab[k]], ++ M_SQRT1_2 / 32768.0 /* pcm_to_double[s->source_pcm_res] */); ++ } ++ ++ /* Down mixing */ ++ if (s->avctx->request_channels == 2 && s->prim_channels > 2) { ++ dca_downmix(s->samples_chanptr, s->amode, s->downmix_coef, s->channel_order_tab); ++ } ++ ++ /* Generate LFE samples for this subsubframe FIXME!!! */ ++ if (s->output & DCA_LFE) { ++ lfe_interpolation_fir(s, s->lfe, 2 * s->lfe, ++ s->lfe_data + 2 * s->lfe * (block_index + 4), ++ s->samples_chanptr[s->lfe_index], ++ 1.0 / (256.0 * 32768.0)); ++ /* Outputs 20bits pcm samples */ ++ } ++ ++ return 0; ++} ++ ++ ++static int dca_subframe_footer(DCAContext *s, int base_channel) ++{ ++ int aux_data_count = 0, i; ++ ++ /* ++ * Unpack optional information ++ */ ++ ++ /* presumably optional information only appears in the core? */ ++ if (!base_channel) { ++ if (s->timestamp) ++ skip_bits_long(&s->gb, 32); ++ ++ if (s->aux_data) ++ aux_data_count = get_bits(&s->gb, 6); ++ ++ for (i = 0; i < aux_data_count; i++) ++ get_bits(&s->gb, 8); ++ ++ if (s->crc_present && (s->downmix || s->dynrange)) ++ get_bits(&s->gb, 16); ++ } ++ ++ return 0; ++} ++ ++/** ++ * Decode a dca frame block ++ * ++ * @param s pointer to the DCAContext ++ */ ++ ++static int dca_decode_block(DCAContext *s, int base_channel, int block_index) ++{ ++ int ret; ++ ++ /* Sanity check */ ++ if (s->current_subframe >= s->subframes) { ++ av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i", ++ s->current_subframe, s->subframes); ++ return AVERROR_INVALIDDATA; ++ } ++ ++ if (!s->current_subsubframe) { ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_header\n"); ++#endif ++ /* Read subframe header */ ++ if ((ret = dca_subframe_header(s, base_channel, block_index))) ++ return ret; ++ } ++ ++ /* Read subsubframe */ ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subsubframe\n"); ++#endif ++ if ((ret = dca_subsubframe(s, base_channel, block_index))) ++ return ret; ++ ++ /* Update state */ ++ s->current_subsubframe++; ++ if (s->current_subsubframe >= s->subsubframes[s->current_subframe]) { ++ s->current_subsubframe = 0; ++ s->current_subframe++; ++ } ++ if (s->current_subframe >= s->subframes) { ++#ifdef TRACE ++ av_log(s->avctx, AV_LOG_DEBUG, "DSYNC dca_subframe_footer\n"); ++#endif ++ /* Read subframe footer */ ++ if ((ret = dca_subframe_footer(s, base_channel))) ++ return ret; ++ } ++ ++ return 0; ++} ++ ++/** ++ * Return the number of channels in an ExSS speaker mask (HD) ++ */ ++static int dca_exss_mask2count(int mask) ++{ ++ /* count bits that mean speaker pairs twice */ ++ return av_popcount(mask) + ++ av_popcount(mask & (DCA_EXSS_CENTER_LEFT_RIGHT | ++ DCA_EXSS_FRONT_LEFT_RIGHT | ++ DCA_EXSS_FRONT_HIGH_LEFT_RIGHT | ++ DCA_EXSS_WIDE_LEFT_RIGHT | ++ DCA_EXSS_SIDE_LEFT_RIGHT | ++ DCA_EXSS_SIDE_HIGH_LEFT_RIGHT | ++ DCA_EXSS_SIDE_REAR_LEFT_RIGHT | ++ DCA_EXSS_REAR_LEFT_RIGHT | ++ DCA_EXSS_REAR_HIGH_LEFT_RIGHT)); ++} ++ ++/** ++ * Skip mixing coefficients of a single mix out configuration (HD) ++ */ ++static void dca_exss_skip_mix_coeffs(GetBitContext *gb, int channels, int out_ch) ++{ ++ int i; ++ ++ for (i = 0; i < channels; i++) { ++ int mix_map_mask = get_bits(gb, out_ch); ++ int num_coeffs = av_popcount(mix_map_mask); ++ skip_bits_long(gb, num_coeffs * 6); ++ } ++} ++ ++/** ++ * Parse extension substream asset header (HD) ++ */ ++static int dca_exss_parse_asset_header(DCAContext *s) ++{ ++ int header_pos = get_bits_count(&s->gb); ++ int header_size; ++ int channels = 0; ++ int embedded_stereo = 0; ++ int embedded_6ch = 0; ++ int drc_code_present; ++ int av_uninit(extensions_mask); ++ int i, j; ++ ++ if (get_bits_left(&s->gb) < 16) ++ return -1; ++ ++ /* We will parse just enough to get to the extensions bitmask with which ++ * we can set the profile value. */ ++ ++ header_size = get_bits(&s->gb, 9) + 1; ++ skip_bits(&s->gb, 3); // asset index ++ ++ if (s->static_fields) { ++ if (get_bits1(&s->gb)) ++ skip_bits(&s->gb, 4); // asset type descriptor ++ if (get_bits1(&s->gb)) ++ skip_bits_long(&s->gb, 24); // language descriptor ++ ++ if (get_bits1(&s->gb)) { ++ /* How can one fit 1024 bytes of text here if the maximum value ++ * for the asset header size field above was 512 bytes? */ ++ int text_length = get_bits(&s->gb, 10) + 1; ++ if (get_bits_left(&s->gb) < text_length * 8) ++ return -1; ++ skip_bits_long(&s->gb, text_length * 8); // info text ++ } ++ ++ skip_bits(&s->gb, 5); // bit resolution - 1 ++ skip_bits(&s->gb, 4); // max sample rate code ++ channels = get_bits(&s->gb, 8) + 1; ++ ++ if (get_bits1(&s->gb)) { // 1-to-1 channels to speakers ++ int spkr_remap_sets; ++ int spkr_mask_size = 16; ++ int num_spkrs[7]; ++ ++ if (channels > 2) ++ embedded_stereo = get_bits1(&s->gb); ++ if (channels > 6) ++ embedded_6ch = get_bits1(&s->gb); ++ ++ if (get_bits1(&s->gb)) { ++ spkr_mask_size = (get_bits(&s->gb, 2) + 1) << 2; ++ skip_bits(&s->gb, spkr_mask_size); // spkr activity mask ++ } ++ ++ spkr_remap_sets = get_bits(&s->gb, 3); ++ ++ for (i = 0; i < spkr_remap_sets; i++) { ++ /* std layout mask for each remap set */ ++ num_spkrs[i] = dca_exss_mask2count(get_bits(&s->gb, spkr_mask_size)); ++ } ++ ++ for (i = 0; i < spkr_remap_sets; i++) { ++ int num_dec_ch_remaps = get_bits(&s->gb, 5) + 1; ++ if (get_bits_left(&s->gb) < 0) ++ return -1; ++ ++ for (j = 0; j < num_spkrs[i]; j++) { ++ int remap_dec_ch_mask = get_bits_long(&s->gb, num_dec_ch_remaps); ++ int num_dec_ch = av_popcount(remap_dec_ch_mask); ++ skip_bits_long(&s->gb, num_dec_ch * 5); // remap codes ++ } ++ } ++ ++ } else { ++ skip_bits(&s->gb, 3); // representation type ++ } ++ } ++ ++ drc_code_present = get_bits1(&s->gb); ++ if (drc_code_present) ++ get_bits(&s->gb, 8); // drc code ++ ++ if (get_bits1(&s->gb)) ++ skip_bits(&s->gb, 5); // dialog normalization code ++ ++ if (drc_code_present && embedded_stereo) ++ get_bits(&s->gb, 8); // drc stereo code ++ ++ if (s->mix_metadata && get_bits1(&s->gb)) { ++ skip_bits(&s->gb, 1); // external mix ++ skip_bits(&s->gb, 6); // post mix gain code ++ ++ if (get_bits(&s->gb, 2) != 3) // mixer drc code ++ skip_bits(&s->gb, 3); // drc limit ++ else ++ skip_bits(&s->gb, 8); // custom drc code ++ ++ if (get_bits1(&s->gb)) // channel specific scaling ++ for (i = 0; i < s->num_mix_configs; i++) ++ skip_bits_long(&s->gb, s->mix_config_num_ch[i] * 6); // scale codes ++ else ++ skip_bits_long(&s->gb, s->num_mix_configs * 6); // scale codes ++ ++ for (i = 0; i < s->num_mix_configs; i++) { ++ if (get_bits_left(&s->gb) < 0) ++ return -1; ++ dca_exss_skip_mix_coeffs(&s->gb, channels, s->mix_config_num_ch[i]); ++ if (embedded_6ch) ++ dca_exss_skip_mix_coeffs(&s->gb, 6, s->mix_config_num_ch[i]); ++ if (embedded_stereo) ++ dca_exss_skip_mix_coeffs(&s->gb, 2, s->mix_config_num_ch[i]); ++ } ++ } ++ ++ switch (get_bits(&s->gb, 2)) { ++ case 0: extensions_mask = get_bits(&s->gb, 12); break; ++ case 1: extensions_mask = DCA_EXT_EXSS_XLL; break; ++ case 2: extensions_mask = DCA_EXT_EXSS_LBR; break; ++ case 3: extensions_mask = 0; /* aux coding */ break; ++ } ++ ++ /* not parsed further, we were only interested in the extensions mask */ ++ ++ if (get_bits_left(&s->gb) < 0) ++ return -1; ++ ++ if (get_bits_count(&s->gb) - header_pos > header_size * 8) { ++ av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n"); ++ return -1; ++ } ++ skip_bits_long(&s->gb, header_pos + header_size * 8 - get_bits_count(&s->gb)); ++ ++ if (extensions_mask & DCA_EXT_EXSS_XLL) ++ s->profile = FF_PROFILE_DTS_HD_MA; ++ else if (extensions_mask & (DCA_EXT_EXSS_XBR | DCA_EXT_EXSS_X96 | ++ DCA_EXT_EXSS_XXCH)) ++ s->profile = FF_PROFILE_DTS_HD_HRA; ++ ++ if (!