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authorfoo86 <foobaz86@gmail.com>2016-01-16 11:07:08 +0300
committerHendrik Leppkes <h.leppkes@gmail.com>2016-01-31 17:09:38 +0100
commit46089967722f74e794865a044f5f682f26628802 (patch)
treeb4ca91d42d3eb0da3229d217323565738c101f87
parentb552f3afa2a76142c9aa87a89e31e75423b4cd3b (diff)
avcodec/dca: remove old decoder
Remove all files and functions which are not going to be reused, and disable all functions and FATE tests temporarily which will be.
Diffstat
-rwxr-xr-xconfigure1
-rw-r--r--libavcodec/Makefile3
-rw-r--r--libavcodec/aarch64/Makefile6
-rw-r--r--libavcodec/aarch64/dcadsp_init.c39
-rw-r--r--libavcodec/aarch64/dcadsp_neon.S109
-rw-r--r--libavcodec/allcodecs.c2
-rw-r--r--libavcodec/arm/Makefile9
-rw-r--r--libavcodec/arm/dca.h1
-rw-r--r--libavcodec/arm/dcadsp_init_arm.c53
-rw-r--r--libavcodec/arm/dcadsp_neon.S64
-rw-r--r--libavcodec/arm/dcadsp_vfp.S476
-rw-r--r--libavcodec/dca.h287
-rw-r--r--libavcodec/dca_exss.c373
-rw-r--r--libavcodec/dca_xll.c747
-rw-r--r--libavcodec/dcadata.c318
-rw-r--r--libavcodec/dcadata.h10
-rw-r--r--libavcodec/dcadec.c2067
-rw-r--r--libavcodec/dcadsp.c134
-rw-r--r--libavcodec/dcadsp.h51
-rw-r--r--libavcodec/dcamath.h47
-rw-r--r--libavcodec/x86/Makefile6
-rw-r--r--libavcodec/x86/dcadsp.asm123
-rw-r--r--libavcodec/x86/dcadsp_init.c42
-rw-r--r--tests/checkasm/Makefile2
-rw-r--r--tests/checkasm/checkasm.c5
-rw-r--r--tests/checkasm/checkasm.h1
-rw-r--r--tests/checkasm/dcadsp.c92
-rw-r--r--tests/fate/acodec.mak4
-rw-r--r--tests/fate/audio.mak9
29 files changed, 17 insertions, 5064 deletions
diff --git a/configure b/configure
index dba81807c2..66e11391a2 100755
--- a/configure
+++ b/configure
@@ -2271,7 +2271,6 @@ comfortnoise_encoder_select="lpc"
cook_decoder_select="audiodsp mdct sinewin"
cscd_decoder_select="lzo"
cscd_decoder_suggest="zlib"
-dca_decoder_select="fmtconvert mdct"
dds_decoder_select="texturedsp"
dirac_decoder_select="dirac_parse dwt golomb videodsp mpegvideoenc"
dnxhd_decoder_select="blockdsp idctdsp"
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index de957d8eee..1ad2e936db 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -222,9 +222,6 @@ OBJS-$(CONFIG_COMFORTNOISE_ENCODER) += cngenc.o
OBJS-$(CONFIG_CPIA_DECODER) += cpia.o
OBJS-$(CONFIG_CSCD_DECODER) += cscd.o
OBJS-$(CONFIG_CYUV_DECODER) += cyuv.o
-OBJS-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadsp.o \
- dcadata.o dca_exss.o \
- dca_xll.o synth_filter.o
OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o dca.o dcadata.o
OBJS-$(CONFIG_DDS_DECODER) += dds.o
OBJS-$(CONFIG_DIRAC_DECODER) += diracdec.o dirac.o diracdsp.o \
diff --git a/libavcodec/aarch64/Makefile b/libavcodec/aarch64/Makefile
index 99f590c650..803f55b4cf 100644
--- a/libavcodec/aarch64/Makefile
+++ b/libavcodec/aarch64/Makefile
@@ -1,5 +1,4 @@
-OBJS-$(CONFIG_DCA_DECODER) += aarch64/dcadsp_init.o \
- aarch64/synth_filter_init.o
+#OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_init.o
OBJS-$(CONFIG_FFT) += aarch64/fft_init_aarch64.o
OBJS-$(CONFIG_FMTCONVERT) += aarch64/fmtconvert_init.o
OBJS-$(CONFIG_H264CHROMA) += aarch64/h264chroma_init_aarch64.o
@@ -18,8 +17,7 @@ OBJS-$(CONFIG_VORBIS_DECODER) += aarch64/vorbisdsp_init.o
ARMV8-OBJS-$(CONFIG_VIDEODSP) += aarch64/videodsp.o
-NEON-OBJS-$(CONFIG_DCA_DECODER) += aarch64/dcadsp_neon.o \
- aarch64/synth_filter_neon.o
+#NEON-OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_neon.o
NEON-OBJS-$(CONFIG_FFT) += aarch64/fft_neon.o
NEON-OBJS-$(CONFIG_FMTCONVERT) += aarch64/fmtconvert_neon.o
NEON-OBJS-$(CONFIG_H264CHROMA) += aarch64/h264cmc_neon.o
diff --git a/libavcodec/aarch64/dcadsp_init.c b/libavcodec/aarch64/dcadsp_init.c
deleted file mode 100644
index 4440e4b95f..0000000000
--- a/libavcodec/aarch64/dcadsp_init.c
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "config.h"
-
-#include "libavutil/aarch64/cpu.h"
-#include "libavutil/attributes.h"
-#include "libavutil/internal.h"
-#include "libavcodec/dcadsp.h"
-
-void ff_dca_lfe_fir0_neon(float *out, const float *in, const float *coefs);
-void ff_dca_lfe_fir1_neon(float *out, const float *in, const float *coefs);
-
-av_cold void ff_dcadsp_init_aarch64(DCADSPContext *s)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (have_neon(cpu_flags)) {
- s->lfe_fir[0] = ff_dca_lfe_fir0_neon;
- s->lfe_fir[1] = ff_dca_lfe_fir1_neon;
- }
-}
diff --git a/libavcodec/aarch64/dcadsp_neon.S b/libavcodec/aarch64/dcadsp_neon.S
deleted file mode 100644
index 0426dc6f46..0000000000
--- a/libavcodec/aarch64/dcadsp_neon.S
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
- * Copyright (c) 2015 Janne Grunau <janne-libav@jannau.net>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/aarch64/asm.S"
-
-function ff_dca_lfe_fir0_neon, export=1
- mov x3, #32 // decifactor
- sub x1, x1, #7*4
- add x4, x0, #2*32*4 - 16 // out2
- mov x7, #-16
-
- ld1 {v0.4s,v1.4s}, [x1]
- // reverse [-num_coeffs + 1, 0]
- ext v3.16b, v0.16b, v0.16b, #8
- ext v2.16b, v1.16b, v1.16b, #8
- rev64 v3.4s, v3.4s
- rev64 v2.4s, v2.4s
-1:
- ld1 {v4.4s,v5.4s}, [x2], #32
- ld1 {v6.4s,v7.4s}, [x2], #32
- subs x3, x3, #4
- fmul v16.4s, v2.4s, v4.4s
- fmul v23.4s, v0.4s, v4.4s
- fmul v17.4s, v2.4s, v6.4s
- fmul v22.4s, v0.4s, v6.4s
-
- fmla v16.4s, v3.4s, v5.4s
- fmla v23.4s, v1.4s, v5.4s
- ld1 {v4.4s,v5.4s}, [x2], #32
- fmla v17.4s, v3.4s, v7.4s
- fmla v22.4s, v1.4s, v7.4s
- ld1 {v6.4s,v7.4s}, [x2], #32
- fmul v18.4s, v2.4s, v4.4s
- fmul v21.4s, v0.4s, v4.4s
- fmul v19.4s, v2.4s, v6.4s
- fmul v20.4s, v0.4s, v6.4s
-
- fmla v18.4s, v3.4s, v5.4s
- fmla v21.4s, v1.4s, v5.4s
- fmla v19.4s, v3.4s, v7.4s
- fmla v20.4s, v1.4s, v7.4s
-
- faddp v16.4s, v16.4s, v17.4s
- faddp v18.4s, v18.4s, v19.4s
- faddp v20.4s, v20.4s, v21.4s
- faddp v22.4s, v22.4s, v23.4s
- faddp v16.4s, v16.4s, v18.4s
- faddp v20.4s, v20.4s, v22.4s
-
- st1 {v16.4s}, [x0], #16
- st1 {v20.4s}, [x4], x7
- b.gt 1b
-
- ret
-endfunc
-
-function ff_dca_lfe_fir1_neon, export=1
- mov x3, #64 // decifactor
- sub x1, x1, #3*4
- add x4, x0, #2*64*4 - 16 // out2
- mov x7, #-16
-
- ld1 {v0.4s}, [x1]
- // reverse [-num_coeffs + 1, 0]
- ext v1.16b, v0.16b, v0.16b, #8
- rev64 v1.4s, v1.4s
-
-1:
- ld1 {v4.4s,v5.4s}, [x2], #32
- ld1 {v6.4s,v7.4s}, [x2], #32
- subs x3, x3, #4
- fmul v16.4s, v1.4s, v4.4s
- fmul v23.4s, v0.4s, v4.4s
- fmul v17.4s, v1.4s, v5.4s
- fmul v22.4s, v0.4s, v5.4s
- fmul v18.4s, v1.4s, v6.4s
- fmul v21.4s, v0.4s, v6.4s
- fmul v19.4s, v1.4s, v7.4s
- fmul v20.4s, v0.4s, v7.4s
- faddp v16.4s, v16.4s, v17.4s
- faddp v18.4s, v18.4s, v19.4s
- faddp v20.4s, v20.4s, v21.4s
- faddp v22.4s, v22.4s, v23.4s
- faddp v16.4s, v16.4s, v18.4s
- faddp v20.4s, v20.4s, v22.4s
- st1 {v16.4s}, [x0], #16
- st1 {v20.4s}, [x4], x7
- b.gt 1b
-
- ret
-endfunc
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index c7c1af5834..b17472933d 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -391,7 +391,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER(BMV_AUDIO, bmv_audio);
REGISTER_DECODER(COOK, cook);
- REGISTER_ENCDEC (DCA, dca);
+ REGISTER_ENCODER(DCA, dca);
REGISTER_DECODER(DSD_LSBF, dsd_lsbf);
REGISTER_DECODER(DSD_MSBF, dsd_msbf);
REGISTER_DECODER(DSD_LSBF_PLANAR, dsd_lsbf_planar);
diff --git a/libavcodec/arm/Makefile b/libavcodec/arm/Makefile
index 6a29a5fbb7..b2f5a5aec5 100644
--- a/libavcodec/arm/Makefile
+++ b/libavcodec/arm/Makefile
@@ -36,8 +36,7 @@ OBJS-$(CONFIG_VP8DSP) += arm/vp8dsp_init_arm.o
# decoders/encoders
OBJS-$(CONFIG_AAC_DECODER) += arm/aacpsdsp_init_arm.o \
arm/sbrdsp_init_arm.o
-OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_init_arm.o \
- arm/synth_filter_init_arm.o
+#OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_init_arm.o
OBJS-$(CONFIG_HEVC_DECODER) += arm/hevcdsp_init_arm.o
OBJS-$(CONFIG_MLP_DECODER) += arm/mlpdsp_init_arm.o
OBJS-$(CONFIG_RV40_DECODER) += arm/rv40dsp_init_arm.o
@@ -88,8 +87,7 @@ VFP-OBJS-$(CONFIG_FMTCONVERT) += arm/fmtconvert_vfp.o
VFP-OBJS-$(CONFIG_MDCT) += arm/mdct_vfp.o
# decoders/encoders
-VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_vfp.o \
- arm/synth_filter_vfp.o
+#VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_vfp.o
# NEON optimizations
@@ -128,8 +126,7 @@ NEON-OBJS-$(CONFIG_VP8DSP) += arm/vp8dsp_init_neon.o \
NEON-OBJS-$(CONFIG_AAC_DECODER) += arm/aacpsdsp_neon.o \
arm/sbrdsp_neon.o
NEON-OBJS-$(CONFIG_LLAUDDSP) += arm/lossless_audiodsp_neon.o
-NEON-OBJS-$(CONFIG_DCA_DECODER) += arm/dcadsp_neon.o \
- arm/synth_filter_neon.o
+#NEON-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_neon.o
NEON-OBJS-$(CONFIG_HEVC_DECODER) += arm/hevcdsp_init_neon.o \
arm/hevcdsp_deblock_neon.o \
arm/hevcdsp_idct_neon.o \
diff --git a/libavcodec/arm/dca.h b/libavcodec/arm/dca.h
index 6e87111a32..ae4b730a8a 100644
--- a/libavcodec/arm/dca.h
+++ b/libavcodec/arm/dca.h
@@ -24,7 +24,6 @@
#include <stdint.h>
#include "config.h"
-#include "libavcodec/dcadsp.h"
#include "libavcodec/mathops.h"
#if HAVE_ARMV6_INLINE && AV_GCC_VERSION_AT_LEAST(4,4) && !CONFIG_THUMB
diff --git a/libavcodec/arm/dcadsp_init_arm.c b/libavcodec/arm/dcadsp_init_arm.c
deleted file mode 100644
index febb4445d2..0000000000
--- a/libavcodec/arm/dcadsp_init_arm.c
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "config.h"
-
-#include "libavutil/arm/cpu.h"
-#include "libavutil/attributes.h"
-#include "libavcodec/dcadsp.h"
-
-void ff_dca_lfe_fir0_neon(float *out, const float *in, const float *coefs);
-void ff_dca_lfe_fir1_neon(float *out, const float *in, const float *coefs);
-
-void ff_dca_lfe_fir32_vfp(float *out, const float *in, const float *coefs);
-void ff_dca_lfe_fir64_vfp(float *out, const float *in, const float *coefs);
-
-void ff_dca_qmf_32_subbands_vfp(float samples_in[32][8], int sb_act,
- SynthFilterContext *synth, FFTContext *imdct,
- float synth_buf_ptr[512],
- int *synth_buf_offset, float synth_buf2[32],
- const float window[512], float *samples_out,
- float raXin[32], float scale);
-
-av_cold void ff_dcadsp_init_arm(DCADSPContext *s)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (have_vfp_vm(cpu_flags)) {
- s->lfe_fir[0] = ff_dca_lfe_fir32_vfp;
- s->lfe_fir[1] = ff_dca_lfe_fir64_vfp;
- s->qmf_32_subbands = ff_dca_qmf_32_subbands_vfp;
- }
- if (have_neon(cpu_flags)) {
- s->lfe_fir[0] = ff_dca_lfe_fir0_neon;
- s->lfe_fir[1] = ff_dca_lfe_fir1_neon;
- }
-}
diff --git a/libavcodec/arm/dcadsp_neon.S b/libavcodec/arm/dcadsp_neon.S
deleted file mode 100644
index 101fee0884..0000000000
--- a/libavcodec/arm/dcadsp_neon.S
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/arm/asm.S"
-
-function ff_dca_lfe_fir0_neon, export=1
- push {r4-r6,lr}
- mov r3, #32 @ decifactor
- mov r6, #256/32
- b dca_lfe_fir
-endfunc
-
-function ff_dca_lfe_fir1_neon, export=1
- push {r4-r6,lr}
- mov r3, #64 @ decifactor
- mov r6, #256/64
-dca_lfe_fir:
- add r4, r0, r3, lsl #2 @ out2
- add r5, r2, #256*4-16 @ cf1
- sub r1, r1, #12
- mov lr, #-16
-1:
- vmov.f32 q2, #0.0 @ v0
- vmov.f32 q3, #0.0 @ v1
- mov r12, r6
-2:
- vld1.32 {q8}, [r2,:128]! @ cf0
- vld1.32 {q9}, [r5,:128], lr @ cf1
- vld1.32 {q1}, [r1], lr @ in
- subs r12, r12, #4
- vrev64.32 q10, q8
- vmla.f32 q3, q1, q9
- vmla.f32 d4, d2, d21
- vmla.f32 d5, d3, d20
- bne 2b
-
- add r1, r1, r6, lsl #2
- subs r3, r3, #1
- vadd.f32 d4, d4, d5
- vadd.f32 d6, d6, d7
- vpadd.f32 d5, d4, d6
- vst1.32 {d5[0]}, [r0,:32]!