(extensions_mask & DCA_EXT_CORE)) ++ av_log(s->avctx, AV_LOG_WARNING, "DTS core detection mismatch.\n"); ++ if ((extensions_mask & DCA_CORE_EXTS) != s->core_ext_mask) ++ av_log(s->avctx, AV_LOG_WARNING, ++ "DTS extensions detection mismatch (%d, %d)\n", ++ extensions_mask & DCA_CORE_EXTS, s->core_ext_mask); ++ ++ return 0; ++} ++ ++static int dca_xbr_parse_frame(DCAContext *s) ++{ ++ int scale_table_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS][2]; ++ int active_bands[DCA_CHSETS_MAX][DCA_CHSET_CHANS_MAX]; ++ int abits_high[DCA_CHSET_CHANS_MAX][DCA_SUBBANDS]; ++ int anctemp[DCA_CHSET_CHANS_MAX]; ++ int chset_fsize[DCA_CHSETS_MAX]; ++ int n_xbr_ch[DCA_CHSETS_MAX]; ++ int hdr_size, num_chsets, xbr_tmode, hdr_pos; ++ int i, j, k, l, chset, chan_base; ++ ++ av_log(s->avctx, AV_LOG_DEBUG, "DTS-XBR: decoding XBR extension\n"); ++ ++ /* get bit position of sync header */ ++ hdr_pos = get_bits_count(&s->gb) - 32; ++ ++ hdr_size = get_bits(&s->gb, 6) + 1; ++ num_chsets = get_bits(&s->gb, 2) + 1; ++ ++ for(i = 0; i < num_chsets; i++) ++ chset_fsize[i] = get_bits(&s->gb, 14) + 1; ++ ++ xbr_tmode = get_bits1(&s->gb); ++ ++ for(i = 0; i < num_chsets; i++) { ++ n_xbr_ch[i] = get_bits(&s->gb, 3) + 1; ++ k = get_bits(&s->gb, 2) + 5; ++ for(j = 0; j < n_xbr_ch[i]; j++) ++ active_bands[i][j] = get_bits(&s->gb, k) + 1; ++ } ++ ++ /* skip to the end of the header */ ++ i = get_bits_count(&s->gb); ++ if(hdr_pos + hdr_size * 8 > i) ++ skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); ++ ++ /* loop over the channel data sets */ ++ /* only decode as many channels as we've decoded base data for */ ++ for(chset = 0, chan_base = 0; ++ chset < num_chsets && chan_base + n_xbr_ch[chset] <= s->prim_channels; ++ chan_base += n_xbr_ch[chset++]) { ++ int start_posn = get_bits_count(&s->gb); ++ int subsubframe = 0; ++ int subframe = 0; ++ ++ /* loop over subframes */ ++ for (k = 0; k < (s->sample_blocks / 8); k++) { ++ /* parse header if we're on first subsubframe of a block */ ++ if(subsubframe == 0) { ++ /* Parse subframe header */ ++ for(i = 0; i < n_xbr_ch[chset]; i++) { ++ anctemp[i] = get_bits(&s->gb, 2) + 2; ++ } ++ ++ for(i = 0; i < n_xbr_ch[chset]; i++) { ++ get_array(&s->gb, abits_high[i], active_bands[chset][i], anctemp[i]); ++ } ++ ++ for(i = 0; i < n_xbr_ch[chset]; i++) { ++ anctemp[i] = get_bits(&s->gb, 3); ++ if(anctemp[i] < 1) { ++ av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: SYNC ERROR\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ } ++ ++ /* generate scale factors */ ++ for(i = 0; i < n_xbr_ch[chset]; i++) { ++ const uint32_t *scale_table; ++ int nbits; ++ ++ if (s->scalefactor_huffman[chan_base+i] == 6) { ++ scale_table = scale_factor_quant7; ++ } else { ++ scale_table = scale_factor_quant6; ++ } ++ ++ nbits = anctemp[i]; ++ ++ for(j = 0; j < active_bands[chset][i]; j++) { ++ if(abits_high[i][j] > 0) { ++ scale_table_high[i][j][0] = ++ scale_table[get_bits(&s->gb, nbits)]; ++ ++ if(xbr_tmode && s->transition_mode[i][j]) { ++ scale_table_high[i][j][1] = ++ scale_table[get_bits(&s->gb, nbits)]; ++ } ++ } ++ } ++ } ++ } ++ ++ /* decode audio array for this block */ ++ for(i = 0; i < n_xbr_ch[chset]; i++) { ++ for(j = 0; j < active_bands[chset][i]; j++) { ++ const int xbr_abits = abits_high[i][j]; ++ const float quant_step_size = lossless_quant_d[xbr_abits]; ++ const int sfi = xbr_tmode && s->transition_mode[i][j] && subsubframe >= s->transition_mode[i][j]; ++ const float rscale = quant_step_size * scale_table_high[i][j][sfi]; ++ float *subband_samples = s->subband_samples[k][chan_base+i][j]; ++ int block[8]; ++ ++ if(xbr_abits <= 0) ++ continue; ++ ++ if(xbr_abits > 7) { ++ get_array(&s->gb, block, 8, xbr_abits - 3); ++ } else { ++ int block_code1, block_code2, size, levels, err; ++ ++ size = abits_sizes[xbr_abits - 1]; ++ levels = abits_levels[xbr_abits - 1]; ++ ++ block_code1 = get_bits(&s->gb, size); ++ block_code2 = get_bits(&s->gb, size); ++ err = decode_blockcodes(block_code1, block_code2, ++ levels, block); ++ if (err) { ++ av_log(s->avctx, AV_LOG_ERROR, ++ "ERROR: DTS-XBR: block code look-up failed\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ } ++ ++ /* scale & sum into subband */ ++ for(l = 0; l < 8; l++) ++ subband_samples[l] += (float)block[l] * rscale; ++ } ++ } ++ ++ /* check DSYNC marker */ ++ if(s->aspf || subsubframe == s->subsubframes[subframe] - 1) { ++ if(get_bits(&s->gb, 16) != 0xffff) { ++ av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: Didn't get subframe DSYNC\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ } ++ ++ /* advance sub-sub-frame index */ ++ if(++subsubframe >= s->subsubframes[subframe]) { ++ subsubframe = 0; ++ subframe++; ++ } ++ } ++ ++ /* skip to next channel set */ ++ i = get_bits_count(&s->gb); ++ if(start_posn + chset_fsize[chset] * 8 != i) { ++ j = start_posn + chset_fsize[chset] * 8 - i; ++ if(j < 0 || j >= 8) ++ av_log(s->avctx, AV_LOG_ERROR, "DTS-XBR: end of channel set," ++ " skipping further than expected (%d bits)\n", j); ++ skip_bits_long(&s->gb, j); ++ } ++ } ++ ++ return 0; ++} ++ ++/* parse initial header for XXCH and dump details */ ++static int dca_xxch_decode_frame(DCAContext *s) ++{ ++ int hdr_size, spkmsk_bits, num_chsets, core_spk, hdr_pos; ++ int i, chset, base_channel, chstart, fsize[8]; ++ ++ /* assume header word has already been parsed */ ++ hdr_pos = get_bits_count(&s->gb) - 32; ++ hdr_size = get_bits(&s->gb, 6) + 1; ++ /*chhdr_crc =*/ skip_bits1(&s->gb); ++ spkmsk_bits = get_bits(&s->gb, 5) + 1; ++ num_chsets = get_bits(&s->gb, 2) + 1; ++ ++ for (i = 0; i < num_chsets; i++) ++ fsize[i] = get_bits(&s->gb, 14) + 1; ++ ++ core_spk = get_bits(&s->gb, spkmsk_bits); ++ s->xxch_core_spkmask = core_spk; ++ s->xxch_nbits_spk_mask = spkmsk_bits; ++ s->xxch_dmix_embedded = 0; ++ ++ /* skip to the end of the header */ ++ i = get_bits_count(&s->gb); ++ if (hdr_pos + hdr_size * 8 > i) ++ skip_bits_long(&s->gb, hdr_pos + hdr_size * 8 - i); ++ ++ for (chset = 0; chset < num_chsets; chset++) { ++ chstart = get_bits_count(&s->gb); ++ base_channel = s->prim_channels; ++ s->xxch_chset = chset; ++ ++ /* XXCH and Core headers differ, see 6.4.2 "XXCH Channel Set Header" vs. ++ 5.3.2 "Primary Audio Coding Header", DTS Spec 1.3.1 */ ++ dca_parse_audio_coding_header(s, base_channel, 1); ++ ++ /* decode channel data */ ++ for (i = 0; i < (s->sample_blocks / 8); i++) { ++ if (dca_decode_block(s, base_channel, i)) { ++ av_log(s->avctx, AV_LOG_ERROR, ++ "Error decoding DTS-XXCH extension\n"); ++ continue; ++ } ++ } ++ ++ /* skip to end of this section */ ++ i = get_bits_count(&s->gb); ++ if (chstart + fsize[chset] * 8 > i) ++ skip_bits_long(&s->gb, chstart + fsize[chset] * 8 - i); ++ } ++ s->xxch_chset = num_chsets; ++ ++ return 0; ++} ++ ++/** ++ * Parse extension substream header (HD) ++ */ ++static void dca_exss_parse_header(DCAContext *s) ++{ ++ int asset_size[8]; ++ int ss_index; ++ int blownup; ++ int num_audiop = 1; ++ int num_assets = 1; ++ int active_ss_mask[8]; ++ int i, j; ++ int start_posn; ++ int hdrsize; ++ uint32_t mkr; ++ ++ if (get_bits_left(&s->gb) < 52) ++ return; ++ ++ start_posn = get_bits_count(&s->gb) - 32; ++ ++ skip_bits(&s->gb, 8); // user data ++ ss_index = get_bits(&s->gb, 2); ++ ++ blownup = get_bits1(&s->gb); ++ hdrsize = get_bits(&s->gb, 8 + 4 * blownup) + 1; // header_size ++ skip_bits(&s->gb, 16 + 4 * blownup); // hd_size ++ ++ s->static_fields = get_bits1(&s->gb); ++ if (s->static_fields) { ++ skip_bits(&s->gb, 2); // reference clock code ++ skip_bits(&s->gb, 3); // frame duration code ++ ++ if (get_bits1(&s->gb)) ++ skip_bits_long(&s->gb, 36); // timestamp ++ ++ /* a single stream can contain multiple audio assets that can be ++ * combined to form multiple audio presentations */ ++ ++ num_audiop = get_bits(&s->gb, 3) + 1; ++ if (num_audiop > 1) { ++ av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio presentations."); ++ /* ignore such streams for now */ ++ return; ++ } ++ ++ num_assets = get_bits(&s->gb, 3) + 1; ++ if (num_assets > 1) { ++ av_log_ask_for_sample(s->avctx, "Multiple DTS-HD audio assets."); ++ /* ignore such streams for now */ ++ return; ++ } ++ ++ for (i = 0; i < num_audiop; i++) ++ active_ss_mask[i] = get_bits(&s->gb, ss_index + 1); ++ ++ for (i = 0; i < num_audiop; i++) ++ for (j = 0; j <= ss_index; j++) ++ if (active_ss_mask[i] & (1 << j)) ++ skip_bits(&s->gb, 8); // active asset mask ++ ++ s->mix_metadata = get_bits1(&s->gb); ++ if (s->mix_metadata) { ++ int mix_out_mask_size; ++ ++ skip_bits(&s->gb, 2); // adjustment level ++ mix_out_mask_size = (get_bits(&s->gb, 2) + 1) << 2; ++ s->num_mix_configs = get_bits(&s->gb, 2) + 1; ++ ++ for (i = 0; i < s->num_mix_configs; i++) { ++ int mix_out_mask = get_bits(&s->gb, mix_out_mask_size); ++ s->mix_config_num_ch[i] = dca_exss_mask2count(mix_out_mask); ++ } ++ } ++ } ++ ++ for (i = 0; i < num_assets; i++) ++ asset_size[i] = get_bits_long(&s->gb, 16 + 4 * blownup); ++ ++ for (i = 0; i < num_assets; i++) { ++ if (dca_exss_parse_asset_header(s)) ++ return; ++ } ++ ++ /* not parsed further, we were only interested in the extensions mask ++ * from the asset header */ ++ ++ if (num_assets > 0) { ++ j = get_bits_count(&s->gb); ++ if (start_posn + hdrsize * 8 > j) ++ skip_bits_long(&s->gb, start_posn + hdrsize * 8 - j); ++ ++ for (i = 0; i < num_assets; i++) { ++ start_posn = get_bits_count(&s->gb); ++ mkr = get_bits_long(&s->gb, 32); ++ ++ /* parse extensions that we know about */ ++ if (mkr == 0x655e315e) { ++ dca_xbr_parse_frame(s); ++ } else if (mkr == 0x47004a03) { ++ dca_xxch_decode_frame(s); ++ s->core_ext_mask |= DCA_EXT_XXCH; /* xxx use for chan reordering */ ++ } else { ++ av_log(s->avctx, AV_LOG_DEBUG, ++ "DTS-ExSS: unknown marker = 0x%08x\n", mkr); ++ } ++ ++ /* skip to end of block */ ++ j = get_bits_count(&s->gb); ++ if (start_posn + asset_size[i] * 8 > j) ++ skip_bits_long(&s->gb, start_posn + asset_size[i] * 8 - j); ++ } ++ } ++} ++ ++/** ++ * Main frame decoding function ++ * FIXME add arguments ++ */ ++static int dca_decode_frame(AVCodecContext *avctx, void *data, ++ int *got_frame_ptr, AVPacket *avpkt) ++{ ++ AVFrame *frame = data; ++ const uint8_t *buf = avpkt->data; ++ int buf_size = avpkt->size; ++ int channel_mask; ++ int channel_layout; ++ int lfe_samples; ++ int num_core_channels = 0; ++ int i, ret; ++ float **samples_flt; ++ float *src_chan; ++ float *dst_chan; ++ DCAContext *s = avctx->priv_data; ++ int core_ss_end; ++ int channels, full_channels; ++ float scale; ++ int achan; ++ int chset; ++ int mask; ++ int lavc; ++ int posn; ++ int j, k; ++ int endch; ++ ++ s->xch_present = 0; ++ ++ s->dca_buffer_size = ff_dca_convert_bitstream(buf, buf_size, s->dca_buffer, ++ DCA_MAX_FRAME_SIZE + DCA_MAX_EXSS_HEADER_SIZE); ++ if (s->dca_buffer_size == AVERROR_INVALIDDATA) { ++ av_log(avctx, AV_LOG_ERROR, "Not a valid DCA frame\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ ++ init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8); ++ if ((ret = dca_parse_frame_header(s)) < 0) { ++ //seems like the frame is corrupt, try with the next one ++ return ret; ++ } ++ //set AVCodec values with parsed data ++ avctx->sample_rate = s->sample_rate; ++ avctx->bit_rate = s->bit_rate; ++ ++ s->profile = FF_PROFILE_DTS; ++ ++ for (i = 0; i < (s->sample_blocks / 8); i++) { ++ if ((ret = dca_decode_block(s, 0, i))) { ++ av_log(avctx, AV_LOG_ERROR, "error decoding block\n"); ++ return ret; ++ } ++ } ++ ++ /* record number of core channels incase less than max channels are requested */ ++ num_core_channels = s->prim_channels; ++ ++ if (s->ext_coding) ++ s->core_ext_mask = dca_ext_audio_descr_mask[s->ext_descr]; ++ else ++ s->core_ext_mask = 0; ++ ++ core_ss_end = FFMIN(s->frame_size, s->dca_buffer_size) * 8; ++ ++ /* only scan for extensions if ext_descr was unknown or indicated a ++ * supported XCh extension */ ++ if (s->core_ext_mask < 0 || s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) { ++ ++ /* if ext_descr was unknown, clear s->core_ext_mask so that the ++ * extensions scan can fill it up */ ++ s->core_ext_mask = FFMAX(s->core_ext_mask, 0); ++ ++ /* extensions start at 32-bit boundaries into bitstream */ ++ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); ++ ++ while (core_ss_end - get_bits_count(&s->gb) >= 32) { ++ uint32_t bits = get_bits_long(&s->gb, 32); ++ ++ switch (bits) { ++ case 0x5a5a5a5a: { ++ int ext_amode, xch_fsize; ++ ++ s->xch_base_channel = s->prim_channels; ++ ++ /* validate sync word using XCHFSIZE field */ ++ xch_fsize = show_bits(&s->gb, 10); ++ if ((s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize) && ++ (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + xch_fsize + 1)) ++ continue; ++ ++ /* skip length-to-end-of-frame field for the moment */ ++ skip_bits(&s->gb, 10); ++ ++ s->core_ext_mask |= DCA_EXT_XCH; ++ ++ /* extension amode(number of channels in extension) should be 1 */ ++ /* AFAIK XCh is not used for more channels */ ++ if ((ext_amode = get_bits(&s->gb, 4)) != 1) { ++ av_log(avctx, AV_LOG_ERROR, "XCh extension amode %d not" ++ " supported!\n", ext_amode); ++ continue; ++ } ++ ++ if (s->xch_base_channel < 2) { ++ av_log_ask_for_sample(avctx, "XCh with fewer than 2 base channels is not supported\n"); ++ continue; ++ } ++ ++ /* much like core primary audio coding header */ ++ dca_parse_audio_coding_header(s, s->xch_base_channel, 0); ++ ++ for (i = 0; i < (s->sample_blocks / 8); i++) ++ if ((ret = dca_decode_block(s, s->xch_base_channel, i))) { ++ av_log(avctx, AV_LOG_ERROR, "error decoding XCh extension\n"); ++ continue; ++ } ++ ++ s->xch_present = 1; ++ break; ++ } ++ case 0x47004a03: ++ /* XXCh: extended channels */ ++ /* usually found either in core or HD part in DTS-HD HRA streams, ++ * but not in DTS-ES which contains XCh extensions instead */ ++ s->core_ext_mask |= DCA_EXT_XXCH; ++ dca_xxch_decode_frame(s); ++ break; ++ ++ case 0x1d95f262: { ++ int fsize96 = show_bits(&s->gb, 12) + 1; ++ if (s->frame_size != (get_bits_count(&s->gb) >> 3) - 4 + fsize96) ++ continue; ++ ++ av_log(avctx, AV_LOG_DEBUG, "X96 extension found at %d bits\n", ++ get_bits_count(&s->gb)); ++ skip_bits(&s->gb, 12); ++ av_log(avctx, AV_LOG_DEBUG, "FSIZE96 = %d bytes\n", fsize96); ++ av_log(avctx, AV_LOG_DEBUG, "REVNO = %d\n", get_bits(&s->gb, 4)); ++ ++ s->core_ext_mask |= DCA_EXT_X96; ++ break; ++ } ++ } ++ ++ skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31); ++ } ++ } else { ++ /* no supported extensions, skip the rest of the core substream */ ++ skip_bits_long(&s->gb, core_ss_end - get_bits_count(&s->gb)); ++ } ++ ++ if (s->core_ext_mask & DCA_EXT_X96) ++ s->profile = FF_PROFILE_DTS_96_24; ++ else if (s->core_ext_mask & (DCA_EXT_XCH | DCA_EXT_XXCH)) ++ s->profile = FF_PROFILE_DTS_ES; ++ ++ /* check for ExSS (HD part) */ ++ if (s->dca_buffer_size - s->frame_size > 32 && ++ get_bits_long(&s->gb, 32) == DCA_HD_MARKER) ++ dca_exss_parse_header(s); ++ ++ avctx->profile = s->profile; ++ ++ full_channels = channels = s->prim_channels + !!s->lfe; ++ ++ /* If we have XXCH then the channel layout is managed differently */ ++ /* note that XLL will also have another way to do things */ ++ if (!(s->core_ext_mask & DCA_EXT_XXCH) ++ || (s->core_ext_mask & DCA_EXT_XXCH && avctx->request_channels > 0 ++ && avctx->request_channels ++ < num_core_channels + !!s->lfe + s->xxch_chset_nch[0])) ++ { /* xxx should also do MA extensions */ ++ if (s->amode < 16) { ++ avctx->channel_layout = dca_core_channel_layout[s->amode]; ++ ++ if (s->xch_present && (!avctx->request_channels || ++ avctx->request_channels ++ > num_core_channels + !!s->lfe)) { ++ avctx->channel_layout |= AV_CH_BACK_CENTER; ++ if (s->lfe) { ++ avctx->channel_layout |= AV_CH_LOW_FREQUENCY; ++ s->channel_order_tab = dca_channel_reorder_lfe_xch[s->amode]; ++ } else { ++ s->channel_order_tab = dca_channel_reorder_nolfe_xch[s->amode]; ++ } ++ if (s->channel_order_tab[s->xch_base_channel] < 0) ++ return AVERROR_INVALIDDATA; ++ } else { ++ channels = num_core_channels + !!s->lfe; ++ s->xch_present = 0; /* disable further xch processing */ ++ if (s->lfe) { ++ avctx->channel_layout |= AV_CH_LOW_FREQUENCY; ++ s->channel_order_tab = dca_channel_reorder_lfe[s->amode]; ++ } else ++ s->channel_order_tab = dca_channel_reorder_nolfe[s->amode]; ++ } ++ ++ if (channels > !!s->lfe && ++ s->channel_order_tab[channels - 1 - !!s->lfe] < 0) ++ return AVERROR_INVALIDDATA; ++ ++ if (av_get_channel_layout_nb_channels(avctx->channel_layout) != channels) { ++ av_log(avctx, AV_LOG_ERROR, "Number of channels %d mismatches layout %d\n", channels, av_get_channel_layout_nb_channels(avctx->channel_layout)); ++ return AVERROR_INVALIDDATA; ++ } ++ ++ if (avctx->request_channels == 2 && s->prim_channels > 2) { ++ channels = 2; ++ s->output = DCA_STEREO; ++ avctx->channel_layout = AV_CH_LAYOUT_STEREO; ++ } ++ else if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { ++ static const int8_t dca_channel_order_native[9] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 }; ++ s->channel_order_tab = dca_channel_order_native; ++ } ++ s->lfe_index = dca_lfe_index[s->amode]; ++ } else { ++ av_log(avctx, AV_LOG_ERROR, ++ "Non standard configuration %d !\n", s->amode); ++ return AVERROR_INVALIDDATA; ++ } ++ ++ s->xxch_dmix_embedded = 0; ++ } else { ++ /* we only get here if an XXCH channel set can be added to the mix */ ++ channel_mask = s->xxch_core_spkmask; ++ ++ if (avctx->request_channels > 0 ++ && avctx->request_channels < s->prim_channels) { ++ channels = num_core_channels + !!s->lfe; ++ for (i = 0; i < s->xxch_chset && channels + s->xxch_chset_nch[i] ++ <= avctx->request_channels; i++) { ++ channels += s->xxch_chset_nch[i]; ++ channel_mask |= s->xxch_spk_masks[i]; ++ } ++ } else { ++ channels = s->prim_channels + !!s->lfe; ++ for (i = 0; i < s->xxch_chset; i++) { ++ channel_mask |= s->xxch_spk_masks[i]; ++ } ++ } ++ ++ /* Given the DTS spec'ed channel mask, generate an avcodec version */ ++ channel_layout = 0; ++ for (i = 0; i < s->xxch_nbits_spk_mask; ++i) { ++ if (channel_mask & (1 << i)) { ++ channel_layout |= map_xxch_to_native[i]; ++ } ++ } ++ ++ /* make sure that we have managed to get equivelant dts/avcodec channel ++ * masks in some sense -- unfortunately some channels could overlap */ ++ if (av_popcount(channel_mask) != av_popcount(channel_layout)) { ++ av_log(avctx, AV_LOG_DEBUG, ++ "DTS-XXCH: Inconsistant avcodec/dts channel layouts\n"); ++ return AVERROR_INVALIDDATA; ++ } ++ ++ avctx->channel_layout = channel_layout; ++ ++ if (!(avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE)) { ++ /* Estimate DTS --> avcodec ordering table */ ++ for (chset = -1, j = 0; chset < s->xxch_chset; ++chset) { ++ mask = chset >= 0 ? s->xxch_spk_masks[chset] ++ : s->xxch_core_spkmask; ++ for (i = 0; i < s->xxch_nbits_spk_mask; i++) { ++ if (mask & ~(DCA_XXCH_LFE1 | DCA_XXCH_LFE2) & (1 << i)) { ++ lavc = map_xxch_to_native[i]; ++ posn = av_popcount(channel_layout & (lavc - 1)); ++ s->xxch_order_tab[j++] = posn; ++ } ++ } ++ } ++ ++ s->lfe_index = av_popcount(channel_layout & (AV_CH_LOW_FREQUENCY-1)); ++ } else { /* native ordering */ ++ for (i = 0; i < channels; i++) ++ s->xxch_order_tab[i] = i; ++ ++ s->lfe_index = channels - 1; ++ } ++ ++ s->channel_order_tab = s->xxch_order_tab; ++ } ++ ++ if (avctx->channels != channels) { ++ if (avctx->channels) ++ av_log(avctx, AV_LOG_INFO, "Number of channels changed in DCA decoder (%d -> %d)\n", avctx->channels, channels); ++ avctx->channels = channels; ++ } ++ ++ /* get output buffer */ ++ frame->nb_samples = 256 * (s->sample_blocks / 8); ++ if ((ret = ff_get_buffer(avctx, frame)) < 0) { ++ av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n"); ++ return ret; ++ } ++ samples_flt = (float **)frame->extended_data; ++ ++ /* allocate buffer for extra channels if downmixing */ ++ if (avctx->channels < full_channels) { ++ ret = av_samples_get_buffer_size(NULL, full_channels - channels, ++ frame->nb_samples, ++ avctx->sample_fmt, 0); ++ if (ret < 0) ++ return ret; ++ ++ av_fast_malloc(&s->extra_channels_buffer, ++ &s->extra_channels_buffer_size, ret); ++ if (!s->extra_channels_buffer) ++ return AVERROR(ENOMEM); ++ ++ ret = av_samples_fill_arrays((uint8_t **)s->extra_channels, NULL, ++ s->extra_channels_buffer, ++ full_channels - channels, ++ frame->nb_samples, avctx->sample_fmt, 0); ++ if (ret < 0) ++ return ret; ++ } ++ ++ /* filter to get final output */ ++ for (i = 0; i < (s->sample_blocks / 8); i++) { ++ int ch; ++ ++ for (ch = 0; ch < channels; ch++) ++ s->samples_chanptr[ch] = samples_flt[ch] + i * 256; ++ for (; ch < full_channels; ch++) ++ s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * 256; ++ ++ dca_filter_channels(s, i); ++ ++ /* If this was marked as a DTS-ES stream we need to subtract back- */ ++ /* channel from SL & SR