- vst1.32 {d5[1]}, [r4,:32]!
- bne 1b
-
- pop {r4-r6,pc}
-endfunc
diff --git a/libavcodec/arm/dcadsp_vfp.S b/libavcodec/arm/dcadsp_vfp.S
deleted file mode 100644
index 2e09f0ee5d..0000000000
--- a/libavcodec/arm/dcadsp_vfp.S
+++ /dev/null
@@ -1,476 +0,0 @@
-/*
- * Copyright (c) 2013 RISC OS Open Ltd
- * Author: Ben Avison <bavison@riscosopen.org>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/arm/asm.S"
-
-POUT .req a1
-PIN .req a2
-PCOEF .req a3
-OLDFPSCR .req a4
-COUNTER .req ip
-
-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 "\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
-function ff_dca_lfe_fir\decifactor\()_vfp, export=1
- fmrx OLDFPSCR, FPSCR
- ldr ip, =0x03030000 @ RunFast mode, short vectors of length 4, stride 1
- fmxr FPSCR, ip
- vldr IN0, [PIN, #-0*4]
- vldr IN1, [PIN, #-1*4]
- vldr IN2, [PIN, #-2*4]
- vldr IN3, [PIN, #-3*4]
- .if \decifactor == 32
- .set JMAX, 8
- vpush {s16-s31}
- 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
- fmxr FPSCR, OLDFPSCR
- bx lr
-endfunc
-.endm
-
- dca_lfe_fir 64
- .ltorg
- dca_lfe_fir 32
-
- .unreq POUT
- .unreq PIN
- .unreq PCOEF
- .unreq OLDFPSCR
- .unreq COUNTER
-
- .unreq IN0
- .unreq IN1
- .unreq IN2
- .unreq IN3
- .unreq IN4
- .unreq IN5
- .unreq IN6
- .unreq IN7
- .unreq COEF0
- .unreq COEF1
- .unreq COEF2
- .unreq COEF3
- .unreq COEF4
- .unreq COEF5
- .unreq COEF6
- .unreq COEF7
- .unreq ACCUM0
- .unreq ACCUM4
- .unreq POST0
- .unreq POST1
- .unreq POST2
- .unreq POST3
-
-
-IN .req a1
-SBACT .req a2
-OLDFPSCR .req a3
-IMDCT .req a4
-WINDOW .req v1
-OUT .req v2
-BUF .req v3
-SCALEINT .req v4 @ only used in softfp case
-COUNT .req v5
-
-SCALE .req s0
-
-/* Stack layout differs in softfp and hardfp cases:
- *
- * hardfp
- * fp -> 6 arg words saved by caller
- * a3,a4,v1-v3,v5,fp,lr on entry (a3 just to pad to 8 bytes)
- * s16-s23 on entry
- * align 16
- * buf -> 8*32*4 bytes buffer
- * s0 on entry
- * sp -> 3 arg words for callee
- *
- * softfp
- * fp -> 7 arg words saved by caller
- * a4,v1-v5,fp,lr on entry
- * s16-s23 on entry
- * align 16
- * buf -> 8*32*4 bytes buffer
- * sp -> 4 arg words for callee
- */
-
-/* void ff_dca_qmf_32_subbands_vfp(float samples_in[32][8], int sb_act,
- * SynthFilterContext *synth, FFTContext *imdct,
- * float (*synth_buf_ptr)[512],
- * int *synth_buf_offset, float (*synth_buf2)[32],
- * const float (*window)[512], float *samples_out,
- * float (*raXin)[32], float scale);
- */
-function ff_dca_qmf_32_subbands_vfp, export=1
-VFP push {a3-a4,v1-v3,v5,fp,lr}
-NOVFP push {a4,v1-v5,fp,lr}
- add fp, sp, #8*4
- vpush {s16-s23}
- @ The buffer pointed at by raXin isn't big enough for us to do a
- @ complete matrix transposition as we want to, so allocate an
- @ alternative buffer from the stack. Align to 4 words for speed.
- sub BUF, sp, #8*32*4
- bic BUF, BUF, #15
- mov sp, BUF
- ldr lr, =0x03330000 @ RunFast mode, short vectors of length 4, stride 2
- fmrx OLDFPSCR, FPSCR
- fmxr FPSCR, lr
- @ COUNT is used to count down 2 things at once:
- @ bits 0-4 are the number of word pairs remaining in the output row
- @ bits 5-31 are the number of words to copy (with possible negation)
- @ from the source matrix before we start zeroing the remainder
- mov COUNT, #(-4 << 5) + 16
- adds COUNT, COUNT, SBACT, lsl #5
- bmi 2f
-1:
- vldr s8, [IN, #(0*8+0)*4]
- vldr s10, [IN, #(0*8+1)*4]
- vldr s12, [IN, #(0*8+2)*4]
- vldr s14, [IN, #(0*8+3)*4]
- vldr s16, [IN, #(0*8+4)*4]
- vldr s18, [IN, #(0*8+5)*4]
- vldr s20, [IN, #(0*8+6)*4]
- vldr s22, [IN, #(0*8+7)*4]
- vneg.f s8, s8
- vldr s9, [IN, #(1*8+0)*4]
- vldr s11, [IN, #(1*8+1)*4]
- vldr s13, [IN, #(1*8+2)*4]
- vldr s15, [IN, #(1*8+3)*4]
- vneg.f s16, s16
- vldr s17, [IN, #(1*8+4)*4]
- vldr s19, [IN, #(1*8+5)*4]
- vldr s21, [IN, #(1*8+6)*4]
- vldr s23, [IN, #(1*8+7)*4]
- vstr d4, [BUF, #(0*32+0)*4]
- vstr d5, [BUF, #(1*32+0)*4]
- vstr d6, [BUF, #(2*32+0)*4]
- vstr d7, [BUF, #(3*32+0)*4]
- vstr d8, [BUF, #(4*32+0)*4]
- vstr d9, [BUF, #(5*32+0)*4]
- vstr d10, [BUF, #(6*32+0)*4]
- vstr d11, [BUF, #(7*32+0)*4]
- vldr s9, [IN, #(3*8+0)*4]
- vldr s11, [IN, #(3*8+1)*4]
- vldr s13, [IN, #(3*8+2)*4]
- vldr s15, [IN, #(3*8+3)*4]
- vldr s17, [IN, #(3*8+4)*4]
- vldr s19, [IN, #(3*8+5)*4]
- vldr s21, [IN, #(3*8+6)*4]
- vldr s23, [IN, #(3*8+7)*4]
- vneg.f s9, s9
- vldr s8, [IN, #(2*8+0)*4]
- vldr s10, [IN, #(2*8+1)*4]
- vldr s12, [IN, #(2*8+2)*4]
- vldr s14, [IN, #(2*8+3)*4]
- vneg.f s17, s17
- vldr s16, [IN, #(2*8+4)*4]
- vldr s18, [IN, #(2*8+5)*4]
- vldr s20, [IN, #(2*8+6)*4]
- vldr s22, [IN, #(2*8+7)*4]
- vstr d4, [BUF, #(0*32+2)*4]
- vstr d5, [BUF, #(1*32+2)*4]
- vstr d6, [BUF, #(2*32+2)*4]
- vstr d7, [BUF, #(3*32+2)*4]
- vstr d8, [BUF, #(4*32+2)*4]
- vstr d9, [BUF, #(5*32+2)*4]
- vstr d10, [BUF, #(6*32+2)*4]
- vstr d11, [BUF, #(7*32+2)*4]
- add IN, IN, #4*8*4
- add BUF, BUF, #4*4
- subs COUNT, COUNT, #(4 << 5) + 2
- bpl 1b
-2: @ Now deal with trailing < 4 samples
- adds COUNT, COUNT, #3 << 5
- bmi 4f @ sb_act was a multiple of 4
- bics lr, COUNT, #0x1F
- bne 3f
- @ sb_act was n*4+1
- vldr s8, [IN, #(0*8+0)*4]
- vldr s10, [IN, #(0*8+1)*4]
- vldr s12, [IN, #(0*8+2)*4]
- vldr s14, [IN, #(0*8+3)*4]
- vldr s16, [IN, #(0*8+4)*4]
- vldr s18, [IN, #(0*8+5)*4]
- vldr s20, [IN, #(0*8+6)*4]
- vldr s22, [IN, #(0*8+7)*4]
- vneg.f s8, s8
- vldr s9, zero
- vldr s11, zero
- vldr s13, zero
- vldr s15, zero
- vneg.f s16, s16
- vldr s17, zero
- vldr s19, zero
- vldr s21, zero
- vldr s23, zero
- vstr d4, [BUF, #(0*32+0)*4]
- vstr d5, [BUF, #(1*32+0)*4]
- vstr d6, [BUF, #(2*32+0)*4]
- vstr d7, [BUF, #(3*32+0)*4]
- vstr d8, [BUF, #(4*32+0)*4]
- vstr d9, [BUF, #(5*32+0)*4]
- vstr d10, [BUF, #(6*32+0)*4]
- vstr d11, [BUF, #(7*32+0)*4]
- add BUF, BUF, #2*4
- sub COUNT, COUNT, #1
- b 4f
-3: @ sb_act was n*4+2 or n*4+3, so do the first 2
- vldr s8, [IN, #(0*8+0)*4]
- vldr s10, [IN, #(0*8+1)*4]
- vldr s12, [IN, #(0*8+2)*4]
- vldr s14, [IN, #(0*8+3)*4]
- vldr s16, [IN, #(0*8+4)*4]
- vldr s18, [IN, #(0*8+5)*4]
- vldr s20, [IN, #(0*8+6)*4]
- vldr s22, [IN, #(0*8+7)*4]
- vneg.f s8, s8
- vldr s9, [IN, #(1*8+0)*4]
- vldr s11, [IN, #(1*8+1)*4]
- vldr s13, [IN, #(1*8+2)*4]
- vldr s15, [IN, #(1*8+3)*4]
- vneg.f s16, s16
- vldr s17, [IN, #(1*8+4)*4]
- vldr s19, [IN, #(1*8+5)*4]
- vldr s21, [IN, #(1*8+6)*4]
- vldr s23, [IN, #(1*8+7)*4]
- vstr d4, [BUF, #(0*32+0)*4]
- vstr d5, [BUF, #(1*32+0)*4]
- vstr d6, [BUF, #(2*32+0)*4]
- vstr d7, [BUF, #(3*32+0)*4]
- vstr d8, [BUF, #(4*32+0)*4]
- vstr d9, [BUF, #(5*32+0)*4]
- vstr d10, [BUF, #(6*32+0)*4]
- vstr d11, [BUF, #(7*32+0)*4]
- add BUF, BUF, #2*4
- sub COUNT, COUNT, #(2 << 5) + 1
- bics lr, COUNT, #0x1F
- bne 4f
- @ sb_act was n*4+3
- vldr s8, [IN, #(2*8+0)*4]
- vldr s10, [IN, #(2*8+1)*4]
- vldr s12, [IN, #(2*8+2)*4]
- vldr s14, [IN, #(2*8+3)*4]
- vldr s16, [IN, #(2*8+4)*4]
- vldr s18, [IN, #(2*8+5)*4]
- vldr s20, [IN, #(2*8+6)*4]
- vldr s22, [IN, #(2*8+7)*4]
- vldr s9, zero
- vldr s11, zero
- vldr s13, zero
- vldr s15, zero
- vldr s17, zero
- vldr s19, zero
- vldr s21, zero
- vldr s23, zero
- vstr d4, [BUF, #(0*32+0)*4]
- vstr d5, [BUF, #(1*32+0)*4]
- vstr d6, [BUF, #(2*32+0)*4]
- vstr d7, [BUF, #(3*32+0)*4]
- vstr d8, [BUF, #(4*32+0)*4]
- vstr d9, [BUF, #(5*32+0)*4]
- vstr d10, [BUF, #(6*32+0)*4]
- vstr d11, [BUF, #(7*32+0)*4]
- add BUF, BUF, #2*4
- sub COUNT, COUNT, #1
-4: @ Now fill the remainder with 0
- vldr s8, zero
- vldr s9, zero
- ands COUNT, COUNT, #0x1F
- beq 6f
-5: vstr d4, [BUF, #(0*32+0)*4]
- vstr d4, [BUF, #(1*32+0)*4]
- vstr d4, [BUF, #(2*32+0)*4]
- vstr d4, [BUF, #(3*32+0)*4]
- vstr d4, [BUF, #(4*32+0)*4]
- vstr d4, [BUF, #(5*32+0)*4]
- vstr d4, [BUF, #(6*32+0)*4]
- vstr d4, [BUF, #(7*32+0)*4]
- add BUF, BUF, #2*4
- subs COUNT, COUNT, #1
- bne 5b
-6:
- fmxr FPSCR, OLDFPSCR
- ldr WINDOW, [fp, #3*4]
- ldr OUT, [fp, #4*4]
- sub BUF, BUF, #32*4
-NOVFP ldr SCALEINT, [fp, #6*4]
- mov COUNT, #8
-VFP vpush {SCALE}
-VFP sub sp, sp, #3*4
-NOVFP sub sp, sp, #4*4
-7:
-VFP ldr a1, [fp, #-7*4] @ imdct
-NOVFP ldr a1, [fp, #-8*4]
- ldmia fp, {a2-a4}
-VFP stmia sp, {WINDOW, OUT, BUF}
-NOVFP stmia sp, {WINDOW, OUT, BUF, SCALEINT}
-VFP vldr SCALE, [sp, #3*4]
- bl X(ff_synth_filter_float_vfp)
- add OUT, OUT, #32*4
- add BUF, BUF, #32*4
- subs COUNT, COUNT, #1
- bne 7b
-
-A sub sp, fp, #(8+8)*4
-T sub fp, fp, #(8+8)*4
-T mov sp, fp
- vpop {s16-s23}
-VFP pop {a3-a4,v1-v3,v5,fp,pc}
-NOVFP pop {a4,v1-v5,fp,pc}
-endfunc
-
- .unreq IN
- .unreq SBACT
- .unreq OLDFPSCR
- .unreq IMDCT
- .unreq WINDOW
- .unreq OUT
- .unreq BUF
- .unreq SCALEINT
- .unreq COUNT
-
- .unreq SCALE
-
- .align 2
-zero: .word 0
diff --git a/libavcodec/dca.h b/libavcodec/dca.h
index dea82aeb2a..ea3f9c5d0d 100644
--- a/libavcodec/dca.h
+++ b/libavcodec/dca.h
@@ -27,282 +27,8 @@
#include <stdint.h>
-#include "libavutil/float_dsp.h"
#include "libavutil/internal.h"
-
-#include "avcodec.h"
-#include "dcadsp.h"
-#include "fmtconvert.h"
-#include "get_bits.h"
-
-#define DCA_PRIM_CHANNELS_MAX (7)
-#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)
-
-#define DCA_PRIM_CHANNELS_MAX (7)
-#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_XLL_FBANDS_MAX (4)
-#define DCA_XLL_SEGMENTS_MAX (16)
-#define DCA_XLL_CHSETS_MAX (16)
-#define DCA_XLL_CHANNELS_MAX (16)
-#define DCA_XLL_AORDER_MAX (15)
-
-/* Arbitrary limit; not sure what the maximum really is, but much larger. */
-#define DCA_XLL_DMIX_NCOEFFS_MAX (18)
-
-#define DCA_MAX_FRAME_SIZE 16384
-#define DCA_MAX_EXSS_HEADER_SIZE 4096
-
-#define DCA_BUFFER_PADDING_SIZE 1024
-
-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
-};
-
-typedef struct XllChSetSubHeader {
- int channels; ///< number of channels in channel set, at most 16
- int residual_encode; ///< residual channel encoding
- int bit_resolution; ///< input sample bit-width
- int bit_width; ///< original input sample bit-width
- int sampling_frequency; ///< sampling frequency
- int samp_freq_interp; ///< sampling frequency interpolation multiplier
- int replacement_set; ///< replacement channel set group
- int active_replace_set; ///< current channel set is active channel set
- int primary_ch_set;
- int downmix_coeff_code_embedded;
- int downmix_embedded;
- int downmix_type;
- int hier_chset; ///< hierarchical channel set
- int downmix_ncoeffs;
- int downmix_coeffs[DCA_XLL_DMIX_NCOEFFS_MAX];
- int ch_mask_enabled;
- int ch_mask;
- int mapping_coeffs_present;
- int num_freq_bands;
-
- /* m_nOrigChanOrder */
- uint8_t orig_chan_order[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX];
- uint8_t orig_chan_order_inv[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX];
- /* Coefficients for channel pairs (at most 8), m_anPWChPairsCoeffs */
- int8_t pw_ch_pairs_coeffs[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX/2];
- /* m_nCurrHighestLPCOrder */
- uint8_t adapt_order_max[DCA_XLL_FBANDS_MAX];
- /* m_pnAdaptPredOrder */
- uint8_t adapt_order[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX];
- /* m_pnFixedPredOrder */
- uint8_t fixed_order[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX];
- /* m_pnLPCReflCoeffsQInd, unsigned version */
- uint8_t lpc_refl_coeffs_q_ind[DCA_XLL_FBANDS_MAX]
- [DCA_XLL_CHANNELS_MAX][DCA_XLL_AORDER_MAX];
-
- int lsb_fsize[DCA_XLL_FBANDS_MAX];
- int8_t scalable_lsbs[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX];
- int8_t bit_width_adj_per_ch[DCA_XLL_FBANDS_MAX][DCA_XLL_CHANNELS_MAX];
-} XllChSetSubHeader;
-
-typedef struct XllNavi {
- GetBitContext gb; // Context for parsing the data segments
- unsigned band_size[DCA_XLL_FBANDS_MAX];
- unsigned segment_size[DCA_XLL_FBANDS_MAX][DCA_XLL_SEGMENTS_MAX];
- unsigned chset_size[DCA_XLL_FBANDS_MAX][DCA_XLL_SEGMENTS_MAX][DCA_XLL_CHSETS_MAX];
-} XllNavi;
-
-typedef struct QMF64_table {
- float dct4_coeff[32][32];
- float dct2_coeff[32][32];
- float rcos[32];
- float rsin[32];
-} QMF64_table;
-
-/* Primary audio coding header */
-typedef struct DCAAudioHeader {
- 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
- uint32_t scalefactor_adj[DCA_PRIM_CHANNELS_MAX][DCA_ABITS_MAX]; ///< scale factor adjustment
-
- int subframes; ///< number of subframes
- int total_channels; ///< number of channels including extensions
- int prim_channels; ///< number of primary audio channels
-} DCAAudioHeader;
-
-typedef struct DCAChan {
- DECLARE_ALIGNED(32, int32_t, subband_samples)[DCA_BLOCKS_MAX][DCA_SUBBANDS][SAMPLES_PER_SUBBAND];
-
- /* Subband samples history (for ADPCM) */
- DECLARE_ALIGNED(32, int32_t, subband_samples_hist)[DCA_SUBBANDS][4];
- int hist_index;
-
- /* Half size is sufficient for core decoding, but for 96 kHz data
- * we need QMF with 64 subbands and 1024 samples. */
- DECLARE_ALIGNED(32, float, subband_fir_hist)[1024];
- DECLARE_ALIGNED(32, float, subband_fir_noidea)[64];
-
- /* Primary audio coding side information */
- int prediction_mode[DCA_SUBBANDS]; ///< prediction mode (ADPCM used or not)
- int prediction_vq[DCA_SUBBANDS]; ///< prediction VQ coefs
- int bitalloc[DCA_SUBBANDS]; ///< bit allocation index
- int transition_mode[DCA_SUBBANDS]; ///< transition mode (transients)
- int32_t scale_factor[DCA_SUBBANDS][2];///< scale factors (2 if transient)
- int joint_huff; ///< joint subband scale factors codebook
- int joint_scale_factor[DCA_SUBBANDS]; ///< joint subband scale factors
-
- int32_t high_freq_vq[DCA_SUBBANDS]; ///< VQ encoded high frequency subbands
-} DCAChan;
-
-
-typedef struct DCAContext {
- const AVClass *class; ///< class for AVOptions
- 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 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 */
- DCAAudioHeader audio_header;
-
- /* Primary audio coding side information */
- int subsubframes[DCA_SUBFRAMES_MAX]; ///< number of subsubframes
- int partial_samples[DCA_SUBFRAMES_MAX]; ///< partial subsubframe samples count
- float downmix_coef[DCA_PRIM_CHANNELS_MAX + 1][2]; ///< stereo downmix coefficients
- int dynrange_coef; ///< dynamic range coefficient
-
- /* Core substream's embedded downmix coefficients (cf. ETSI TS 102 114 V1.4.1)
- * Input: primary audio channels (incl. LFE if present)
- * Output: downmix audio channels (up to 4, no LFE) */
- uint8_t core_downmix; ///< embedded downmix coefficients available
- uint8_t core_downmix_amode; ///< audio channel arrangement of embedded downmix
- uint16_t core_downmix_codes[DCA_PRIM_CHANNELS_MAX + 1][4]; ///< embedded downmix coefficients (9-bit codes)
-
-
- 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(32, float, raXin)[32];
-
- DCAChan dca_chan[DCA_PRIM_CHANNELS_MAX];
-
- int output; ///< type of output
-
- 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
- int exss_ext_mask; ///< Non-core extensions
-
- /* XCh extension information */
- int xch_present; ///< XCh extension present and valid
- int xch_base_channel; ///< index of first (only) channel containing XCH data
- int xch_disable; ///< whether the XCh extension should be decoded or not
-
- /* 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;
-
- /* XLL extension information */
- int xll_disable;
- int xll_nch_sets; ///< number of channel sets per frame
- int xll_channels; ///< total number of channels (in all channel sets)
- int xll_residual_channels; ///< number of residual channels
- int xll_segments; ///< number of segments per frame
- int xll_log_smpl_in_seg; ///< supposedly this is "nBits4SamplLoci"
- int xll_smpl_in_seg; ///< samples in segment per one frequency band for the first channel set
- int xll_bits4seg_size; ///< number of bits used to read segment size
- int xll_banddata_crc; ///< presence of CRC16 within each frequency band
- int xll_scalable_lsb;
- int xll_bits4ch_mask; ///< channel position mask
- int xll_fixed_lsb_width;
- XllChSetSubHeader xll_chsets[DCA_XLL_CHSETS_MAX];
- XllNavi xll_navi;
- int *xll_sample_buf;
- unsigned int xll_sample_buf_size;
-
- /* 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 one2one_map_chtospkr;
-
- int debug_flag; ///< used for suppressing repeated error messages output
- AVFloatDSPContext *fdsp;
- FFTContext imdct;
- SynthFilterContext synth;
- DCADSPContext dcadsp;
- QMF64_table *qmf64_table;
- FmtConvertContext fmt_conv;
-} DCAContext;
+#include "libavutil/intreadwrite.h"
extern av_export const uint32_t avpriv_dca_sample_rates[16];
@@ -310,15 +36,6 @@ extern av_export const uint32_t avpriv_dca_sample_rates[16];
* Convert bitstream to one representation based on sync marker
*/
int avpriv_dca_convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst,
- int max_size);
-
-int ff_dca_xbr_parse_frame(DCAContext *s);
-int ff_dca_xxch_decode_frame(DCAContext *s);
-
-void ff_dca_exss_parse_header(DCAContext *s);
-
-int ff_dca_xll_decode_header(DCAContext *s);
-int ff_dca_xll_decode_navi(DCAContext *s, int asset_end);
-int ff_dca_xll_decode_audio(DCAContext *s, AVFrame *frame);
+ int max_size);
#endif /* AVCODEC_DCA_H */
diff --git a/libavcodec/dca_exss.c b/libavcodec/dca_exss.c
deleted file mode 100644
index ed014906ce..0000000000
--- a/libavcodec/dca_exss.c
+++ /dev/null
@@ -1,373 +0,0 @@
-/*
- * DCA ExSS extension
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/common.h"
-#include "libavutil/log.h"
-
-#include "dca.h"
-#include "dca_syncwords.h"
-#include "get_bits.h"
-
-/* extensions that reside in core substream */
-#define DCA_CORE_EXTS (DCA_EXT_XCH | DCA_EXT_XXCH | DCA_EXT_X96)
-
-/* 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,
-};
-
-/**
- * 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 extensions_mask = 0;
- int i, j;
-
- if (get_bits_left(&s->gb) < 16)
- return AVERROR_INVALIDDATA;
-
- /* 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 AVERROR_INVALIDDATA;
- 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;
-
- s->one2one_map_chtospkr = get_bits1(&s->gb);
- if (s->one2one_map_chtospkr) {
- 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 AVERROR_INVALIDDATA;
-
- 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 AVERROR_INVALIDDATA;
- 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 AVERROR_INVALIDDATA;
-
- if (get_bits_count(&s->gb) - header_pos > header_size * 8) {
- av_log(s->avctx, AV_LOG_WARNING, "Asset header size mismatch.\n");
- return AVERROR_INVALIDDATA;
- }
- 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;
-}
-
-/**
- * Parse extension substream header (HD)
- */
-void ff_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_pos;
- int hdrsize;
- uint32_t mkr;
-
- if (get_bits_left(&s->gb) < 52)
- return;
-
- start_pos = 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) {
- avpriv_request_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) {
- avpriv_request_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);
- }
- }
- }
-
- av_assert0(num_assets > 0); // silence a warning
-
- for (i = 0; i < num_assets; i++)
- asset_size[i] = get_bits_long(&s->gb, 16 + 4 * blownup) + 1;
-
- for (i = 0; i < num_assets; i++) {
- if (dca_exss_parse_asset_header(s))
- return;
- }
-
- j = get_bits_count(&s->gb);
- if (start_pos + hdrsize * 8 > j)
- skip_bits_long(&s->gb, start_pos + hdrsize * 8 - j);
-
- for (i = 0; i < num_assets; i++) {
- int end_pos;
- start_pos = get_bits_count(&s->gb);
- end_pos = start_pos + asset_size[i] * 8;
- mkr = get_bits_long(&s->gb, 32);
-
- /* parse extensions that we know about */
- switch (mkr) {
- case DCA_SYNCWORD_XBR:
- ff_dca_xbr_parse_frame(s);
- break;
- case DCA_SYNCWORD_XXCH:
- ff_dca_xxch_decode_frame(s);
- s->core_ext_mask |= DCA_EXT_XXCH; /* xxx use for chan reordering */
- break;
- case DCA_SYNCWORD_XLL:
- if (s->xll_disable) {
- av_log(s->avctx, AV_LOG_DEBUG,
- "DTS-XLL: ignoring XLL extension\n");
- break;
- }
- av_log(s->avctx, AV_LOG_DEBUG,
- "DTS-XLL: decoding XLL extension\n");
- if (ff_dca_xll_decode_header(s) == 0 &&
- ff_dca_xll_decode_navi(s, end_pos) == 0)
- s->exss_ext_mask |= DCA_EXT_EXSS_XLL;
- break;
- default:
- 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 (j > end_pos)
- av_log(s->avctx, AV_LOG_ERROR,
- "DTS-ExSS: Processed asset too long.\n");
- if (j < end_pos)
- skip_bits_long(&s->gb, end_pos - j);
- }
-}
diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c
deleted file mode 100644
index 98fd4c8eaa..0000000000
--- a/libavcodec/dca_xll.c
+++ /dev/null
@@ -1,747 +0,0 @@
-/*
- * DCA XLL extension
- *
- * Copyright (C) 2012 Paul B Mahol
- * Copyright (C) 2014 Niels Möller
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/attributes.h"
-#include "libavutil/common.h"
-#include "libavutil/internal.h"
-
-#include "avcodec.h"
-#include "dca.h"
-#include "dcadata.h"
-#include "get_bits.h"
-#include "unary.h"
-
-/* Sign as bit 0 */
-static inline int get_bits_sm(GetBitContext *s, unsigned n)
-{
- int x = get_bits(s, n);
- if (x & 1)
- return -(x >> 1) - 1;
- else
- return x >> 1;
-}
-
-/* Return -1 on error. */
-static int32_t get_dmix_coeff(DCAContext *s, int inverse)
-{
- unsigned code = get_bits(&s->gb, 9);
- int32_t sign = (int32_t) (code >> 8) - 1;
- unsigned idx = code & 0xff;
- int inv_offset = FF_DCA_DMIXTABLE_SIZE -FF_DCA_INV_DMIXTABLE_SIZE;
- if (idx >= FF_DCA_DMIXTABLE_SIZE) {
- av_log(s->avctx, AV_LOG_ERROR,
- "XLL: Invalid channel set downmix code %x\n", code);
- return -1;
- } else if (!inverse) {
- return (ff_dca_dmixtable[idx] ^ sign) - sign;
- } else if (idx < inv_offset) {
- av_log(s->avctx, AV_LOG_ERROR,
- "XLL: Invalid channel set inverse downmix code %x\n", code);
- return -1;
- } else {
- return (ff_dca_inv_dmixtable[idx - inv_offset] ^ sign) - sign;
- }
-}
-
-static int32_t dca_get_dmix_coeff(DCAContext *s)
-{
- return get_dmix_coeff(s, 0);
-}
-
-static int32_t dca_get_inv_dmix_coeff(DCAContext *s)
-{
- return get_dmix_coeff(s, 1);
-}
-
-/* parse XLL header */
-int ff_dca_xll_decode_header(DCAContext *s)
-{
- int hdr_pos, hdr_size;
- av_unused int version, frame_size;
- int i, chset_index;
-
- /* get bit position of sync header */
- hdr_pos = get_bits_count(&s->gb) - 32;
-
- version = get_bits(&s->gb, 4) + 1;
- hdr_size = get_bits(&s->gb, 8) + 1;
-
- frame_size = get_bits_long(&s->gb, get_bits(&s->gb, 5) + 1) + 1;
-
- s->xll_channels =
- s->xll_residual_channels = 0;
- s->xll_nch_sets = get_bits(&s->gb, 4) + 1;
- s->xll_segments = 1 << get_bits(&s->gb, 4);
- s->xll_log_smpl_in_seg = get_bits(&s->gb, 4);
- s->xll_smpl_in_seg = 1 << s->xll_log_smpl_in_seg;
- s->xll_bits4seg_size = get_bits(&s->gb, 5) + 1;
- s->xll_banddata_crc = get_bits(&s->gb, 2);
- s->xll_scalable_lsb = get_bits1(&s->gb);
- s->xll_bits4ch_mask = get_bits(&s->gb, 5) + 1;
-
- if (s->xll_scalable_lsb) {
- s->xll_fixed_lsb_width = get_bits(&s->gb, 4);
- if (s->xll_fixed_lsb_width)
- av_log(s->avctx, AV_LOG_WARNING,
- "XLL: fixed lsb width = %d, non-zero not supported.\n",
- s->xll_fixed_lsb_width);
- }
- /* skip to the end of the common 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_index = 0; chset_index < s->xll_nch_sets; chset_index++) {
- XllChSetSubHeader *chset = &s->xll_chsets[chset_index];
- hdr_pos = get_bits_count(&s->gb);
- hdr_size = get_bits(&s->gb, 10) + 1;
-
- chset->channels = get_bits(&s->gb, 4) + 1;
- chset->residual_encode = get_bits(&s->gb, chset->channels);
- chset->bit_resolution = get_bits(&s->gb, 5) + 1;
- chset->bit_width = get_bits(&s->gb, 5) + 1;
- chset->sampling_frequency = ff_dca_sampling_freqs[get_bits(&s->gb, 4)];
- chset->samp_freq_interp = get_bits(&s->gb, 2);
- chset->replacement_set = get_bits(&s->gb, 2);
- if (chset->replacement_set)
- chset->active_replace_set = get_bits(&s->gb, 1);
-
- if (s->one2one_map_chtospkr) {
- chset->primary_ch_set = get_bits(&s->gb, 1);
- chset->downmix_coeff_code_embedded = get_bits(&s->gb, 1);
- if (chset->downmix_coeff_code_embedded) {
- chset->downmix_embedded = get_bits(&s->gb, 1);
- if (chset->primary_ch_set) {
- chset->downmix_type = get_bits(&s->gb, 3);
- if (chset->downmix_type > 6) {
- av_log(s->avctx, AV_LOG_ERROR,
- "XLL: Invalid channel set downmix type\n");
- return AVERROR_INVALIDDATA;
- }
- }
- }
- chset->hier_chset = get_bits(&s->gb, 1);
-
- if (chset->downmix_coeff_code_embedded) {
- /* nDownmixCoeffs is specified as N * M. For a primary
- * channel set, it appears that N = number of
- * channels, and M is the number of downmix channels.