to remove matrixed back-channel signal */ ++ if ((s->source_pcm_res & 1) && s->xch_present) { ++ float *back_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel]]; ++ float *lt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 2]]; ++ float *rt_chan = s->samples_chanptr[s->channel_order_tab[s->xch_base_channel - 1]]; ++ s->fdsp.vector_fmac_scalar(lt_chan, back_chan, -M_SQRT1_2, 256); ++ s->fdsp.vector_fmac_scalar(rt_chan, back_chan, -M_SQRT1_2, 256); ++ } ++ ++ /* If stream contains XXCH, we might need to undo an embedded downmix */ ++ if (s->xxch_dmix_embedded) { ++ /* Loop over channel sets in turn */ ++ ch = num_core_channels; ++ for (chset = 0; chset < s->xxch_chset; chset++) { ++ endch = ch + s->xxch_chset_nch[chset]; ++ mask = s->xxch_dmix_embedded; ++ ++ /* undo downmix */ ++ for (j = ch; j < endch; j++) { ++ if (mask & (1 << j)) { /* this channel has been mixed-out */ ++ src_chan = s->samples_chanptr[s->channel_order_tab[j]]; ++ for (k = 0; k < endch; k++) { ++ achan = s->channel_order_tab[k]; ++ scale = s->xxch_dmix_coeff[j][k]; ++ if (scale != 0.0) { ++ dst_chan = s->samples_chanptr[achan]; ++ s->fdsp.vector_fmac_scalar(dst_chan, src_chan, ++ -scale, 256); ++ } ++ } ++ } ++ } ++ ++ /* if a downmix has been embedded then undo the pre-scaling */ ++ if ((mask & (1 << ch)) && s->xxch_dmix_sf[chset] != 1.0f) { ++ scale = s->xxch_dmix_sf[chset]; ++ ++ for (j = 0; j < ch; j++) { ++ src_chan = s->samples_chanptr[s->channel_order_tab[j]]; ++ for (k = 0; k < 256; k++) ++ src_chan[k] *= scale; ++ } ++ ++ /* LFE channel is always part of core, scale if it exists */ ++ if (s->lfe) { ++ src_chan = s->samples_chanptr[s->lfe_index]; ++ for (k = 0; k < 256; k++) ++ src_chan[k] *= scale; ++ } ++ } ++ ++ ch = endch; ++ } ++ ++ } ++ } ++ ++ /* update lfe history */ ++ lfe_samples = 2 * s->lfe * (s->sample_blocks / 8); ++ for (i = 0; i < 2 * s->lfe * 4; i++) ++ s->lfe_data[i] = s->lfe_data[i + lfe_samples]; ++ ++ *got_frame_ptr = 1; ++ ++ return buf_size; ++} ++ ++ ++ ++/** ++ * DCA initialization ++ * ++ * @param avctx pointer to the AVCodecContext ++ */ ++ ++static av_cold int dca_decode_init(AVCodecContext *avctx) ++{ ++ DCAContext *s = avctx->priv_data; ++ ++ s->avctx = avctx; ++ dca_init_vlcs(); ++ ++ avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT); ++ ff_mdct_init(&s->imdct, 6, 1, 1.0); ++ ff_synth_filter_init(&s->synth); ++ ff_dcadsp_init(&s->dcadsp); ++ ff_fmt_convert_init(&s->fmt_conv, avctx); ++ ++ avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; ++ ++ /* allow downmixing to stereo */ ++ if (avctx->channels > 0 && avctx->request_channels < avctx->channels && ++ avctx->request_channels == 2) { ++ avctx->channels = avctx->request_channels; ++ } ++ ++ return 0; ++} ++ ++static av_cold int dca_decode_end(AVCodecContext *avctx) ++{ ++ DCAContext *s = avctx->priv_data; ++ ff_mdct_end(&s->imdct); ++ av_freep(&s->extra_channels_buffer); ++ return 0; ++} ++ ++static const AVProfile profiles[] = { ++ { FF_PROFILE_DTS, "DTS" }, ++ { FF_PROFILE_DTS_ES, "DTS-ES" }, ++ { FF_PROFILE_DTS_96_24, "DTS 96/24" }, ++ { FF_PROFILE_DTS_HD_HRA, "DTS-HD HRA" }, ++ { FF_PROFILE_DTS_HD_MA, "DTS-HD MA" }, ++ { FF_PROFILE_UNKNOWN }, ++}; ++ ++AVCodec ff_dca_decoder = { ++ .name = "dca", ++ .type = AVMEDIA_TYPE_AUDIO, ++ .id = AV_CODEC_ID_DTS, ++ .priv_data_size = sizeof(DCAContext), ++ .init = dca_decode_init, ++ .decode = dca_decode_frame, ++ .close = dca_decode_end, ++ .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"), ++ .capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, ++ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, ++ AV_SAMPLE_FMT_NONE }, ++ .profiles = NULL_IF_CONFIG_SMALL(profiles), ++}; +diff --git a/libavcodec/fmtconvert.c b/libavcodec/fmtconvert.c +index c03117c..fab92e2 100644 +--- a/libavcodec/fmtconvert.c ++++ b/libavcodec/fmtconvert.c +@@ -29,6 +29,12 @@ static void int32_to_float_fmul_scalar_c(float *dst, const int *src, float mul, + dst[i] = src[i] * mul; + } + ++static void int32_to_float_fmul_scalar_array_c(FmtConvertContext *c, float *dst, const int *src, float *mul, int len){ ++ int i; ++ for(i=0; iint32_to_float_fmul_scalar(dst, src, *mul++, 8); ++} ++ + static av_always_inline int float_to_int16_one(const float *src){ + return av_clip_int16(lrintf(*src)); + } +@@ -78,6 +84,7 @@ void ff_float_interleave_c(float *dst, const float **src, unsigned int len, + av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx) + { + c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c; ++ c->int32_to_float_fmul_scalar_array = int32_to_float_fmul_scalar_array_c; + c->float_to_int16 = float_to_int16_c; + c->float_to_int16_interleave = float_to_int16_interleave_c; + c->float_interleave = ff_float_interleave_c; +diff --git a/libavcodec/fmtconvert.h b/libavcodec/fmtconvert.h +index c058475..ce44615 100644 +--- a/libavcodec/fmtconvert.h ++++ b/libavcodec/fmtconvert.h +@@ -38,6 +38,20 @@ + void (*int32_to_float_fmul_scalar)(float *dst, const int *src, float mul, int len); + + /** ++ * Convert an array of int32_t to float and multiply by a float value from another array, ++ * stepping along the float array once for each 8 integers. ++ * @param c pointer to FmtConvertContext. ++ * @param dst destination array of float. ++ * constraints: 16-byte aligned ++ * @param src source array of int32_t. ++ * constraints: 16-byte aligned ++ * @param mul source array of float multipliers. ++ * @param len number of elements to convert. ++ * constraints: multiple of 8 ++ */ ++ void (*int32_to_float_fmul_scalar_array)(struct FmtConvertContext *c, float *dst, const int *src, float *mul, int len); ++ ++ /** + * Convert an array of float to an array of int16_t. + * + * Convert floats from in the range [-32768.0,32767.0] to ints +-- +1.8.1.6 + + +From a54cb0a8b8e3fe207dfd0652cf339d988329f591 Mon Sep 17 00:00:00 2001 +From: Ben Avison +Date: Thu, 27 Jun 2013 23:11:44 +0100 +Subject: [PATCH 53/55] Add VFP-accelerated version of imdct_half + +--- + libavcodec/arm/Makefile | 1 + + libavcodec/arm/fft_init_arm.c | 6 + + libavcodec/arm/mdct_vfp.S | 193 +++++++++++++++++++++++++++ + libavcodec/arm/synth_filter_vfp.S | 2 +- + 4 files changed, 201 insertions(+), 1 deletion(-) + create mode 100644 libavcodec/arm/mdct_vfp.S + +diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile +index b8b4df2..f6e683a 100644 +--- a/libavcodec/arm/Makefile ++++ b/libavcodec/arm/Makefile +@@ -37,6 +37,7 @@ OBJS-$(HAVE_ARMV6) += arm/dsputil_init_armv6.o \ + $(ARMV6-OBJS-yes) + + VFP-OBJS-$(HAVE_ARMV6) += arm/fmtconvert_vfp.o \ ++ arm/mdct_vfp.o \ + arm/synth_filter_vfp.o + + OBJS-$(HAVE_ARMVFP) += arm/dsputil_vfp.o \ +diff --git a/libavcodec/arm/fft_init_arm.c b/libavcodec/arm/fft_init_arm.c +index 605b3dd..baed753 100644 +--- a/libavcodec/arm/fft_init_arm.c ++++ b/libavcodec/arm/fft_init_arm.c +@@ -25,6 +25,8 @@ + void ff_fft_permute_neon(FFTContext *s, FFTComplex *z); + void ff_fft_calc_neon(FFTContext *s, FFTComplex *z); + ++void ff_imdct_half_vfp(FFTContext *s, FFTSample *output, const FFTSample *input); ++ + void ff_imdct_calc_neon(FFTContext *s, FFTSample *output, const FFTSample *input); + void ff_imdct_half_neon(FFTContext *s, FFTSample *output, const FFTSample *input); + void ff_mdct_calc_neon(FFTContext *s, FFTSample *output, const FFTSample *input); +@@ -46,6 +48,10 @@ void ff_synth_filter_float_neon(FFTContext *imdct, + av_cold void ff_fft_init_arm(FFTContext *s) + { + if (HAVE_NEON) { ++ if (HAVE_ARMVFP) { ++ s->imdct_half = ff_imdct_half_vfp; ++ } ++ + s->fft_permute = ff_fft_permute_neon; + s->fft_calc = ff_fft_calc_neon; + #if CONFIG_MDCT +diff --git a/libavcodec/arm/mdct_vfp.S b/libavcodec/arm/mdct_vfp.S +new file mode 100644 +index 0000000..2e2126e +--- /dev/null ++++ b/libavcodec/arm/mdct_vfp.S +@@ -0,0 +1,193 @@ ++/* ++ * Copyright (c) 2013 RISC OS Open Ltd ++ * ++ * 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 ++ * ++ * Author: Ben Avison ++ */ ++ ++#include "libavcodec/arm/asm.S" ++ ++CONTEXT .req a1 ++ORIGOUT .req a2 ++IN .req a3 ++OUT .req v1 ++REVTAB .req v2 ++TCOS .req v3 ++TSIN .req v4 ++OLDFPSCR .req v5 ++J0 .req a2 ++J1 .req a4 ++J2 .req ip ++J3 .req lr ++ ++.macro prerotation_innerloop ++ .set trig_lo, k ++ .set trig_hi, n4 - k - 2 ++ .set in_lo, trig_lo * 2 ++ .set in_hi, trig_hi * 2 ++ vldr d8, [TCOS, #trig_lo*4] @ s16,s17 ++ vldr d9, [TCOS, #trig_hi*4] @ s18,s19 ++ vldr s0, [IN, #in_hi*4 + 12] ++ vldr s1, [IN, #in_hi*4 + 4] ++ vldr s2, [IN, #in_lo*4 + 12] ++ vldr s3, [IN, #in_lo*4 + 4] ++ vmul.f s8, s0, s16 @ vector operation ++ vldr d10, [TSIN, #trig_lo*4] @ s20,s21 ++ vldr d11, [TSIN, #trig_hi*4] @ s22,s23 ++ vldr s4, [IN, #in_lo*4] ++ vldr s5, [IN, #in_lo*4 + 8] ++ vldr s6, [IN, #in_hi*4] ++ vldr s7, [IN, #in_hi*4 + 8] ++ ldr J0, [REVTAB, #trig_lo*2] ++ vmul.f s12, s0, s20 @ vector operation ++ ldr J2, [REVTAB, #trig_hi*2] ++ mov J1, J0, lsr #16 ++ and J0, J0, #255 @ halfword value will be < n4 ++ vmls.f s8, s4, s20 @ vector operation ++ mov J3, J2, lsr #16 ++ and J2, J2, #255 @ halfword value will be < n4 ++ add J0, OUT, J0, lsl #3 ++ vmla.f s12, s4, s16 @ vector operation ++ add J1, OUT, J1, lsl #3 ++ add J2, OUT, J2, lsl #3 ++ add J3, OUT, J3, lsl #3 ++ vstr s8, [J0] ++ vstr s9, [J1] ++ vstr s10, [J2] ++ vstr s11, [J3] ++ vstr s12, [J0, #4] ++ vstr s13, [J1, #4] ++ vstr s14, [J2, #4] ++ vstr s15, [J3, #4] ++ .set k, k + 2 ++.endm ++ ++.macro postrotation_innerloop tail, head ++ .set trig_lo_head, n8 - k - 2 ++ .set trig_hi_head, n8 + k ++ .set out_lo_head, trig_lo_head * 2 ++ .set out_hi_head, trig_hi_head * 2 ++ .set trig_lo_tail, n8 - (k - 2) - 2 ++ .set trig_hi_tail, n8 + (k - 2) ++ .set out_lo_tail, trig_lo_tail * 2 ++ .set out_hi_tail, trig_hi_tail * 2 ++ .if (k & 2) == 0 ++ TCOS_D0_HEAD .req d10 @ s20,s21 ++ TCOS_D1_HEAD .req d11 @ s22,s23 ++ TCOS_S0_TAIL .req s24 ++ .else ++ TCOS_D0_HEAD .req d12 @ s24,s25 ++ TCOS_D1_HEAD .req d13 @ s26,s27 ++ TCOS_S0_TAIL .req s20 ++ .endif ++ .ifnc "\tail","" ++ vmls.f s8, s0, TCOS_S0_TAIL @ vector operation ++ .endif ++ .ifnc "\head","" ++ vldr d8, [TSIN, #trig_lo_head*4] @ s16,s17 ++ vldr d9, [TSIN, #trig_hi_head*4] @ s18,s19 ++ vldr TCOS_D0_HEAD, [TCOS, #trig_lo_head*4] ++ .endif ++ .ifnc "\tail","" ++ vmla.f s12, s4, TCOS_S0_TAIL @ vector operation ++ .endif ++ .ifnc "\head","" ++ vldr s0, [OUT, #out_lo_head*4] ++ vldr s1, [OUT, #out_lo_head*4 + 8] ++ vldr s2, [OUT, #out_hi_head*4] ++ vldr s3, [OUT, #out_hi_head*4 + 8] ++ vldr s4, [OUT, #out_lo_head*4 + 4] ++ vldr s5, [OUT, #out_lo_head*4 + 12] ++ vldr s6, [OUT, #out_hi_head*4 + 4] ++ vldr s7, [OUT, #out_hi_head*4 + 12] ++ .endif ++ .ifnc "\tail","" ++ vstr s8, [OUT, #out_lo_tail*4] ++ vstr s9, [OUT, #out_lo_tail*4 + 8] ++ vstr s10, [OUT, #out_hi_tail*4] ++ vstr s11, [OUT, #out_hi_tail*4 + 8] ++ .endif ++ .ifnc "\head","" ++ vmul.f s8, s4, s16 @ vector operation ++ .endif ++ .ifnc "\tail","" ++ vstr s12, [OUT, #out_hi_tail*4 + 12] ++ vstr s13, [OUT, #out_hi_tail*4 + 4] ++ vstr s14, [OUT, #out_lo_tail*4 + 12] ++ vstr s15, [OUT, #out_lo_tail*4 + 4] ++ .endif ++ .ifnc "\head","" ++ vmul.f s12, s0, s16 @ vector operation ++ vldr TCOS_D1_HEAD, [TCOS, #trig_hi_head*4] ++ .endif ++ .unreq TCOS_D0_HEAD ++ .unreq TCOS_D1_HEAD ++ .unreq TCOS_S0_TAIL ++ .ifnc "\head","" ++ .set k, k + 2 ++ .endif ++.endm ++ ++ ++/* void ff_imdct_half_vfp(FFTContext *s, ++ * FFTSample *output, ++ * const FFTSample *input) ++ */ ++function ff_imdct_half_vfp, export=1 ++ ldr ip, [CONTEXT, #5*4] @ mdct_bits ++ teq ip, #6 ++ bne ff_imdct_half_c @ only case currently accelerated is the one used by DCA ++ ++ .set n, 1<<6 ++ .set n2, n/2 ++ .set n4, n/4 ++ .set n8, n/8 ++ ++ push {v1-v5,lr} ++ vpush {s16-s27} ++ vmrs OLDFPSCR, FPSCR ++ ldr lr, =0x03030000 @ RunFast mode, short vectors of length 4, stride 1 ++ vmsr FPSCR, lr ++ mov OUT, ORIGOUT ++ ldr REVTAB, [CONTEXT, #2*4] ++ ldr TCOS, [CONTEXT, #6*4] ++ ldr TSIN, [CONTEXT, #7*4] ++ ++ .set k, 0 ++ .rept n8/2 ++ prerotation_innerloop ++ .endr ++ ++ vmsr FPSCR, OLDFPSCR ++ mov ORIGOUT, OUT ++ ldr ip, [CONTEXT, #9*4] ++ blx ip @ s->fft_calc(s, output) ++ ldr lr, =0x03030000 @ RunFast mode, short vectors of length 4, stride 1 ++ vmsr FPSCR, lr ++ ++ .set k, 0 ++ postrotation_innerloop , head ++ .rept n8/2 - 1 ++ postrotation_innerloop tail, head ++ .endr ++ postrotation_innerloop tail ++ ++ vmsr FPSCR, OLDFPSCR ++ vpop {s16-s27} ++ pop {v1-v5,pc} ++endfunc +diff --git a/libavcodec/arm/synth_filter_vfp.S b/libavcodec/arm/synth_filter_vfp.S +index 8c54267..40bf2c4 100644 +--- a/libavcodec/arm/synth_filter_vfp.S ++++ b/libavcodec/arm/synth_filter_vfp.S +@@ -133,7 +133,7 @@ function ff_synth_filter_float_vfp, export=1 + str lr, [P_SB_OFF] @ rotate offset, modulo buffer size, ready for next call + ldr a3, [sp, #(16+6+2)*4] @ fetch in from stack, to pass to imdct_half + VFP vmov s16, SCALE @ imdct_half is free to corrupt s0, but it contains one of our arguments in hardfp case +- bl ff_imdct_half_c ++ bl ff_imdct_half_vfp + VFP vmov SCALE, s16 + + vmrs OLDFPSCR, FPSCR +-- +1.8.1.6 + + +From 872db4f60fd2314cab83eeddf72a46a53708bf62 Mon Sep 17 00:00:00 2001 +From: Ben Avison +Date: Fri, 28 Jun 2013 21:21:06 +0100 +Subject: [PATCH 54/55] Add VFP_accelerated version of dca_lfe_fir + +--- + libavcodec/arm/Makefile | 3 +- + libavcodec/arm/dcadsp_init_arm.c | 4 + + libavcodec/arm/dcadsp_vfp.S | 189 ++++++++++++++++++++++++++++ + 3 files changed, 195 insertions(+), 1 deletion(-) + create mode 100644 libavcodec/arm/dcadsp_vfp.S + +diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile +index f6e683a..ad1ae07 100644 +--- a/libavcodec/arm/Makefile ++++ b/libavcodec/arm/Makefile +@@ -36,7 +36,8 @@ OBJS-$(HAVE_ARMV6) += arm/dsputil_init_armv6.o \ + arm/simple_idct_armv6.o \ + $(ARMV6-OBJS-yes) + +-VFP-OBJS-$(HAVE_ARMV6) += arm/fmtconvert_vfp.o \ ++VFP-OBJS-$(HAVE_ARMV6) += arm/dcadsp_vfp.o \ ++ arm/fmtconvert_vfp.o \ + arm/mdct_vfp.o \ + arm/synth_filter_vfp.o + +diff --git a/libavcodec/arm/dcadsp_init_arm.c b/libavcodec/arm/dcadsp_init_arm.c +index 5663cd7..d62fc75 100644 +--- a/libavcodec/arm/dcadsp_init_arm.c ++++ b/libavcodec/arm/dcadsp_init_arm.c +@@ -22,11 +22,15 @@ + #include "libavutil/attributes.h" + #include "libavcodec/dcadsp.h" + ++void