- *
- * For a non-primary channel set, N is specified as
- * number of channels + 1, and M is derived from the
- * channel set hierarchy, and at least in simple cases
- * M is the number of channels in preceding channel
- * sets. */
- if (chset->primary_ch_set) {
- static const char dmix_table[7] = { 1, 2, 2, 3, 3, 4, 4 };
- chset->downmix_ncoeffs = chset->channels * dmix_table[chset->downmix_type];
- } else
- chset->downmix_ncoeffs = (chset->channels + 1) * s->xll_channels;
-
- if (chset->downmix_ncoeffs > DCA_XLL_DMIX_NCOEFFS_MAX) {
- avpriv_request_sample(s->avctx,
- "XLL: More than %d downmix coefficients",
- DCA_XLL_DMIX_NCOEFFS_MAX);
- return AVERROR_PATCHWELCOME;
- } else if (chset->primary_ch_set) {
- for (i = 0; i < chset->downmix_ncoeffs; i++)
- if ((chset->downmix_coeffs[i] = dca_get_dmix_coeff(s)) == -1)
- return AVERROR_INVALIDDATA;
- } else {
- unsigned c, r;
- for (c = 0, i = 0; c < s->xll_channels; c++, i += chset->channels + 1) {
- if ((chset->downmix_coeffs[i] = dca_get_inv_dmix_coeff(s)) == -1)
- return AVERROR_INVALIDDATA;
- for (r = 1; r <= chset->channels; r++) {
- int32_t coeff = dca_get_dmix_coeff(s);
- if (coeff == -1)
- return AVERROR_INVALIDDATA;
- chset->downmix_coeffs[i + r] =
- (chset->downmix_coeffs[i] * (int64_t) coeff + (1 << 15)) >> 16;
- }
- }
- }
- }
- chset->ch_mask_enabled = get_bits(&s->gb, 1);
- if (chset->ch_mask_enabled)
- chset->ch_mask = get_bits(&s->gb, s->xll_bits4ch_mask);
- else
- /* Skip speaker configuration bits */
- skip_bits_long(&s->gb, 25 * chset->channels);
- } else {
- chset->primary_ch_set = 1;
- chset->downmix_coeff_code_embedded = 0;
- /* Spec: NumChHierChSet = 0, NumDwnMixCodeCoeffs = 0, whatever that means. */
- chset->mapping_coeffs_present = get_bits(&s->gb, 1);
- if (chset->mapping_coeffs_present) {
- avpriv_report_missing_feature(s->avctx, "XLL: mapping coefficients");
- return AVERROR_PATCHWELCOME;
- }
- }
- if (chset->sampling_frequency > 96000)
- chset->num_freq_bands = 2 * (1 + get_bits(&s->gb, 1));
- else
- chset->num_freq_bands = 1;
-
- if (chset->num_freq_bands > 1) {
- avpriv_report_missing_feature(s->avctx, "XLL: num_freq_bands > 1");
- return AVERROR_PATCHWELCOME;
- }
-
- if (get_bits(&s->gb, 1)) { /* pw_ch_decor_enabled */
- int bits = av_ceil_log2(chset->channels);
- for (i = 0; i < chset->channels; i++) {
- unsigned j = get_bits(&s->gb, bits);
- if (j >= chset->channels) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Original channel order value %u too large, only %d channels.\n",
- j, chset->channels);
- return AVERROR_INVALIDDATA;
- }
- chset->orig_chan_order[0][i] = j;
- chset->orig_chan_order_inv[0][j] = i;
- }
- for (i = 0; i < chset->channels / 2; i++) {
- if (get_bits(&s->gb, 1)) /* bChPFlag */
- chset->pw_ch_pairs_coeffs[0][i] = get_bits_sm(&s->gb, 7);
- else
- chset->pw_ch_pairs_coeffs[0][i] = 0;
- }
- } else {
- for (i = 0; i < chset->channels; i++)
- chset->orig_chan_order[0][i] =
- chset->orig_chan_order_inv[0][i] = i;
- for (i = 0; i < chset->channels / 2; i++)
- chset->pw_ch_pairs_coeffs[0][i] = 0;
- }
- /* Adaptive prediction order */
- chset->adapt_order_max[0] = 0;
- for (i = 0; i < chset->channels; i++) {
- chset->adapt_order[0][i] = get_bits(&s->gb, 4);
- if (chset->adapt_order_max[0] < chset->adapt_order[0][i])
- chset->adapt_order_max[0] = chset->adapt_order[0][i];
- }
- /* Fixed prediction order, used in case the adaptive order
- * above is zero */
- for (i = 0; i < chset->channels; i++)
- chset->fixed_order[0][i] =
- chset->adapt_order[0][i] ? 0 : get_bits(&s->gb, 2);
-
- for (i = 0; i < chset->channels; i++) {
- unsigned j;
- for (j = 0; j < chset->adapt_order[0][i]; j++)
- chset->lpc_refl_coeffs_q_ind[0][i][j] = get_bits(&s->gb, 8);
- }
-
- if (s->xll_scalable_lsb) {
- chset->lsb_fsize[0] = get_bits(&s->gb, s->xll_bits4seg_size);
-
- for (i = 0; i < chset->channels; i++)
- chset->scalable_lsbs[0][i] = get_bits(&s->gb, 4);
- for (i = 0; i < chset->channels; i++)
- chset->bit_width_adj_per_ch[0][i] = get_bits(&s->gb, 4);
- } else {
- memset(chset->scalable_lsbs[0], 0,
- chset->channels * sizeof(chset->scalable_lsbs[0][0]));
- memset(chset->bit_width_adj_per_ch[0], 0,
- chset->channels * sizeof(chset->bit_width_adj_per_ch[0][0]));
- }
-
- s->xll_channels += chset->channels;
- s->xll_residual_channels += chset->channels -
- av_popcount(chset->residual_encode);
-
- /* FIXME: Parse header data for extra frequency bands. */
-
- /* Skip to end of channel set sub header. */
- i = get_bits_count(&s->gb);
- if (hdr_pos + 8 * hdr_size < i) {
- av_log(s->avctx, AV_LOG_ERROR,
- "chset header too large, %d bits, should be <= %d bits\n",
- i - hdr_pos, 8 * hdr_size);
- return AVERROR_INVALIDDATA;
- }
- if (hdr_pos + 8 * hdr_size > i)
- skip_bits_long(&s->gb, hdr_pos + 8 * hdr_size - i);
- }
- return 0;
-}
-
-/* parse XLL navigation table */
-int ff_dca_xll_decode_navi(DCAContext *s, int asset_end)
-{
- int nbands, band, chset, seg, data_start;
-
- /* FIXME: Supports only a single frequency band */
- nbands = 1;
-
- for (band = 0; band < nbands; band++) {
- s->xll_navi.band_size[band] = 0;
- for (seg = 0; seg < s->xll_segments; seg++) {
- /* Note: The spec, ETSI TS 102 114 V1.4.1 (2012-09), says
- * we should read a base value for segment_size from the
- * stream, before reading the sizes of the channel sets.
- * But that's apparently incorrect. */
- s->xll_navi.segment_size[band][seg] = 0;
-
- for (chset = 0; chset < s->xll_nch_sets; chset++)
- if (band < s->xll_chsets[chset].num_freq_bands) {
- s->xll_navi.chset_size[band][seg][chset] =
- get_bits(&s->gb, s->xll_bits4seg_size) + 1;
- s->xll_navi.segment_size[band][seg] +=
- s->xll_navi.chset_size[band][seg][chset];
- }
- s->xll_navi.band_size[band] += s->xll_navi.segment_size[band][seg];
- }
- }
- /* Align to 8 bits and skip 16-bit CRC. */
- skip_bits_long(&s->gb, 16 + ((-get_bits_count(&s->gb)) & 7));
-
- data_start = get_bits_count(&s->gb);
- if (data_start + 8 * s->xll_navi.band_size[0] > asset_end) {
- av_log(s->avctx, AV_LOG_ERROR,
- "XLL: Data in NAVI table exceeds containing asset\n"
- "start: %d (bit), size %u (bytes), end %d (bit), error %u\n",
- data_start, s->xll_navi.band_size[0], asset_end,
- data_start + 8 * s->xll_navi.band_size[0] - asset_end);
- return AVERROR_INVALIDDATA;
- }
- init_get_bits(&s->xll_navi.gb, s->gb.buffer + data_start / 8,
- 8 * s->xll_navi.band_size[0]);
- return 0;
-}
-
-static void dca_xll_inv_adapt_pred(int *samples, int nsamples, unsigned order,
- const int *prev, const uint8_t *q_ind)
-{
- static const uint16_t table[0x81] = {
- 0, 3070, 5110, 7140, 9156, 11154, 13132, 15085,
- 17010, 18904, 20764, 22588, 24373, 26117, 27818, 29474,
- 31085, 32648, 34164, 35631, 37049, 38418, 39738, 41008,
- 42230, 43404, 44530, 45609, 46642, 47630, 48575, 49477,
- 50337, 51157, 51937, 52681, 53387, 54059, 54697, 55302,
- 55876, 56421, 56937, 57426, 57888, 58326, 58741, 59132,
- 59502, 59852, 60182, 60494, 60789, 61066, 61328, 61576,
- 61809, 62029, 62236, 62431, 62615, 62788, 62951, 63105,
- 63250, 63386, 63514, 63635, 63749, 63855, 63956, 64051,
- 64140, 64224, 64302, 64376, 64446, 64512, 64573, 64631,
- 64686, 64737, 64785, 64830, 64873, 64913, 64950, 64986,
- 65019, 65050, 65079, 65107, 65133, 65157, 65180, 65202,
- 65222, 65241, 65259, 65275, 65291, 65306, 65320, 65333,
- 65345, 65357, 65368, 65378, 65387, 65396, 65405, 65413,
- 65420, 65427, 65434, 65440, 65446, 65451, 65456, 65461,
- 65466, 65470, 65474, 65478, 65481, 65485, 65488, 65491,
- 65535, /* Final value is for the -128 corner case, see below. */
- };
- int c[DCA_XLL_AORDER_MAX];
- int64_t s;
- unsigned i, j;
-
- for (i = 0; i < order; i++) {
- if (q_ind[i] & 1)
- /* The index value 0xff corresponds to a lookup of entry 0x80 in
- * the table, and no value is provided in the specification. */
- c[i] = -table[(q_ind[i] >> 1) + 1];
- else
- c[i] = table[q_ind[i] >> 1];
- }
- /* The description in the spec is a bit convoluted. We can convert
- * the reflected values to direct values in place, using a
- * sequence of reflections operating on two values. */
- for (i = 1; i < order; i++) {
- /* i = 1: scale c[0]
- * i = 2: reflect c[0] <-> c[1]
- * i = 3: scale c[1], reflect c[0] <-> c[2]
- * i = 4: reflect c[0] <-> c[3] reflect c[1] <-> c[2]
- * ... */
- if (i & 1)
- c[i / 2] += ((int64_t) c[i] * c[i / 2] + 0x8000) >> 16;
- for (j = 0; j < i / 2; j++) {
- int r0 = c[j];
- int r1 = c[i - j - 1];
- c[j] += ((int64_t) c[i] * r1 + 0x8000) >> 16;
- c[i - j - 1] += ((int64_t) c[i] * r0 + 0x8000) >> 16;
- }
- }
- /* Apply predictor. */
- /* NOTE: Processing samples in this order means that the
- * predictor is applied to the newly reconstructed samples. */
- if (prev) {
- for (i = 0; i < order; i++) {
- for (j = s = 0; j < i; j++)
- s += (int64_t) c[j] * samples[i - 1 - j];
- for (; j < order; j++)
- s += (int64_t) c[j] * prev[DCA_XLL_AORDER_MAX + i - 1 - j];
-
- samples[i] -= av_clip_intp2((s + 0x8000) >> 16, 24);
- }
- }
- for (i = order; i < nsamples; i++) {
- for (j = s = 0; j < order; j++)
- s += (int64_t) c[j] * samples[i - 1 - j];
-
- /* NOTE: Equations seem to imply addition, while the
- * pseudocode seems to use subtraction.*/
- samples[i] -= av_clip_intp2((s + 0x8000) >> 16, 24);
- }
-}
-
-int ff_dca_xll_decode_audio(DCAContext *s, AVFrame *frame)
-{
- /* FIXME: Decodes only the first frequency band. */
- int seg, chset_i;
-
- /* Coding parameters for each channel set. */
- struct coding_params {
- int seg_type;
- int rice_code_flag[16];
- int pancAuxABIT[16];
- int pancABIT0[16]; /* Not sure what this is */
- int pancABIT[16]; /* Not sure what this is */
- int nSamplPart0[16];
- } param_state[16];
-
- GetBitContext *gb = &s->xll_navi.gb;
- int *history;
-
- /* Layout: First the sample buffer for one segment per channel,
- * followed by history buffers of DCA_XLL_AORDER_MAX samples for
- * each channel. */
- av_fast_malloc(&s->xll_sample_buf, &s->xll_sample_buf_size,
- (s->xll_smpl_in_seg + DCA_XLL_AORDER_MAX) *
- s->xll_channels * sizeof(*s->xll_sample_buf));
- if (!s->xll_sample_buf)
- return AVERROR(ENOMEM);
-
- history = s->xll_sample_buf + s->xll_smpl_in_seg * s->xll_channels;
-
- for (seg = 0; seg < s->xll_segments; seg++) {
- unsigned in_channel;
-
- for (chset_i = in_channel = 0; chset_i < s->xll_nch_sets; chset_i++) {
- /* The spec isn't very explicit, but I think the NAVI sizes are in bytes. */
- int end_pos = get_bits_count(gb) +
- 8 * s->xll_navi.chset_size[0][seg][chset_i];
- int i, j;
- struct coding_params *params = &param_state[chset_i];
- /* I think this flag means that we should keep seg_type and
- * other parameters from the previous segment. */
- int use_seg_state_code_param;
- XllChSetSubHeader *chset = &s->xll_chsets[chset_i];
- if (in_channel >= s->avctx->channels)
- /* FIXME: Could go directly to next segment */
- goto next_chset;
-
- if (s->avctx->sample_rate != chset->sampling_frequency) {
- av_log(s->avctx, AV_LOG_WARNING,
- "XLL: unexpected chset sample rate %d, expected %d\n",
- chset->sampling_frequency, s->avctx->sample_rate);
- goto next_chset;
- }
- if (seg != 0)
- use_seg_state_code_param = get_bits(gb, 1);
- else
- use_seg_state_code_param = 0;
-
- if (!use_seg_state_code_param) {
- int num_param_sets, i;
- unsigned bits4ABIT;
-
- params->seg_type = get_bits(gb, 1);
- num_param_sets = params->seg_type ? 1 : chset->channels;
-
- if (chset->bit_width > 16) {
- bits4ABIT = 5;
- } else {
- if (chset->bit_width > 8)
- bits4ABIT = 4;
- else
- bits4ABIT = 3;
- if (s->xll_nch_sets > 1)
- bits4ABIT++;
- }
-
- for (i = 0; i < num_param_sets; i++) {
- params->rice_code_flag[i] = get_bits(gb, 1);
- if (!params->seg_type && params->rice_code_flag[i] && get_bits(gb, 1))
- params->pancAuxABIT[i] = get_bits(gb, bits4ABIT) + 1;
- else
- params->pancAuxABIT[i] = 0;
- }
-
- for (i = 0; i < num_param_sets; i++) {
- if (!seg) {
- /* Parameters for part 1 */
- params->pancABIT0[i] = get_bits(gb, bits4ABIT);
- if (params->rice_code_flag[i] == 0 && params->pancABIT0[i] > 0)
- /* For linear code */
- params->pancABIT0[i]++;
-
- /* NOTE: In the spec, not indexed by band??? */
- if (params->seg_type == 0)
- params->nSamplPart0[i] = chset->adapt_order[0][i];
- else
- params->nSamplPart0[i] = chset->adapt_order_max[0];
- } else
- params->nSamplPart0[i] = 0;
-
- /* Parameters for part 2 */
- params->pancABIT[i] = get_bits(gb, bits4ABIT);
- if (params->rice_code_flag[i] == 0 && params->pancABIT[i] > 0)
- /* For linear code */
- params->pancABIT[i]++;
- }
- }
- for (i = 0; i < chset->channels; i++) {
- int param_index = params->seg_type ? 0 : i;
- int part0 = params->nSamplPart0[param_index];
- int bits = part0 ? params->pancABIT0[param_index] : 0;
- int *sample_buf = s->xll_sample_buf +
- (in_channel + i) * s->xll_smpl_in_seg;
-
- if (!params->rice_code_flag[param_index]) {
- /* Linear code */
- if (bits)
- for (j = 0; j < part0; j++)
- sample_buf[j] = get_bits_sm(gb, bits);
- else
- memset(sample_buf, 0, part0 * sizeof(sample_buf[0]));
-
- /* Second part */
- bits = params->pancABIT[param_index];
- if (bits)
- for (j = part0; j < s->xll_smpl_in_seg; j++)
- sample_buf[j] = get_bits_sm(gb, bits);
- else
- memset(sample_buf + part0, 0,
- (s->xll_smpl_in_seg - part0) * sizeof(sample_buf[0]));
- } else {
- int aux_bits = params->pancAuxABIT[param_index];
-
- for (j = 0; j < part0; j++) {
- /* FIXME: Is this identical to Golomb code? */
- int t = get_unary(gb, 1, 33) << bits;
- /* FIXME: Could move this test outside of the loop, for efficiency. */
- if (bits)
- t |= get_bits(gb, bits);
- sample_buf[j] = (t & 1) ? -(t >> 1) - 1 : (t >> 1);
- }
-
- /* Second part */
- bits = params->pancABIT[param_index];
-
- /* Follow the spec's suggestion of using the
- * buffer also to store the hybrid-rice flags. */
- memset(sample_buf + part0, 0,
- (s->xll_smpl_in_seg - part0) * sizeof(sample_buf[0]));
-
- if (aux_bits > 0) {
- /* For hybrid rice encoding, some samples are linearly
- * coded. According to the spec, "nBits4SamplLoci" bits
- * are used for each index, but this value is not
- * defined. I guess we should use log2(xll_smpl_in_seg)
- * bits. */
- int count = get_bits(gb, s->xll_log_smpl_in_seg);
- av_log(s->avctx, AV_LOG_DEBUG, "aux count %d (bits %d)\n",
- count, s->xll_log_smpl_in_seg);
-
- for (j = 0; j < count; j++)
- sample_buf[get_bits(gb, s->xll_log_smpl_in_seg)] = 1;
- }
- for (j = part0; j < s->xll_smpl_in_seg; j++) {
- if (!sample_buf[j]) {
- int t = get_unary(gb, 1, 33);
- if (bits)
- t = (t << bits) | get_bits(gb, bits);
- sample_buf[j] = (t & 1) ? -(t >> 1) - 1 : (t >> 1);
- } else
- sample_buf[j] = get_bits_sm(gb, aux_bits);
- }
- }
- }
-
- for (i = 0; i < chset->channels; i++) {
- unsigned adapt_order = chset->adapt_order[0][i];
- int *sample_buf = s->xll_sample_buf +
- (in_channel + i) * s->xll_smpl_in_seg;
- int *prev = history + (in_channel + i) * DCA_XLL_AORDER_MAX;
-
- if (!adapt_order) {
- unsigned order;
- for (order = chset->fixed_order[0][i]; order > 0; order--) {
- unsigned j;
- for (j = 1; j < s->xll_smpl_in_seg; j++)
- sample_buf[j] += sample_buf[j - 1];
- }
- } else
- /* Inverse adaptive prediction, in place. */
- dca_xll_inv_adapt_pred(sample_buf, s->xll_smpl_in_seg,
- adapt_order, seg ? prev : NULL,
- chset->lpc_refl_coeffs_q_ind[0][i]);
- memcpy(prev, sample_buf + s->xll_smpl_in_seg - DCA_XLL_AORDER_MAX,
- DCA_XLL_AORDER_MAX * sizeof(*prev));
- }
- for (i = 1; i < chset->channels; i += 2) {
- int coeff = chset->pw_ch_pairs_coeffs[0][i / 2];
- if (coeff != 0) {
- int *sample_buf = s->xll_sample_buf +
- (in_channel + i) * s->xll_smpl_in_seg;
- int *prev = sample_buf - s->xll_smpl_in_seg;
- unsigned j;
- for (j = 0; j < s->xll_smpl_in_seg; j++)
- /* Shift is unspecified, but should apparently be 3. */
- sample_buf[j] += ((int64_t) coeff * prev[j] + 4) >> 3;
- }
- }
-
- if (s->xll_scalable_lsb) {
- int lsb_start = end_pos - 8 * chset->lsb_fsize[0] -
- 8 * (s->xll_banddata_crc & 2);
- int done;
- i = get_bits_count(gb);
- if (i > lsb_start) {
- av_log(s->avctx, AV_LOG_ERROR,
- "chset data lsb exceeds NAVI size, end_pos %d, lsb_start %d, pos %d\n",
- end_pos, lsb_start, i);
- return AVERROR_INVALIDDATA;
- }
- if (i < lsb_start)
- skip_bits_long(gb, lsb_start - i);
-
- for (i = done = 0; i < chset->channels; i++) {
- int bits = chset->scalable_lsbs[0][i];
- if (bits > 0) {
- /* The channel reordering is conceptually done
- * before adding the lsb:s, so we need to do
- * the inverse permutation here. */
- unsigned pi = chset->orig_chan_order_inv[0][i];
- int *sample_buf = s->xll_sample_buf +
- (in_channel + pi) * s->xll_smpl_in_seg;
- int adj = chset->bit_width_adj_per_ch[0][i];
- int msb_shift = bits;
- unsigned j;
-
- if (adj > 0)
- msb_shift += adj - 1;
-
- for (j = 0; j < s->xll_smpl_in_seg; j++)
- sample_buf[j] = (sample_buf[j] << msb_shift) +
- (get_bits(gb, bits) << adj);
-
- done += bits * s->xll_smpl_in_seg;
- }
- }
- if (done > 8 * chset->lsb_fsize[0]) {
- av_log(s->avctx, AV_LOG_ERROR,
- "chset lsb exceeds lsb_size\n");
- return AVERROR_INVALIDDATA;
- }
- }
-
- /* Store output. */
- for (i = 0; i < chset->channels; i++) {
- int *sample_buf = s->xll_sample_buf +
- (in_channel + i) * s->xll_smpl_in_seg;
- int shift = 1 - chset->bit_resolution;
- int out_channel = chset->orig_chan_order[0][i];
- float *out;
-
- /* XLL uses the channel order C, L, R, and we want L,
- * R, C. FIXME: Generalize. */
- if (chset->ch_mask_enabled &&
- (chset->ch_mask & 7) == 7 && out_channel < 3)
- out_channel = out_channel ? out_channel - 1 : 2;
-
- out_channel += in_channel;
- if (out_channel >= s->avctx->channels)
- continue;
-
- out = (float *) frame->extended_data[out_channel];
- out += seg * s->xll_smpl_in_seg;
-
- /* NOTE: A one bit means residual encoding is *not* used. */
- if ((chset->residual_encode >> i) & 1) {
- /* Replace channel samples.
- * FIXME: Most likely not the right thing to do. */
- for (j = 0; j < s->xll_smpl_in_seg; j++)
- out[j] = ldexpf(sample_buf[j], shift);
- } else {
- /* Add residual signal to core channel */
- for (j = 0; j < s->xll_smpl_in_seg; j++)
- out[j] += ldexpf(sample_buf[j], shift);
- }
- }
-
- if (chset->downmix_coeff_code_embedded &&
- !chset->primary_ch_set && chset->hier_chset) {
- /* Undo hierarchical downmix of earlier channels. */
- unsigned mix_channel;
- for (mix_channel = 0; mix_channel < in_channel; mix_channel++) {
- float *mix_buf;
- const int *col;
- float coeff;
- unsigned row;
- /* Similar channel reorder C, L, R vs L, R, C reorder. */
- if (chset->ch_mask_enabled &&
- (chset->ch_mask & 7) == 7 && mix_channel < 3)
- mix_buf = (float *) frame->extended_data[mix_channel ? mix_channel - 1 : 2];
- else
- mix_buf = (float *) frame->extended_data[mix_channel];
-
- mix_buf += seg * s->xll_smpl_in_seg;
- col = &chset->downmix_coeffs[mix_channel * (chset->channels + 1)];
-
- /* Scale */
- coeff = ldexpf(col[0], -16);
- for (j = 0; j < s->xll_smpl_in_seg; j++)
- mix_buf[j] *= coeff;
-
- for (row = 0;
- row < chset->channels && in_channel + row < s->avctx->channels;
- row++)
- if (col[row + 1]) {
- const float *new_channel =
- (const float *) frame->extended_data[in_channel + row];
- new_channel += seg * s->xll_smpl_in_seg;
- coeff = ldexpf(col[row + 1], -15);
- for (j = 0; j < s->xll_smpl_in_seg; j++)
- mix_buf[j] -= coeff * new_channel[j];
- }
- }
- }
-
-next_chset:
- in_channel += chset->channels;
- /* Skip to next channel set using the NAVI info. */
- i = get_bits_count(gb);
- if (i > end_pos) {
- av_log(s->avctx, AV_LOG_ERROR,
- "chset data exceeds NAVI size\n");
- return AVERROR_INVALIDDATA;
- }
- if (i < end_pos)
- skip_bits_long(gb, end_pos - i);
- }
- }
- return 0;
-}
diff --git a/libavcodec/dcadata.c b/libavcodec/dcadata.c
index af2c75b74e..0d0c218838 100644
--- a/libavcodec/dcadata.c
+++ b/libavcodec/dcadata.c
@@ -22,7 +22,6 @@
#include <stdint.h>
-#include "libavutil/channel_layout.h"
#include "libavutil/mem.h"
#include "dca.h"
@@ -7509,76 +7508,6 @@ DECLARE_ALIGNED(16, const float, ff_dca_lfe_fir_128)[256] = {
};
#undef SCALE
-
-#define SCALE(c) ((float)(c) / (256.0f * 32768.0f * 8388608.0f))
-DECLARE_ALIGNED(16, const float, ff_dca_lfe_xll_fir_64)[256] = {
- SCALE( 6103), SCALE( 52170), SCALE(-558064), SCALE(1592440),
- SCALE(6290049), SCALE(1502534), SCALE(-546669), SCALE( 53047),
- SCALE( 1930), SCALE( 51089), SCALE(-568920), SCALE(1683709),
- SCALE(6286575), SCALE(1414057), SCALE(-534782), SCALE( 53729),
- SCALE( 2228), SCALE( 49794), SCALE(-579194), SCALE(1776276),
- SCALE(6279634), SCALE(1327070), SCALE(-522445), SCALE( 54228),
- SCALE( 2552), SCALE( 48275), SCALE(-588839), SCALE(1870070),
- SCALE(6269231), SCALE(1241632), SCALE(-509702), SCALE( 54550),
- SCALE( 2904), SCALE( 46523), SCALE(-597808), SCALE(1965017),
- SCALE(6255380), SCALE(1157798), SCALE(-496595), SCALE( 54708),
- SCALE( 3287), SCALE( 44529), SCALE(-606054), SCALE(2061044),
- SCALE(6238099), SCALE(1075621), SCALE(-483164), SCALE( 54710),
- SCALE( 3704), SCALE( 42282), SCALE(-613529), SCALE(2158071),
- SCALE(6217408), SCALE( 995149), SCALE(-469451), SCALE( 54566),
- SCALE( 4152), SCALE( 39774), SCALE(-620186), SCALE(2256019),
- SCALE(6193332), SCALE( 916430), SCALE(-455494), SCALE( 54285),
- SCALE( 4631), SCALE( 36995), SCALE(-625976), SCALE(2354805),
- SCALE(6165900), SCALE( 839507), SCALE(-441330), SCALE( 53876),
- SCALE( 5139), SCALE( 33937), SCALE(-630850), SCALE(2454343),
- SCALE(6135146), SCALE( 764419), SCALE(-426998), SCALE( 53348),
- SCALE( 5682), SCALE( 30591), SCALE(-634759), SCALE(2554547),
- SCALE(6101107), SCALE( 691203), SCALE(-412531), SCALE( 52711),
- SCALE( 6264), SCALE( 26948), SCALE(-637655), SCALE(2655326),
- SCALE(6063824), SCALE( 619894), SCALE(-397966), SCALE( 51972),
- SCALE( 6886), SCALE( 23001), SCALE(-639488), SCALE(2756591),
- SCALE(6023343), SCALE( 550521), SCALE(-383335), SCALE( 51140),
- SCALE( 7531), SCALE( 18741), SCALE(-640210), SCALE(2858248),
- SCALE(5979711), SCALE( 483113), SCALE(-368671), SCALE( 50224),
- SCALE( 8230), SCALE( 14162), SCALE(-639772), SCALE(2960201),
- SCALE(5932981), SCALE( 417692), SCALE(-354003), SCALE( 49231),
- SCALE( 8959), SCALE( 9257), SCALE(-638125), SCALE(3062355),
- SCALE(5883210), SCALE( 354281), SCALE(-339362), SCALE( 48168),
- SCALE( 9727), SCALE( 4018), SCALE(-635222), SCALE(3164612),
- SCALE(5830457), SCALE( 292897), SCALE(-324777), SCALE( 47044),
- SCALE( 10535), SCALE( -1558), SCALE(-631014), SCALE(3266872),
- SCALE(5774785), SCALE( 233555), SCALE(-310273), SCALE( 45866),
- SCALE( 11381), SCALE( -7480), SCALE(-625455), SCALE(3369035),
- SCALE(5716260), SCALE( 176267), SCALE(-295877), SCALE( 44640),
- SCALE( 12267), SCALE( -13750), SCALE(-618499), SCALE(3471000),
- SCALE(5654952), SCALE( 121042), SCALE(-281613), SCALE( 43373),
- SCALE( 13190), SCALE( -20372), SCALE(-610098), SCALE(3572664),
- SCALE(5590933), SCALE( 67886), SCALE(-267505), SCALE( 42072),
- SCALE( 14152), SCALE( -27352), SCALE(-600209), SCALE(3673924),
- SCALE(5524280), SCALE( 16800), SCALE(-253574), SCALE( 40743),
- SCALE( 15153), SCALE( -34691), SCALE(-588788), SCALE(3774676),
- SCALE(5455069), SCALE( -32214), SCALE(-239840), SCALE( 39391),
- SCALE( 16192), SCALE( -42390), SCALE(-575791), SCALE(3874816),
- SCALE(5383383), SCALE( -79159), SCALE(-226323), SCALE( 38022),
- SCALE( 17267), SCALE( -50453), SCALE(-561178), SCALE(3974239),
- SCALE(5309305), SCALE(-124041), SCALE(-213041), SCALE( 36642),
- SCALE( 18377), SCALE( -58879), SCALE(-544906), SCALE(4072841),
- SCALE(5232922), SCALE(-166869), SCALE(-200010), SCALE( 35256),
- SCALE( 19525), SCALE( -67667), SCALE(-526937), SCALE(4170517),
- SCALE(5154321), SCALE(-207653), SCALE(-187246), SCALE( 33866),
- SCALE( 20704), SCALE( -76817), SCALE(-507233), SCALE(4267162),
- SCALE(5073593), SCALE(-246406), SCALE(-174764), SCALE( 32480),
- SCALE( 21915), SCALE( -86327), SCALE(-485757), SCALE(4362672),
- SCALE(4990831), SCALE(-283146), SCALE(-162575), SCALE( 31101),