ff_dca_lfe_fir_vfp(float *out, const float *in, const float *coefs, ++ int decifactor, float scale); + void ff_dca_lfe_fir_neon(float *out, const float *in, const float *coefs, + int decifactor, float scale); + + void av_cold ff_dcadsp_init_arm(DCADSPContext *s) + { ++ if (HAVE_ARMVFP) ++ s->lfe_fir = ff_dca_lfe_fir_vfp; + if (HAVE_NEON) + s->lfe_fir = ff_dca_lfe_fir_neon; + } +diff --git a/libavcodec/arm/dcadsp_vfp.S b/libavcodec/arm/dcadsp_vfp.S +new file mode 100644 +index 0000000..606e4af +--- /dev/null ++++ b/libavcodec/arm/dcadsp_vfp.S +@@ -0,0 +1,189 @@ ++/* ++ * Copyright (c) 2013 RISC OS Open Ltd ++ * ++ * 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 ++ * ++ * Author: Ben Avison ++ */ ++ ++#include "libavcodec/arm/asm.S" ++ ++POUT .req a1 ++PIN .req a2 ++PCOEF .req a3 ++DECIFACTOR .req a4 ++OLDFPSCR .req a4 ++COUNTER .req ip ++ ++SCALE32 .req s28 @ use vector of 4 in place of 9th scalar when decifactor=32 / JMAX=8 ++SCALE64 .req s0 @ spare register in scalar bank when decifactor=64 / JMAX=4 ++IN0 .req s4 ++IN1 .req s5 ++IN2 .req s6 ++IN3 .req s7 ++IN4 .req s0 ++IN5 .req s1 ++IN6 .req s2 ++IN7 .req s3 ++COEF0 .req s8 @ coefficient elements ++COEF1 .req s9 ++COEF2 .req s10 ++COEF3 .req s11 ++COEF4 .req s12 ++COEF5 .req s13 ++COEF6 .req s14 ++COEF7 .req s15 ++ACCUM0 .req s16 @ double-buffered multiply-accumulate results ++ACCUM4 .req s20 ++POST0 .req s24 @ do long-latency post-multiply in this vector in parallel ++POST1 .req s25 ++POST2 .req s26 ++POST3 .req s27 ++ ++ ++.macro inner_loop decifactor, dir, tail, head ++ .ifc "\dir","up" ++ .set X, 0 ++ .set Y, 4 ++ .else ++ .set X, 4*JMAX*4 - 4 ++ .set Y, -4 ++ .endif ++ .ifnc "\head","" ++ vldr COEF0, [PCOEF, #X + (0*JMAX + 0) * Y] ++ vldr COEF1, [PCOEF, #X + (1*JMAX + 0) * Y] ++ vldr COEF2, [PCOEF, #X + (2*JMAX + 0) * Y] ++ vldr COEF3, [PCOEF, #X + (3*JMAX + 0) * Y] ++ .endif ++ .ifnc "\tail","" ++ vadd.f POST0, ACCUM0, ACCUM4 @ vector operation ++ .endif ++ .ifnc "\head","" ++ vmul.f ACCUM0, COEF0, IN0 @ vector = vector * scalar ++ vldr COEF4, [PCOEF, #X + (0*JMAX + 1) * Y] ++ vldr COEF5, [PCOEF, #X + (1*JMAX + 1) * Y] ++ vldr COEF6, [PCOEF, #X + (2*JMAX + 1) * Y] ++ .endif ++ .ifnc "\tail","" ++ vmul.f POST0, POST0, SCALE\decifactor @ vector operation (SCALE may be scalar) ++ .endif ++ .ifnc "\head","" ++ vldr COEF7, [PCOEF, #X + (3*JMAX + 1) * Y] ++ .ifc "\tail","" ++ vmul.f ACCUM4, COEF4, IN1 @ vector operation ++ .endif ++ vldr COEF0, [PCOEF, #X + (0*JMAX + 2) * Y] ++ vldr COEF1, [PCOEF, #X + (1*JMAX + 2) * Y] ++ .ifnc "\tail","" ++ vmul.f ACCUM4, COEF4, IN1 @ vector operation ++ .endif ++ vldr COEF2, [PCOEF, #X + (2*JMAX + 2) * Y] ++ vldr COEF3, [PCOEF, #X + (3*JMAX + 2) * Y] ++ .endif ++ .ifnc "\tail","" ++ vstmia POUT!, {POST0-POST3} ++ .endif ++ .ifnc "\head","" ++ vmla.f ACCUM0, COEF0, IN2 @ vector = vector * scalar ++ vldr COEF4, [PCOEF, #X + (0*JMAX + 3) * Y] ++ vldr COEF5, [PCOEF, #X + (1*JMAX + 3) * Y] ++ vldr COEF6, [PCOEF, #X + (2*JMAX + 3) * Y] ++ vldr COEF7, [PCOEF, #X + (3*JMAX + 3) * Y] ++ vmla.f ACCUM4, COEF4, IN3 @ vector = vector * scalar ++ .if \decifactor == 32 ++ vldr COEF0, [PCOEF, #X + (0*JMAX + 4) * Y] ++ vldr COEF1, [PCOEF, #X + (1*JMAX + 4) * Y] ++ vldr COEF2, [PCOEF, #X + (2*JMAX + 4) * Y] ++ vldr COEF3, [PCOEF, #X + (3*JMAX + 4) * Y] ++ vmla.f ACCUM0, COEF0, IN4 @ vector = vector * scalar ++ vldr COEF4, [PCOEF, #X + (0*JMAX + 5) * Y] ++ vldr COEF5, [PCOEF, #X + (1*JMAX + 5) * Y] ++ vldr COEF6, [PCOEF, #X + (2*JMAX + 5) * Y] ++ vldr COEF7, [PCOEF, #X + (3*JMAX + 5) * Y] ++ vmla.f ACCUM4, COEF4, IN5 @ vector = vector * scalar ++ vldr COEF0, [PCOEF, #X + (0*JMAX + 6) * Y] ++ vldr COEF1, [PCOEF, #X + (1*JMAX + 6) * Y] ++ vldr COEF2, [PCOEF, #X + (2*JMAX + 6) * Y] ++ vldr COEF3, [PCOEF, #X + (3*JMAX + 6) * Y] ++ vmla.f ACCUM0, COEF0, IN6 @ vector = vector * scalar ++ vldr COEF4, [PCOEF, #X + (0*JMAX + 7) * Y] ++ vldr COEF5, [PCOEF, #X + (1*JMAX + 7) * Y] ++ vldr COEF6, [PCOEF, #X + (2*JMAX + 7) * Y] ++ vldr COEF7, [PCOEF, #X + (3*JMAX + 7) * Y] ++ vmla.f ACCUM4, COEF4, IN7 @ vector = vector * scalar ++ .endif ++ .endif ++.endm ++ ++.macro dca_lfe_fir decifactor ++ .if \decifactor == 32 ++ .set JMAX, 8 ++ vpush {s16-s31} ++ vmov SCALE32, s0 @ duplicate scalar across vector ++ vldr IN4, [PIN, #-4*4] ++ vldr IN5, [PIN, #-5*4] ++ vldr IN6, [PIN, #-6*4] ++ vldr IN7, [PIN, #-7*4] ++ .else ++ .set JMAX, 4 ++ vpush {s16-s27} ++ .endif ++ ++ mov COUNTER, #\decifactor/4 - 1 ++ inner_loop \decifactor, up,, head ++1: add PCOEF, PCOEF, #4*JMAX*4 ++ subs COUNTER, COUNTER, #1 ++ inner_loop \decifactor, up, tail, head ++ bne 1b ++ inner_loop \decifactor, up, tail ++ ++ mov COUNTER, #\decifactor/4 - 1 ++ inner_loop \decifactor, down,, head ++1: sub PCOEF, PCOEF, #4*JMAX*4 ++ subs COUNTER, COUNTER, #1 ++ inner_loop \decifactor, down, tail, head ++ bne 1b ++ inner_loop \decifactor, down, tail ++ ++ .if \decifactor == 32 ++ vpop {s16-s31} ++ .else ++ vpop {s16-s27} ++ .endif ++ vmsr FPSCR, OLDFPSCR ++ bx lr ++.endm ++ ++ ++/* void ff_dca_lfe_fir_vfp(float *out, const float *in, const float *coefs, ++ * int decifactor, float scale) ++ */ ++function ff_dca_lfe_fir_vfp, export=1 ++ teq DECIFACTOR, #32 ++ vmrs OLDFPSCR, FPSCR ++ ldr ip, =0x03030000 @ RunFast mode, short vectors of length 4, stride 1 ++ vmsr FPSCR, ip ++NOVFP vldr s0, [sp] ++ vldr IN0, [PIN, #-0*4] ++ vldr IN1, [PIN, #-1*4] ++ vldr IN2, [PIN, #-2*4] ++ vldr IN3, [PIN, #-3*4] ++ beq 32f ++64: dca_lfe_fir 64 ++ .ltorg ++32: dca_lfe_fir 32 ++endfunc +-- +1.8.1.6 + + +From 59230eeb1c1ac9bbb3089fd86172d5fafbdebe94 Mon Sep 17 00:00:00 2001 +From: Ben Avison +Date: Tue, 9 Jul 2013 17:44:50 +0100 +Subject: [PATCH 55/55] Add VFP-accelerated version of fft16 + +--- + libavcodec/arm/Makefile | 1 + + libavcodec/arm/fft_vfp.S | 299 +++++++++++++++++++++++++++++++++++ + libavcodec/arm/mdct_vfp.S | 5 +- + 3 files changed, 302 insertions(+), 3 deletions(-) + create mode 100644 libavcodec/arm/fft_vfp.S + +diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile +index ad1ae07..798dfe3 100644 +--- a/libavcodec/arm/Makefile ++++ b/libavcodec/arm/Makefile +@@ -37,6 +37,7 @@ OBJS-$(HAVE_ARMV6) += arm/dsputil_init_armv6.o \ + $(ARMV6-OBJS-yes) + + VFP-OBJS-$(HAVE_ARMV6) += arm/dcadsp_vfp.o \ ++ arm/fft_vfp.o \ + arm/fmtconvert_vfp.o \ + arm/mdct_vfp.o \ + arm/synth_filter_vfp.o +diff --git a/libavcodec/arm/fft_vfp.S b/libavcodec/arm/fft_vfp.S +new file mode 100644 +index 0000000..a7106ea +--- /dev/null ++++ b/libavcodec/arm/fft_vfp.S +@@ -0,0 +1,299 @@ ++/* ++ * Copyright (c) 2013 RISC OS Open Ltd ++ * ++ * 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 ++ * ++ * Author: Ben Avison ++ */ ++ ++#include "libavcodec/arm/asm.S" ++ ++@ TODO: * FFTs wider than 16 ++@ * dispatch code ++ ++function fft4_vfp ++ vldr d0, [a1, #0*2*4] @ s0,s1 = z[0] ++ vldr d4, [a1, #1*2*4] @ s8,s9 = z[1] ++ vldr d1, [a1, #2*2*4] @ s2,s3 = z[2] ++ vldr d5, [a1, #3*2*4] @ s10,s11 = z[3] ++ @ stall ++ vadd.f s12, s0, s8 @ i0 ++ vadd.f s13, s1, s9 @ i1 ++ vadd.f s14, s2, s10 @ i2 ++ vadd.f s15, s3, s11 @ i3 ++ vsub.f s8, s0, s8 @ i4 ++ vsub.f s9, s1, s9 @ i5 ++ vsub.f s10, s2, s10 @ i6 ++ vsub.f s11, s3, s11 @ i7 ++ @ stall ++ @ stall ++ vadd.f s0, s12, s14 @ z[0].re ++ vsub.f s4, s12, s14 @ z[2].re ++ vadd.f s1, s13, s15 @ z[0].im ++ vsub.f s5, s13, s15 @ z[2].im ++ vadd.f s7, s9, s10 @ z[3].im ++ vsub.f s3, s9, s10 @ z[1].im ++ vadd.f s2, s8, s11 @ z[1].re ++ vsub.f s6, s8, s11 @ z[3].re ++ @ stall ++ @ stall ++ vstr d0, [a1, #0*2*4] ++ vstr d2, [a1, #2*2*4] ++ @ stall ++ @ stall ++ vstr d1, [a1, #1*2*4] ++ vstr d3, [a1, #3*2*4] ++ ++ bx lr ++endfunc ++ ++.macro macro_fft8_head ++ @ FFT4 ++ vldr d4, [a1, #0 * 2*4] ++ vldr d6, [a1, #1 * 2*4] ++ vldr d5, [a1, #2 * 2*4] ++ vldr d7, [a1, #3 * 2*4] ++ @ BF ++ vldr d12, [a1, #4 * 2*4] ++ vadd.f s16, s8, s12 @ vector op ++ vldr d14, [a1, #5 * 2*4] ++ vldr d13, [a1, #6 * 2*4] ++ vldr d15, [a1, #7 * 2*4] ++ vsub.f s20, s8, s12 @ vector op ++ vadd.f s0, s16, s18 ++ vsub.f s2, s16, s18 ++ vadd.f s1, s17, s19 ++ vsub.f s3, s17, s19 ++ vadd.f s7, s21, s22 ++ vsub.f s5, s21, s22 ++ vadd.f s4, s20, s23 ++ vsub.f s6, s20, s23 ++ vsub.f s20, s24, s28 @ vector op ++ vstr d0, [a1, #0 * 2*4] @ transfer s0-s7 to s24-s31 via memory ++ vstr d1, [a1, #1 * 2*4] ++ vldr s0, cos1pi4 ++ vadd.f s16, s24, s28 @ vector op ++ vstr d2, [a1, #2 * 2*4] ++ vstr d3, [a1, #3 * 2*4] ++ vldr d12, [a1, #0 * 2*4] ++ @ TRANSFORM ++ vmul.f s20, s20, s0 @ vector x scalar op ++ vldr d13, [a1, #1 * 2*4] ++ vldr d14, [a1, #2 * 2*4] ++ vldr d15, [a1, #3 * 2*4] ++ @ BUTTERFLIES ++ vadd.f s0, s18, s16 ++ vadd.f s1, s17, s19 ++ vsub.f s2, s17, s19 ++ vsub.f s3, s18, s16 ++ vadd.f s4, s21, s20 ++ vsub.f s5, s21, s20 ++ vadd.f s6, s22, s23 ++ vsub.f s7, s22, s23 ++ vadd.f s8, s0, s24 @ vector op ++ vstr d0, [a1, #0 * 2*4] @ transfer s0-s3 to s12-s15 via memory ++ vstr d1, [a1, #1 * 2*4] ++ vldr d6, [a1, #0 * 2*4] ++ vldr d7, [a1, #1 * 2*4] ++ vadd.f s1, s5, s6 ++ vadd.f s0, s7, s4 ++ vsub.f s2, s5, s6 ++ vsub.f s3, s7, s4 ++ vsub.f s12, s24, s12 @ vector op ++ vsub.f s5, s29, s1 ++ vsub.f s4, s28, s0 ++ vsub.f s6, s30, s2 ++ vsub.f s7, s31, s3 ++ vadd.f s16, s0, s28 @ vector op ++ vstr d6, [a1, #4 * 2*4] ++ vstr d7, [a1, #6 * 2*4] ++ vstr d4, [a1, #0 * 2*4] ++ vstr d5, [a1, #2 * 2*4] ++ vstr d2, [a1, #5 * 2*4] ++ vstr d3, [a1, #7 * 2*4] ++.endm ++ ++.macro macro_fft8_tail ++ vstr d8, [a1, #1 * 2*4] ++ vstr d9, [a1, #3 * 2*4] ++.endm ++ ++function fft8_vfp ++ ldr a3, =0x03030000 @ RunFast mode, vector length 4, stride 1 ++ vmrs a2, FPSCR ++ vmsr FPSCR, a3 ++ vpush {s16-s31} ++ ++ macro_fft8_head ++ macro_fft8_tail ++ ++ vpop {s16-s31} ++ vmsr FPSCR, a2 ++ bx lr ++endfunc ++ ++.align 3 ++cos1pi4: @ cos(1*pi/4) = sqrt(2) ++ .float 0.707106769084930419921875 ++cos1pi8: @ cos(1*pi/8) = sqrt(2+sqrt(2))/2 ++ .float 0.92387950420379638671875 ++cos3pi8: @ cos(2*pi/8) = sqrt(2-sqrt(2))/2 ++ .float 0.3826834261417388916015625 ++ ++function ff_fft16_vfp, export=1 ++ ldr a3, =0x03030000 @ RunFast mode, vector length 4, stride 1 ++ vmrs a2, FPSCR ++ vmsr FPSCR, a3 ++ vpush {s16-s31} ++ ++ macro_fft8_head ++ @ FFT4(z+8) ++ vldr d10, [a1, #8 * 2*4] ++ vldr d12, [a1, #9 * 2*4] ++ vldr d11, [a1, #10 * 2*4] ++ vldr d13, [a1, #11 * 2*4] ++ macro_fft8_tail ++ vadd.f s16, s20, s24 @ vector op ++ @ FFT4(z+12) ++ vldr d4, [a1, #12 * 2*4] ++ vldr d6, [a1, #13 * 2*4] ++ vldr d5, [a1, #14 * 2*4] ++ vsub.f s20, s20, s24 @ vector op ++ vldr d7, [a1, #15 * 2*4] ++ vadd.f s0, s16, s18 ++ vsub.f s4, s16, s18 ++ vadd.f s1, s17, s19 ++ vsub.f s5, s17, s19 ++ vadd.f s7, s21, s22 ++ vsub.f s3, s21, s22 ++ vadd.f s2, s20, s23 ++ vsub.f s6, s20, s23 ++ vadd.f s16, s8, s12 @ vector op ++ vstr d0, [a1, #8 * 2*4] ++ vstr d2, [a1, #10 * 2*4] ++ vstr d1, [a1, #9 * 2*4] ++ vsub.f s20, s8, s12 ++ vstr d3, [a1, #11 * 2*4] ++ @ TRANSFORM(z[2],z[6],z[10],z[14],cos1pi4,cos1pi4) ++ vldr d12, [a1, #10 * 2*4] ++ vadd.f s0, s16, s18 ++ vadd.f s1, s17, s19 ++ vsub.f s6, s16, s18 ++ vsub.f s7, s17, s19 ++ vsub.f s3, s21, s22 ++ vadd.f s2, s20, s23 ++ vadd.f s5, s21, s22 ++ vsub.f s4, s20, s23 ++ vstr d0, [a1, #12 * 2*4] ++ vmov s0, s6 ++ @ TRANSFORM(z[1],z[5],z[9],z[13],cos1pi8,cos3pi8) ++ vldr d6, [a1, #9 * 2*4] ++ vstr d1, [a1, #13 * 2*4] ++ vldr d1, cos1pi4 @ s2 = cos1pi4, s3 = cos1pi8 ++ vstr d2, [a1, #15 * 2*4] ++ vldr d7, [a1, #13 * 2*4] ++ vadd.f s4, s25, s24 ++ vsub.f s5, s25, s24 ++ vsub.f s6, s0, s7 ++ vadd.f s7, s0, s7 ++ vmul.f s20, s12, s3 @ vector op ++ @ TRANSFORM(z[3],z[7],z[11],z[15],cos3pi8,cos1pi8) ++ vldr d4, [a1, #11 * 2*4] ++ vldr d5, [a1, #15 * 2*4] ++ vldr s1, cos3pi8 ++ vmul.f s24, s4, s2 @ vector * scalar op ++ vmul.f s28, s12, s1 @ vector * scalar op ++ vmul.f s12, s8, s1 @ vector * scalar op ++ vadd.f s4, s20, s29 ++ vsub.f s5, s21, s28 ++ vsub.f s6, s22, s31 ++ vadd.f s7, s23, s30 ++ vmul.f s8, s8, s3 @ vector * scalar op ++ vldr d8, [a1, #1 * 2*4] ++ vldr d9, [a1, #5 * 2*4] ++ vldr d10, [a1, #3 * 2*4] ++ vldr d11, [a1, #7 * 2*4] ++ vldr d14, [a1, #2 * 2*4] ++ vadd.f s0, s6, s4 ++ vadd.f s1, s5, s7 ++ vsub.f s2, s5, s7 ++ vsub.f s3, s6, s4 ++ vadd.f s4, s12, s9 ++ vsub.f s5, s13, s8 ++ vsub.f s6, s14, s11 ++ vadd.f s7, s15, s10 ++ vadd.f s12, s0, s16 @ vector op ++ vstr d0, [a1, #1 * 2*4] ++ vstr d1, [a1, #5 * 2*4] ++ vldr d4, [a1, #1 * 2*4] ++ vldr d5, [a1, #5 * 2*4] ++ vadd.f s0, s6, s4 ++ vadd.f s1, s5, s7 ++ vsub.f s2, s5, s7 ++ vsub.f s3, s6, s4 ++ vsub.f s8, s16, s8 @ vector op ++ vstr d6, [a1, #1 * 2*4] ++ vstr d7, [a1, #5 * 2*4] ++ vldr d15, [a1, #6 * 2*4] ++ vsub.f s4, s20, s0 ++ vsub.f s5, s21, s1 ++ vsub.f s6, s22, s2 ++ vsub.f s7, s23, s3 ++ vadd.f s20, s0, s20 @ vector op ++ vstr d4, [a1, #9 * 2*4] ++ @ TRANSFORM_ZERO(z[0],z[4],z[8],z[12]) ++ vldr d6, [a1, #8 * 2*4] ++ vstr d5, [a1, #13 * 2*4] ++ vldr d7, [a1, #12 * 2*4] ++ vstr d2, [a1, #11 * 2*4] ++ vldr d8, [a1, #0 * 2*4] ++ vstr d3, [a1, #15 * 2*4] ++ vldr d9, [a1, #4 * 2*4] ++ vadd.f s0, s26, s24 ++ vadd.f s1, s25, s27 ++ vsub.f s2, s25, s27 ++ vsub.f s3, s26, s24 ++ vadd.f s4, s14, s12 ++ vadd.f s5, s13, s15 ++ vsub.f s6, s13, s15 ++ vsub.f s7, s14, s12 ++ vadd.f s8, s0, s28 @ vector op ++ vstr d0, [a1, #3 * 2*4] ++ vstr d1, [a1, #7 * 2*4] ++ vldr d6, [a1, #3 * 2*4] ++ vldr d7, [a1, #7 * 2*4] ++ vsub.f s0, s16, s4 ++ vsub.f s1, s17, s5 ++ vsub.f s2, s18, s6 ++ vsub.f s3, s19, s7 ++ vsub.f s12, s28, s12 @ vector op ++ vadd.f s16, s4, s16 @ vector op ++ vstr d10, [a1, #3 * 2*4] ++ vstr d11, [a1, #7 * 2*4] ++ vstr d4, [a1, #2 * 2*4] ++ vstr d5, [a1, #6 * 2*4] ++ vstr d0, [a1, #8 * 2*4] ++ vstr d1, [a1, #12 * 2*4] ++ vstr d6, [a1, #10 * 2*4] ++ vstr d7, [a1, #14 * 2*4] ++ vstr d8, [a1, #0 * 2*4] ++ vstr d9, [a1, #4 * 2*4] ++ ++ vpop {s16-s31} ++ vmsr FPSCR, a2 ++ bx lr ++endfunc +diff --git a/libavcodec/arm/mdct_vfp.S b/libavcodec/arm/mdct_vfp.S +index 2e2126e..56ad48e 100644 +--- a/libavcodec/arm/mdct_vfp.S ++++ b/libavcodec/arm/mdct_vfp.S +@@ -174,9 +174,8 @@ function ff_imdct_half_vfp, export=1 + .endr + + vmsr FPSCR, OLDFPSCR +- mov ORIGOUT, OUT +- ldr ip, [CONTEXT, #9*4] +- blx ip @ s->fft_calc(s, output) ++ mov a1, OUT ++ bl ff_fft16_vfp + ldr lr, =0x03030000 @ RunFast mode, short vectors of length 4, stride 1 + vmsr FPSCR, lr + +-- +1.8.1.6 +