- SCALE( 23157), SCALE( -96193), SCALE(-462476), SCALE(4456942),
- SCALE(4906129), SCALE(-317890), SCALE(-150692), SCALE( 29732),
- SCALE( 24426), SCALE(-106412), SCALE(-437356), SCALE(4549871),
- SCALE(4819584), SCALE(-350658), SCALE(-139125), SCALE( 28376),
- SCALE( 25721), SCALE(-116977), SCALE(-410365), SCALE(4641355),
- SCALE(4731293), SCALE(-381475), SCALE(-127884), SCALE( 27038),
-};
-#undef SCALE
-
DECLARE_ALIGNED(16, const float, ff_dca_fir_64bands)[1024] = {
/* Bank 0 */
-7.1279389866041690e-8, -7.0950903150874990e-8,
@@ -8178,220 +8107,11 @@ const uint32_t ff_dca_inv_dmixtable[FF_DCA_INV_DMIXTABLE_SIZE] = {
65536,
};
-const float ff_dca_default_coeffs[10][6][2] = {
- { { 0.707107, 0.707107 }, { 0.000000, 0.000000 }, }, // A [LFE]
- { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 }, }, // A + B (dual mono) [LFE]
- { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 }, }, // L + R (stereo) [LFE]
- { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 }, }, // (L+R) + (L-R) (sum-difference) [LFE]
- { { 1.000000, 0.000000 }, { 0.000000, 1.000000 }, { 0.000000, 0.000000 }, }, // LT + RT (left and right total) [LFE]
- { { 0.501187, 0.501187 }, { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.000000, 0.000000 }, }, // C + L + R [LFE]
- { { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.501187 }, { 0.000000, 0.000000 }, }, // L + R + S [LFE]
- { { 0.501187, 0.501187 }, { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.501187 }, { 0.000000, 0.000000 }, }, // C + L + R + S [LFE]
- { { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.000000 }, { 0.000000, 0.501187 }, { 0.000000, 0.000000 }, }, // L + R + SL + SR [LFE]
- { { 0.501187, 0.501187 }, { 0.707107, 0.000000 }, { 0.000000, 0.707107 }, { 0.501187, 0.000000 }, { 0.000000, 0.501187 }, { 0.000000, 0.000000 }, }, // C + L + R + SL + SR [LFE]
-};
-
const int32_t ff_dca_sampling_freqs[16] = {
8000, 16000, 32000, 64000, 128000, 22050, 44100, 88200,
176400, 352800, 12000, 24000, 48000, 96000, 192000, 384000,
};
-/* downmix coeffs
- *
- * TABLE 9
- * ______________________________________
- * Down-mix coefficients for 8-channel source
- * audio (5 + 3 format)
- * lt
- * cen- rt lt ctr rt
- * lt ter ctr center
- * rt srd srd srd
- * ______________________________________
- * 1 0.71 0.74 1.0 0.71 0.71 0.58 0.58 0.58
- * 2 left 1.0 0.89 0.71 0.46 0.71 0.50
- * rt 0.45 0.71 0.89 1.0 0.50 0.71
- * 3 lt 1.0 0.89 0.71 0.45
- * rt 0.45 0.71 0.89 1.0
- * srd 0.71 0.71 0.71
- * 4 lt 1.0 0.89 0.71 0.45
- * rt 0.45 0.71 0.89 1.0
- * lt srd 1.0 0.71
- * rt srd 0.71 0.71
- * 4 lt 1.0 0.5
- * ctr 0.87 1.0 0.87
- * rt 0.5 1.0
- * srd 0.71 0.71 0.71
- * 5 lt 1.0 0.5
- * ctr 0.87 1.0 0.87
- * rt 0.5 1.0
- * lt srd 1.0 0.71
- * rt srd 0.71 1.0
- * 6 lt 1.0 0.5
- * lt ctr 0.87 0.71
- * rt ctr 0.71 0.87
- * rt 0.5 1.0
- * lt srd 1.0 0.71
- * rt srd 0.71 1.0
- * 6 lt 1.0 0.5
- * ctr 0.86 1.0 0.86
- * rt 0.5 1.0
- * lt srd 1.0
- * ctr srd 1.0
- * rt srd 1.0
- * 7 lt 1.0
- * lt ctr 1.0
- * ctr 1.0
- * rt ctr 1.0
- * rt 1.0
- * lt srd 1.0 0.71
- * rt srd 0.71 1.0
- * 7 lt 1.0 0.5
- * lt ctr 0.87 0.71
- * rt ctr 0.71 0.87
- * rt 0.5 1.0
- * lt srd 1.0
- * ctr srd 1.0
- * rt srd 1.0
- * 8 lt 1.0 0.5
- * lt ctr 0.87 0.71
- * rt ctr 0.71 0.87
- * rt 0.5 1.0
- * lt 1 srd 0.87 0.35
- * lt 2 srd 0.5 0.61
- * rt 2 srd 0.61 0.50
- * rt 2 srd 0.35 0.87
- *
- * Generation of Lt Rt
- *
- * In the case when the playback system has analog or digital surround
- * multi-channel capability, a down matrix from 5, 4, or 3 channel to
- * Lt Rt may be desirable. In the case when the number of decoded audio
- * channels exceeds 5, 4 or 3 respectively a first stage down mix to 5,
- * 4 or 3 chs should be used as described above.
- *
- * The down matrixing equations for 5-channel source audio to a
- * two-channel Lt Rt playback system are given by:
- *
- * Left = left + 0.7 * center - 0.7 * (lt surround + rt surround)
- *
- * Right = right + 0.7 * center + 0.7 * (lt surround + rt surround)
- *
- * Embedded mixing to 2-channel
- *
- * One concern arising from the proliferation of multi-channel audio
- * systems is that most home systems presently have only two channel
- * playback capability. To accommodate this a fixed 2-channel down
- * matrix processes is commonly used following the multi-channel
- * decoding stage. However, for music only applications the image
- * quality etc. of the down matrixed signal may not match that of an
- * equivalent stereo recording found on CD.
- *
- * The concept of embedded mixing is to allow the producer to
- * dynamically specify the matrixing coefficients within the audio
- * frame itself. In this way the stereo down mix at the decoder may be
- * better matched to a 2-channel playback environment.
- *
- * CHS*2, 7-bit down mix indexes (MCOEFFS) are transmitted along with
- * the multi-channel audio once in every frame. The indexes are
- * converted to attenuation factors using a 7 bit LUT. The 2-ch down
- * mix equations are as follows,
- *
- * Left Ch = sum (MCOEFF[n] * Ch[n]) for n=1, CHS
- *
- * Right Ch = sum (MCOEFF[n + CHS] * Ch[n]) for n=1, CHS
- *
- * where Ch(n) represents the subband samples in the (n)th audio channel.
- */
-
-const uint32_t ff_dca_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 */
-};
-
-/* -1 are reserved or unknown */
-const int ff_dca_ext_audio_descr_mask[8] = {
- DCA_EXT_XCH,
- -1,
- DCA_EXT_X96,
- DCA_EXT_XCH | DCA_EXT_X96,
- -1,
- -1,
- DCA_EXT_XXCH,
- -1,
-};
-
-/* 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
- */
-const uint64_t ff_dca_core_channel_layout[16] = {
- 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
-};
-
const int8_t ff_dca_lfe_index[16] = {
1, 2, 2, 2, 2, 3, 2, 3, 2, 3, 2, 3, 1, 3, 2, 3
};
@@ -8415,25 +8135,6 @@ const int8_t ff_dca_channel_reorder_lfe[16][9] = {
{ 4, 2, 5, 0, 1, 6, 8, 7, -1 },
};
-const int8_t ff_dca_channel_reorder_lfe_xch[16][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 },
-};
-
const int8_t ff_dca_channel_reorder_nolfe[16][9] = {
{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
@@ -8453,25 +8154,6 @@ const int8_t ff_dca_channel_reorder_nolfe[16][9] = {
{ 3, 2, 4, 0, 1, 5, 7, 6, -1 },
};
-const int8_t ff_dca_channel_reorder_nolfe_xch[16][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 },
-};
-
const uint16_t ff_dca_vlc_offs[63] = {
0, 512, 640, 768, 1282, 1794, 2436, 3080, 3770, 4454, 5364,
5372, 5380, 5388, 5392, 5396, 5412, 5420, 5428, 5460, 5492, 5508,
diff --git a/libavcodec/dcadata.h b/libavcodec/dcadata.h
index 262c37efb9..3d318fe6ec 100644
--- a/libavcodec/dcadata.h
+++ b/libavcodec/dcadata.h
@@ -45,7 +45,6 @@ extern const float ff_dca_fir_32bands_nonperfect[512];
extern const float ff_dca_lfe_fir_64[256];
extern const float ff_dca_lfe_fir_128[256];
-extern const float ff_dca_lfe_xll_fir_64[256];
extern const float ff_dca_fir_64bands[1024];
#define FF_DCA_DMIXTABLE_SIZE 242
@@ -54,21 +53,12 @@ extern const float ff_dca_fir_64bands[1024];
extern const uint16_t ff_dca_dmixtable[FF_DCA_DMIXTABLE_SIZE];
extern const uint32_t ff_dca_inv_dmixtable[FF_DCA_INV_DMIXTABLE_SIZE];
-extern const float ff_dca_default_coeffs[10][6][2];
-
-extern const uint32_t ff_dca_map_xxch_to_native[28];
-extern const int ff_dca_ext_audio_descr_mask[8];
-
-extern const uint64_t ff_dca_core_channel_layout[16];
-
extern const int32_t ff_dca_sampling_freqs[16];
extern const int8_t ff_dca_lfe_index[16];
extern const int8_t ff_dca_channel_reorder_lfe[16][9];
-extern const int8_t ff_dca_channel_reorder_lfe_xch[16][9];
extern const int8_t ff_dca_channel_reorder_nolfe[16][9];
-extern const int8_t ff_dca_channel_reorder_nolfe_xch[16][9];
extern const uint16_t ff_dca_vlc_offs[63];
diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c
deleted file mode 100644
index 6b8d02d59a..0000000000
--- a/libavcodec/dcadec.c
+++ /dev/null
@@ -1,2067 +0,0 @@
-/*
- * DCA compatible decoder
- * Copyright (C) 2004 Gildas Bazin
- * Copyright (C) 2004 Benjamin Zores
- * Copyright (C) 2006 Benjamin Larsson
- * Copyright (C) 2007 Konstantin Shishkov
- * Copyright (C) 2012 Paul B Mahol
- * Copyright (C) 2014 Niels Möller
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include <math.h>
-#include <stddef.h>
-#include <stdio.h>
-
-#include "libavutil/attributes.h"
-#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/opt.h"
-#include "libavutil/samplefmt.h"
-
-#include "avcodec.h"
-#include "dca.h"
-#include "dca_syncwords.h"
-#include "dcadata.h"
-#include "dcadsp.h"
-#include "dcahuff.h"
-#include "fft.h"
-#include "fmtconvert.h"
-#include "get_bits.h"
-#include "internal.h"
-#include "mathops.h"
-#include "profiles.h"
-#include "synth_filter.h"
-
-#if ARCH_ARM
-# include "arm/dca.h"
-#endif
-
-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
-};
-
-
-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,
-};
-
-#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_NSYNCAUX 0x9A1105A0
-
-/** Bit allocation */
-typedef struct BitAlloc {
- 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;
-}
-
-static float dca_dmix_code(unsigned code);
-
-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[ff_dca_vlc_offs[i]];
- dca_bitalloc_index.vlc[i].table_allocated = ff_dca_vlc_offs[i + 1] - ff_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[ff_dca_vlc_offs[i + 5]];
- dca_scalefactor.vlc[i].table_allocated = ff_dca_vlc_offs[i + 6] - ff_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[ff_dca_vlc_offs[i + 10]];
- dca_tmode.vlc[i].table_allocated = ff_dca_vlc_offs[i + 11] - ff_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[ff_dca_vlc_offs[c]];
- dca_smpl_bitalloc[i + 1].vlc[j].table_allocated = ff_dca_vlc_offs[c + 1] - ff_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 uint8_t adj_table[4] = { 16, 18, 20, 23 };
- 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 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;
- if (xxch && nchans >= 3) {
- av_log(s->avctx, AV_LOG_ERROR, "nchans %d is too large\n", nchans);
- return AVERROR_INVALIDDATA;
- } else if (nchans + base_channel > DCA_PRIM_CHANNELS_MAX) {
- av_log(s->avctx, AV_LOG_ERROR, "channel sum %d + %d is too large\n", nchans, base_channel);
- return AVERROR_INVALIDDATA;
- }
-
- s->audio_header.total_channels = nchans + base_channel;
- s->audio_header.prim_channels = s->audio_header.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);
- coeff = get_bits(&s->gb, 6);
-
- if (coeff<1 || coeff>61) {
- av_log(s->avctx, AV_LOG_ERROR, "6bit coeff %d is out of range\n", coeff);
- return AVERROR_INVALIDDATA;
- }
-
- scale_factor = -1.0f / dca_dmix_code((coeff<<2)-3);
-
- s->xxch_dmix_sf[s->xxch_chset] = scale_factor;
-
- for (i = base_channel; i < s->audio_header.prim_channels; i++) {
- mask[i] = get_bits(&s->gb, s->xxch_nbits_spk_mask);
- }
-
- for (j = base_channel; j < s->audio_header.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);
- ichan = dca_xxch2index(s, 1 << i);
- if ((coeff&63)<1 || (coeff&63)>61) {
- av_log(s->avctx, AV_LOG_ERROR, "7bit coeff %d is out of range\n", coeff);
- return AVERROR_INVALIDDATA;
- }
- s->xxch_dmix_coeff[j][ichan] = dca_dmix_code((coeff<<2)-3);
- }
- }
- }
- }
- }
-
- if (s->audio_header.prim_channels > DCA_PRIM_CHANNELS_MAX)
- s->audio_header.prim_channels = DCA_PRIM_CHANNELS_MAX;
-
- for (i = base_channel; i < s->audio_header.prim_channels; i++) {
- s->audio_header.subband_activity[i] = get_bits(&s->gb, 5) + 2;
- if (s->audio_header.subband_activity[i] > DCA_SUBBANDS)
- s->audio_header.subband_activity[i] = DCA_SUBBANDS;
- }
- for (i = base_channel; i < s->audio_header.prim_channels; i++) {
- s->audio_header.vq_start_subband[i] = get_bits(&s->gb, 5) + 1;
- if (s->audio_header.vq_start_subband[i] > DCA_SUBBANDS)
- s->audio_header.vq_start_subband[i] = DCA_SUBBANDS;
- }
- get_array(&s->gb, s->audio_header.joint_intensity + base_channel,
- s->audio_header.prim_channels - base_channel, 3);
- get_array(&s->gb, s->audio_header.transient_huffman + base_channel,
- s->audio_header.prim_channels - base_channel, 2);
- get_array(&s->gb, s->audio_header.scalefactor_huffman + base_channel,
- s->audio_header.prim_channels - base_channel, 3);
- get_array(&s->gb, s->audio_header.bitalloc_huffman + base_channel,
- s->audio_header.prim_channels - base_channel, 3);
-
- /* Get codebooks quantization indexes */
- if (!base_channel)
- memset(s->audio_header.quant_index_huffman, 0, sizeof(s->audio_header.quant_index_huffman));
- for (j = 1; j < 11; j++)
- for (i = base_channel; i < s->audio_header.prim_channels; i++)
- s->audio_header.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->audio_header.prim_channels; i++)
- s->audio_header.scalefactor_adj[i][j] = 16;
-
- for (j = 1; j < 11; j++)
- for (i = base_channel; i < s->audio_header.prim_channels; i++)
- if (s->audio_header.quant_index_huffman[i][j] < thr[j])
- s->audio_header.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;
-
- 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 = ff_dca_bit_rates[s->bit_rate_index];
- if (!s->bit_rate)
- return AVERROR_INVALIDDATA;
-
- skip_bits1(&s->gb); // always 0 (reserved, cf. ETSI TS 102 114 V1.4.1)
- 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 > 2) {
- s->lfe = 0;
- av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE value: %d\n", s->lfe);
- return AVERROR_INVALIDDATA;
- }
-
- /* 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;
-
- /* Primary audio coding header */
- s->audio_header.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;
- if (block_index + s->subsubframes[s->current_subframe] > (s->sample_blocks / SAMPLES_PER_SUBBAND)) {
- s->subsubframes[s->current_subframe] = 1;
- return AVERROR_INVALIDDATA;
- }
- s->partial_samples[s->current_subframe] = get_bits(&s->gb, 3);
- }
-
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- for (k = 0; k < s->audio_header.subband_activity[j]; k++)
- s->dca_chan[j].prediction_mode[k] = get_bits(&s->gb, 1);
- }
-
- /* Get prediction codebook */
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
- if (s->dca_chan[j].prediction_mode[k] > 0) {
- /* (Prediction coefficient VQ address) */
- s->dca_chan[j].prediction_vq[k] = get_bits(&s->gb, 12);
- }
- }
- }
-
- /* Bit allocation index */
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- for (k = 0; k < s->audio_header.vq_start_subband[j]; k++) {
- if (s->audio_header.bitalloc_huffman[j] == 6)
- s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 5);
- else if (s->audio_header.bitalloc_huffman[j] == 5)
- s->dca_chan[j].bitalloc[k] = get_bits(&s->gb, 4);
- else if (s->audio_header.bitalloc_huffman[j] == 7) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid bit allocation index\n");
- return AVERROR_INVALIDDATA;
- } else {
- s->dca_chan[j].bitalloc[k] =
- get_bitalloc(&s->gb, &dca_bitalloc_index, s->audio_header.bitalloc_huffman[j]);
- }
-
- if (s->dca_chan[j].bitalloc[k] > 26) {
- ff_dlog(s->avctx, "bitalloc index [%i][%i] too big (%i)\n",
- j, k, s->dca_chan[j].bitalloc[k]);
- return AVERROR_INVALIDDATA;
- }
- }
- }
-
- /* Transition mode */
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
- s->dca_chan[j].transition_mode[k] = 0;
- if (s->subsubframes[s->current_subframe] > 1 &&
- k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].bitalloc[k] > 0) {
- s->dca_chan[j].transition_mode[k] =
- get_bitalloc(&s->gb, &dca_tmode, s->audio_header.transient_huffman[j]);
- }
- }
- }
-
- if (get_bits_left(&s->gb) < 0)
- return AVERROR_INVALIDDATA;
-
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- const uint32_t *scale_table;
- int scale_sum, log_size;
-
- memset(s->dca_chan[j].scale_factor, 0,
- s->audio_header.subband_activity[j] * sizeof(s->dca_chan[j].scale_factor[0][0]) * 2);
-
- if (s->audio_header.scalefactor_huffman[j] == 6) {
- scale_table = ff_dca_scale_factor_quant7;
- log_size = 7;
- } else {
- scale_table = ff_dca_scale_factor_quant6;
- log_size = 6;
- }
-
- /* When huffman coded, only the difference is encoded */
- scale_sum = 0;
-
- for (k = 0; k < s->audio_header.subband_activity[j]; k++) {
- if (k >= s->audio_header.vq_start_subband[j] || s->dca_chan[j].bitalloc[k] > 0) {
- scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
- s->dca_chan[j].scale_factor[k][0] = scale_table[scale_sum];
- }
-
- if (k < s->audio_header.vq_start_subband[j] && s->dca_chan[j].transition_mode[k]) {
- /* Get second scale factor */
- scale_sum = get_scale(&s->gb, s->audio_header.scalefactor_huffman[j], scale_sum, log_size);
- s->dca_chan[j].scale_factor[k][1] = scale_table[scale_sum];
- }
- }
- }
-
- /* Joint subband scale factor codebook select */
- for (j = base_channel; j < s->audio_header.prim_channels; j++) {
- /* Transmitted only if joint subband coding enabled */
- if (s->audio_header.joint_intensity[j] > 0)
- s->dca_chan[j].joint_huff = 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->audio_header.prim_channels; j++) {
- int source_channel;
-
- /* Transmitted only if joint subband coding enabled */
- if (s->audio_header.joint_intensity[j] > 0) {
- int scale = 0;
- source_channel = s->audio_header.joint_intensity[j] - 1;
-
- /* When huffman coded, only the difference is encoded
- * (is this valid as well for joint scales ???) */
-
- for (k = s->audio_header.subband_activity[j];
- k < s->audio_header.subband_activity[source_channel]; k++) {
- scale = get_scale(&s->gb, s->dca_chan[j].joint_huff, 64 /* bias */, 7);
- s->dca_chan[j].joint_scale_factor[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;
- }
- }
- }
-
- /* 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->audio_header.prim_channels; j++)
- for (k = s->audio_header.vq_start_subband[j]; k < s->audio_header.subband_activity[j]; k++)
- /* 1 vector -> 32 samples */
- s->dca_chan[j].high_freq_vq[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) {
- avpriv_request_sample(s->avctx, "LFEScaleIndex larger than 127");
- return AVERROR_INVALIDDATA;
- }
- s->lfe_scale_factor = ff_dca_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;
- }
-
- return 0;
-}
-
-static void qmf_32_subbands(DCAContext *s, int chans,
- float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], float *samples_out,
- float scale)
-{
- const float *prCoeff;
-
- int sb_act = s->audio_header.subband_activity[chans];
-
- scale *= sqrt(1 / 8.0);
-
- /* Select filter */
- if (!s->multirate_inter) /* Non-perfect reconstruction */
- prCoeff = ff_dca_fir_32bands_nonperfect;
- else /* Perfect reconstruction */
- prCoeff = ff_dca_fir_32bands_perfect;
-
- s->dcadsp.qmf_32_subbands(samples_in, sb_act, &s->synth, &s->imdct,
- s->dca_chan[chans].subband_fir_hist,
- &s->dca_chan[chans].hist_index,
- s->dca_chan[chans].subband_fir_noidea, prCoeff,
- samples_out, s->raXin, scale);
-}
-
-static QMF64_table *qmf64_precompute(void)
-{
- unsigned i, j;
- QMF64_table *table = av_malloc(sizeof(*table));
- if (!table)
- return NULL;
-
- for (i = 0; i < 32; i++)
- for (j = 0; j < 32; j++)
- table->dct4_coeff[i][j] = cos((2 * i + 1) * (2 * j + 1) * M_PI / 128);
- for (i = 0; i < 32; i++)
- for (j = 0; j < 32; j++)
- table->dct2_coeff[i][j] = cos((2 * i + 1) * j * M_PI / 64);
-
- /* FIXME: Is the factor 0.125 = 1/8 right? */
- for (i = 0; i < 32; i++)
- table->rcos[i] = 0.125 / cos((2 * i + 1) * M_PI / 256);
- for (i = 0; i < 32; i++)
- table->rsin[i] = -0.125 / sin((2 * i + 1) * M_PI / 256);
-
- return table;
-}
-
-/* FIXME: Totally unoptimized. Based on the reference code and
- * http://multimedia.cx/mirror/dca-transform.pdf, with guessed tweaks
- * for doubling the size. */
-static void qmf_64_subbands(DCAContext *s, int chans,
- float samples_in[DCA_SUBBANDS_X96K][SAMPLES_PER_SUBBAND],
- float *samples_out, float scale)
-{
- float raXin[64];
- float A[32], B[32];
- float *raX = s->dca_chan[chans].subband_fir_hist;
- float *raZ = s->dca_chan[chans].subband_fir_noidea;
- unsigned i, j, k, subindex;
-
- for (i = s->audio_header.subband_activity[chans]; i < DCA_SUBBANDS_X96K; i++)
- raXin[i] = 0.0;
- for (subindex = 0; subindex < SAMPLES_PER_SUBBAND; subindex++) {
- for (i = 0; i < s->audio_header.subband_activity[chans]; i++)
- raXin[i] = samples_in[i][subindex];
-
- for (k = 0; k < 32; k++) {
- A[k] = 0.0;
- for (i = 0; i < 32; i++)
- A[k] += (raXin[2 * i] + raXin[2 * i + 1]) * s->qmf64_table->dct4_coeff[k][i];
- }
- for (k = 0; k < 32; k++) {
- B[k] = raXin[0] * s->qmf64_table->dct2_coeff[k][0];
- for (i = 1; i < 32; i++)
- B[k] += (raXin[2 * i] + raXin[2 * i - 1]) * s->qmf64_table->dct2_coeff[k][i];
- }
- for (k = 0; k < 32; k++) {
- raX[k] = s->qmf64_table->rcos[k] * (A[k] + B[k]);
- raX[63 - k] = s->qmf64_table->rsin[k] * (A[k] - B[k]);
- }
-
- for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
- float out = raZ[i];
- for (j = 0; j < 1024; j += 128)
- out += ff_dca_fir_64bands[j + i] * (raX[j + i] - raX[j + 63 - i]);
- *samples_out++ = out * scale;
- }
-
- for (i = 0; i < DCA_SUBBANDS_X96K; i++) {
- float hist = 0.0;
- for (j = 0; j < 1024; j += 128)
- hist += ff_dca_fir_64bands[64 + j + i] * (-raX[i + j] - raX[j + 63 - i]);
-
- raZ[i] = hist;
- }
-
- /* FIXME: Make buffer circular, to avoid this move. */
- memmove(raX + 64, raX, (1024 - 64) * sizeof(*raX));
- }
-}
-
-static void lfe_interpolation_fir(DCAContext *s, const float *samples_in,
- float *samples_out)
-{
- /* 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 idx;
- const float *prCoeff;
- int deciindex;
-
- /* Select decimation filter */
- if (s->lfe == 1) {
- idx = 1;
- prCoeff = ff_dca_lfe_fir_128;
- } else {
- idx = 0;
- if (s->exss_ext_mask & DCA_EXT_EXSS_XLL)
- prCoeff = ff_dca_lfe_xll_fir_64;
- else
- prCoeff = ff_dca_lfe_fir_64;
- }
- /* Interpolation */
- for (deciindex = 0; deciindex < 2 * s->lfe; deciindex++) {
- s->dcadsp.lfe_fir[idx](samples_out, samples_in, prCoeff);
- samples_in++;
- samples_out += 2 * 32 * (1 + idx);
- }
-}
-
-/* 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 lfe_present,
- float coef[DCA_PRIM_CHANNELS_MAX + 1][2],
- const int8_t *channel_mapping)
-{
- int c, l, r, sl, sr, s;
- int i;
- float t, u, v;
-
- switch (srcfmt) {
- case DCA_MONO:
- case DCA_4F2R:
- av_log(NULL, AV_LOG_ERROR, "Not implemented!\n");
- break;
- case DCA_CHANNEL:
- case DCA_STEREO:
- case DCA_STEREO_TOTAL:
- case DCA_STEREO_SUMDIFF:
- 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;
- }
- if (lfe_present) {
- int lf_buf = ff_dca_lfe_index[srcfmt];
- int lf_idx = ff_dca_channels[srcfmt];
- for (i = 0; i < 256; i++) {
- samples[0][i] += samples[lf_buf][i] * coef[lf_idx][0];
- samples[1][i] += samples[lf_buf][i] * coef[lf_idx][1];
- }
- }
-}
-
-#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, int32_t *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, int32_t *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 };
-
-static int dca_subsubframe(DCAContext *s, int base_channel, int block_index)
-{
- int k, l;
- int subsubframe = s->current_subsubframe;
- const uint32_t *quant_step_table;
-
- /*
- * Audio data
- */
-
- /* Select quantization step size table */
- if (s->bit_rate_index == 0x1f)
- quant_step_table = ff_dca_lossless_quant;
- else
- quant_step_table = ff_dca_lossy_quant;
-
- for (k = base_channel; k < s->audio_header.prim_channels; k++) {
- int32_t (*subband_samples)[8] = s->dca_chan[k].subband_samples[block_index];
-
- if (get_bits_left(&s->gb) < 0)
- return AVERROR_INVALIDDATA;
-
- for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
- int m;
-
- /* Select the mid-tread linear quantizer */
- int abits = s->dca_chan[k].bitalloc[l];
-
- uint32_t quant_step_size = quant_step_table[abits];
-
- /*
- * Extract bits from the bit stream
- */
- if (!abits)
- memset(subband_samples[l], 0, SAMPLES_PER_SUBBAND *
- sizeof(subband_samples[l][0]));
- else {
- uint32_t rscale;
- /* Deal with transients */
- int sfi = s->dca_chan[k].transition_mode[l] &&
- subsubframe >= s->dca_chan[k].transition_mode[l];
- /* Determine quantization index code book and its type.
- Select quantization index code book */
- int sel = s->audio_header.quant_index_huffman[k][abits];
-
- rscale = (s->dca_chan[k].scale_factor[l][sfi] *
- s->audio_header.scalefactor_adj[k][sel] + 8) >> 4;
-
- 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, subband_samples[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 < SAMPLES_PER_SUBBAND; m++)
- subband_samples[l][m] = get_sbits(&s->gb, abits - 3);
- }
- } else {
- /* Huffman coded */
- for (m = 0; m < SAMPLES_PER_SUBBAND; m++)
- subband_samples[l][m] = get_bitalloc(&s->gb,
- &dca_smpl_bitalloc[abits], sel);
- }
- s->dcadsp.dequantize(subband_samples[l], quant_step_size, rscale);
- }
- }
-
- for (l = 0; l < s->audio_header.vq_start_subband[k]; l++) {
- int m;
- /*
- * Inverse ADPCM if in prediction mode
- */
- if (s->dca_chan[k].prediction_mode[l]) {
- int n;
- if (s->predictor_history)
- subband_samples[l][0] += (ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][3] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][1] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][2] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][2] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][1] +
- ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][3] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][0]) +
- (1 << 12) >> 13;
- for (m = 1; m < SAMPLES_PER_SUBBAND; m++) {
- int64_t sum = ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][0] *
- (int64_t)subband_samples[l][m - 1];
- for (n = 2; n <= 4; n++)
- if (m >= n)
- sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
- (int64_t)subband_samples[l][m - n];
- else if (s->predictor_history)
- sum += ff_dca_adpcm_vb[s->dca_chan[k].prediction_vq[l]][n - 1] *
- (int64_t)s->dca_chan[k].subband_samples_hist[l][m - n + 4];
- subband_samples[l][m] += (int32_t)(sum + (1 << 12) >> 13);
- }
- }
-
- }
- /* Backup predictor history for adpcm */
- for (l = 0; l < DCA_SUBBANDS; l++)
- AV_COPY128(s->dca_chan[k].subband_samples_hist[l], &subband_samples[l][4]);
-
-
- /*
- * Decode VQ encoded high frequencies
- */
- if (s->audio_header.subband_activity[k] > s->audio_header.vq_start_subband[k]) {
- if (!(s->debug_flag & 0x01)) {
- av_log(s->avctx, AV_LOG_DEBUG,
- "Stream with high frequencies VQ coding\n");
- s->debug_flag |= 0x01;
- }
-
- s->dcadsp.decode_hf(subband_samples, s->dca_chan[k].high_freq_vq,
- ff_dca_high_freq_vq,
- subsubframe * SAMPLES_PER_SUBBAND,
- s->dca_chan[k].scale_factor,
- s->audio_header.vq_start_subband[k],
- s->audio_header.subband_activity[k]);
- }
- }
-
- /* Check for DSYNC after subsubframe */
- if (s->aspf || subsubframe == s->subsubframes[s->current_subframe] - 1) {
- if (get_bits(&s->gb, 16) != 0xFFFF) {
- av_log(s->avctx, AV_LOG_ERROR, "Didn't get subframe DSYNC\n");
- return AVERROR_INVALIDDATA;
- }
- }
-
- return 0;
-}
-
-static int dca_filter_channels(DCAContext *s, int block_index, int upsample)
-{
- int k;
-
- if (upsample) {
- LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS_X96K], [SAMPLES_PER_SUBBAND]);
-
- if (!s->qmf64_table) {
- s->qmf64_table = qmf64_precompute();
- if (!s->qmf64_table)
- return AVERROR(ENOMEM);
- }
-
- /* 64 subbands QMF */
- for (k = 0; k < s->audio_header.prim_channels; k++) {
- int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
- s->dca_chan[k].subband_samples[block_index];
-
- s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
- DCA_SUBBANDS_X96K * SAMPLES_PER_SUBBAND);
-
- if (s->channel_order_tab[k] >= 0)
- qmf_64_subbands(s, k, samples,
- s->samples_chanptr[s->channel_order_tab[k]],
- /* Upsampling needs a factor 2 here. */
- M_SQRT2 / 32768.0);
- }
- } else {
- /* 32 subbands QMF */
- LOCAL_ALIGNED(32, float, samples, [DCA_SUBBANDS], [SAMPLES_PER_SUBBAND]);
-
- for (k = 0; k < s->audio_header.prim_channels; k++) {
- int32_t (*subband_samples)[SAMPLES_PER_SUBBAND] =
- s->dca_chan[k].subband_samples[block_index];
-
- s->fmt_conv.int32_to_float(samples[0], subband_samples[0],
- DCA_SUBBANDS * SAMPLES_PER_SUBBAND);
-
- if (s->channel_order_tab[k] >= 0)
- qmf_32_subbands(s, k, samples,
- s->samples_chanptr[s->channel_order_tab[k]],
- M_SQRT1_2 / 32768.0);
- }
- }
-
- /* Generate LFE samples for this subsubframe FIXME!!! */
- if (s->lfe) {
- float *samples = s->samples_chanptr[s->lfe_index];
- lfe_interpolation_fir(s,
- s->lfe_data + 2 * s->lfe * (block_index + 4),
- samples);
- if (upsample) {
- unsigned i;
- /* Should apply the filter in Table 6-11 when upsampling. For
- * now, just duplicate. */
- for (i = 255; i > 0; i--) {
- samples[2 * i] =
- samples[2 * i + 1] = samples[i];
- }
- samples[1] = samples[0];
- }
- }
-
- /* FIXME: This downmixing is probably broken with upsample.
- * Probably totally broken also with XLL in general. */
- /* Downmixing to Stereo */
- if (s->audio_header.prim_channels + !!s->lfe > 2 &&
- s->avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
- dca_downmix(s->samples_chanptr, s->amode, !!s->lfe, s->downmix_coef,
- s->channel_order_tab);
- }
-
- return 0;
-}
-
-static int dca_subframe_footer(DCAContext *s, int base_channel)
-{
- int in, out, aux_data_count, aux_data_end, reserved;
- uint32_t nsyncaux;
-
- /*
- * 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);
-
- // align (32-bit)
- skip_bits_long(&s->gb, (-get_bits_count(&s->gb)) & 31);
-
- aux_data_end = 8 * aux_data_count + get_bits_count(&s->gb);
-
- if ((nsyncaux = get_bits_long(&s->gb, 32)) != DCA_NSYNCAUX) {
- av_log(s->avctx, AV_LOG_ERROR, "nSYNCAUX mismatch %#"PRIx32"\n",
- nsyncaux);
- return AVERROR_INVALIDDATA;
- }
-
- if (get_bits1(&s->gb)) { // bAUXTimeStampFlag
- avpriv_request_sample(s->avctx,
- "Auxiliary Decode Time Stamp Flag");
- // align (4-bit)
- skip_bits(&s->gb, (-get_bits_count(&s->gb)) & 4);
- // 44 bits: nMSByte (8), nMarker (4), nLSByte (28), nMarker (4)
- skip_bits_long(&s->gb, 44);
- }
-
- if ((s->core_downmix = get_bits1(&s->gb))) {
- int am = get_bits(&s->gb, 3);
- switch (am) {
- case 0:
- s->core_downmix_amode = DCA_MONO;
- break;
- case 1:
- s->core_downmix_amode = DCA_STEREO;
- break;
- case 2:
- s->core_downmix_amode = DCA_STEREO_TOTAL;
- break;
- case 3:
- s->core_downmix_amode = DCA_3F;
- break;
- case 4:
- s->core_downmix_amode = DCA_2F1R;
- break;
- case 5:
- s->core_downmix_amode = DCA_2F2R;
- break;
- case 6:
- s->core_downmix_amode = DCA_3F1R;
- break;
- default:
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid mode %d for embedded downmix coefficients\n",
- am);
- return AVERROR_INVALIDDATA;
- }
- for (out = 0; out < ff_dca_channels[s->core_downmix_amode]; out++) {
- for (in = 0; in < s->audio_header.prim_channels + !!s->lfe; in++) {
- uint16_t tmp = get_bits(&s->gb, 9);
- if ((tmp & 0xFF) > 241) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid downmix coefficient code %"PRIu16"\n",
- tmp);
- return AVERROR_INVALIDDATA;
- }
- s->core_downmix_codes[in][out] = tmp;
- }
- }
- }
-
- align_get_bits(&s->gb); // byte align
- skip_bits(&s->gb, 16); // nAUXCRC16
-
- /*
- * additional data (reserved, cf. ETSI TS 102 114 V1.4.1)
- *
- * Note: don't check for overreads, aux_data_count can't be trusted.
- */
- if ((reserved = (aux_data_end - get_bits_count(&s->gb))) > 0) {
- avpriv_request_sample(s->avctx,
- "Core auxiliary data reserved content");
- skip_bits_long(&s->gb, reserved);
- }
- }
-
- if (s->crc_present && 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->audio_header.subframes) {
- av_log(s->avctx, AV_LOG_DEBUG, "check failed: %i>%i",
- s->current_subframe, s->audio_header.subframes);
- return AVERROR_INVALIDDATA;
- }
-
- if (!s->current_subsubframe) {
- /* Read subframe header */
- if ((ret = dca_subframe_header(s, base_channel, block_index)))
- return ret;
- }
-
- /* Read subsubframe */
- 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->audio_header.subframes) {
- /* Read subframe footer */
- if ((ret = dca_subframe_footer(s, base_channel)))
- return ret;
- }
-
- return 0;
-}
-
-int ff_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;
- if (active_bands[i][j] > DCA_SUBBANDS) {
- av_log(s->avctx, AV_LOG_ERROR, "too many active subbands (%d)\n", active_bands[i][j]);
- return AVERROR_INVALIDDATA;
- }
- }
- }
-
- /* 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->audio_header.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;
- int scale_table_size;
-
- if (s->audio_header.scalefactor_huffman[chan_base+i] == 6) {
- scale_table = ff_dca_scale_factor_quant7;
- scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
- } else {
- scale_table = ff_dca_scale_factor_quant6;
- scale_table_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
- }
-
- nbits = anctemp[i];
-
- for(j = 0; j < active_bands[chset][i]; j++) {
- if(abits_high[i][j] > 0) {
- int index = get_bits(&s->gb, nbits);
- if (index >= scale_table_size) {
- av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
- return AVERROR_INVALIDDATA;
- }
- scale_table_high[i][j][0] = scale_table[index];
-
- if(xbr_tmode && s->dca_chan[i].transition_mode[j]) {
- int index = get_bits(&s->gb, nbits);
- if (index >= scale_table_size) {
- av_log(s->avctx, AV_LOG_ERROR, "scale table index %d invalid\n", index);
- return AVERROR_INVALIDDATA;
- }
- scale_table_high[i][j][1] = scale_table[index];
- }
- }
- }
- }
- }
-
- /* 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 uint32_t quant_step_size = ff_dca_lossless_quant[xbr_abits];
- const int sfi = xbr_tmode && s->dca_chan[i].transition_mode[j] && subsubframe >= s->dca_chan[i].transition_mode[j];
- const uint32_t rscale = scale_table_high[i][j][sfi];
- int32_t *subband_samples = s->dca_chan[chan_base+i].subband_samples[k][j];
- int32_t block[SAMPLES_PER_SUBBAND];
-
- if(xbr_abits <= 0)
- continue;
-
- if(xbr_abits > 7) {
- get_array(&s->gb, block, SAMPLES_PER_SUBBAND, 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 */
- s->dcadsp.dequantize(block, quant_step_size, rscale);
- for(l = 0; l < SAMPLES_PER_SUBBAND; l++)
- subband_samples[l] += block[l];
- }
- }
-
- /* 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 */
-int ff_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->audio_header.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;
-}
-
-static float dca_dmix_code(unsigned code)
-{
- int sign = (code >> 8) - 1;
- code &= 0xff;
- return ((ff_dca_dmixtable[code] ^ sign) - sign) * (1.0 / (1 << 15));
-}
-
-static int scan_for_extensions(AVCodecContext *avctx)
-{
- DCAContext *s = avctx->priv_data;
- int core_ss_end, ret = 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);
- int i;
-
- switch (bits) {
- case DCA_SYNCWORD_XCH: {
- int ext_amode, xch_fsize;
-
- s->xch_base_channel = s->audio_header.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) {
- avpriv_request_sample(avctx, "XCh with fewer than 2 base channels");
- 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 DCA_SYNCWORD_XXCH:
- /* 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;
- ff_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_SYNCWORD_SUBSTREAM)
- ff_dca_exss_parse_header(s);
-
- return ret;
-}
-
-static int set_channel_layout(AVCodecContext *avctx, int *channels, int num_core_channels)
-{
- DCAContext *s = avctx->priv_data;
- int i, j, chset, mask;
- int channel_layout, channel_mask;
- int posn, lavc;
-
- /* 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)) {
- /* xxx should also do MA extensions */
- if (s->amode < 16) {
- avctx->channel_layout = ff_dca_core_channel_layout[s->amode];
-
- if (s->audio_header.prim_channels + !!s->lfe > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
- /*
- * Neither the core's auxiliary data nor our default tables contain
- * downmix coefficients for the additional channel coded in the XCh
- * extension, so when we're doing a Stereo downmix, don't decode it.
- */
- s->xch_disable = 1;
- }
-
- if (s->xch_present && !s->xch_disable) {
- if (avctx->channel_layout & AV_CH_BACK_CENTER) {
- avpriv_request_sample(avctx, "XCh with Back center channel");
- return AVERROR_INVALIDDATA;
- }
- avctx->channel_layout |= AV_CH_BACK_CENTER;
- if (s->lfe) {
- avctx->channel_layout |= AV_CH_LOW_FREQUENCY;
- s->channel_order_tab = ff_dca_channel_reorder_lfe_xch[s->amode];
- } else {
- s->channel_order_tab = ff_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 = ff_dca_channel_reorder_lfe[s->amode];
- } else
- s->channel_order_tab = ff_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 (num_core_channels + !!s->lfe > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
- *channels = 2;
- s->output = s->audio_header.prim_channels == 2 ? s->amode : 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 = ff_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;
-
- {
- *channels = s->audio_header.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 |= ff_dca_map_xxch_to_native[i];
- }
- }
-
- /* make sure that we have managed to get equivalent 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: Inconsistent 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 = ff_dca_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;
- }
-
- return 0;
-}
-
-/**
- * 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 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 channels, full_channels;
- float scale;
- int achan;
- int chset;
- int mask;
- int j, k;
- int endch;
- int upsample = 0;
-
- s->exss_ext_mask = 0;
- s->xch_present = 0;
-
- s->dca_buffer_size = AVERROR_INVALIDDATA;
- for (i = 0; i < buf_size - 3 && s->dca_buffer_size == AVERROR_INVALIDDATA; i++)
- s->dca_buffer_size = avpriv_dca_convert_bitstream(buf + i, buf_size - i, 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;
- }
-
- 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;
-
- s->profile = FF_PROFILE_DTS;
-
- for (i = 0; i < (s->sample_blocks / SAMPLES_PER_SUBBAND); 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->audio_header.prim_channels;
-
- if (s->audio_header.prim_channels + !!s->lfe > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO) {
- /* Stereo downmix coefficients
- *
- * The decoder can only downmix to 2-channel, so we need to ensure
- * embedded downmix coefficients are actually targeting 2-channel.
- */
- if (s->core_downmix && (s->core_downmix_amode == DCA_STEREO ||
- s->core_downmix_amode == DCA_STEREO_TOTAL)) {
- for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- /* Range checked earlier */
- s->downmix_coef[i][0] = dca_dmix_code(s->core_downmix_codes[i][0]);
- s->downmix_coef[i][1] = dca_dmix_code(s->core_downmix_codes[i][1]);
- }
- s->output = s->core_downmix_amode;
- } else {
- int am = s->amode & DCA_CHANNEL_MASK;
- if (am >= FF_ARRAY_ELEMS(ff_dca_default_coeffs)) {
- av_log(s->avctx, AV_LOG_ERROR,
- "Invalid channel mode %d\n", am);
- return AVERROR_INVALIDDATA;
- }
- if (num_core_channels + !!s->lfe >
- FF_ARRAY_ELEMS(ff_dca_default_coeffs[0])) {
- avpriv_request_sample(s->avctx, "Downmixing %d channels",
- s->audio_header.prim_channels + !!s->lfe);
- return AVERROR_PATCHWELCOME;
- }
- for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- s->downmix_coef[i][0] = ff_dca_default_coeffs[am][i][0];
- s->downmix_coef[i][1] = ff_dca_default_coeffs[am][i][1];
- }
- }
- ff_dlog(s->avctx, "Stereo downmix coeffs:\n");
- for (i = 0; i < num_core_channels + !!s->lfe; i++) {
- ff_dlog(s->avctx, "L, input channel %d = %f\n", i,
- s->downmix_coef[i][0]);
- ff_dlog(s->avctx, "R, input channel %d = %f\n", i,
- s->downmix_coef[i][1]);
- }
- ff_dlog(s->avctx, "\n");
- }
-
- if (s->ext_coding)
- s->core_ext_mask = ff_dca_ext_audio_descr_mask[s->ext_descr];
- else
- s->core_ext_mask = 0;
-
- ret = scan_for_extensions(avctx);
-
- avctx->profile = s->profile;
-
- full_channels = channels = s->audio_header.prim_channels + !!s->lfe;
-
- ret = set_channel_layout(avctx, &channels, num_core_channels);
- if (ret < 0)
- return ret;
-
- /* get output buffer */
- frame->nb_samples = 256 * (s->sample_blocks / SAMPLES_PER_SUBBAND);
- if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
- int xll_nb_samples = s->xll_segments * s->xll_smpl_in_seg;
- /* Check for invalid/unsupported conditions first */
- if (s->xll_residual_channels > channels) {
- av_log(s->avctx, AV_LOG_WARNING,
- "DCA: too many residual channels (%d, core channels %d). Disabling XLL\n",
- s->xll_residual_channels, channels);
- s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
- } else if (xll_nb_samples != frame->nb_samples &&
- 2 * frame->nb_samples != xll_nb_samples) {
- av_log(s->avctx, AV_LOG_WARNING,
- "DCA: unsupported upsampling (%d XLL samples, %d core samples). Disabling XLL\n",
- xll_nb_samples, frame->nb_samples);
- s->exss_ext_mask &= ~DCA_EXT_EXSS_XLL;
- } else {
- if (2 * frame->nb_samples == xll_nb_samples) {
- av_log(s->avctx, AV_LOG_INFO,
- "XLL: upsampling core channels by a factor of 2\n");
- upsample = 1;
-
- frame->nb_samples = xll_nb_samples;
- // FIXME: Is it good enough to copy from the first channel set?
- avctx->sample_rate = s->xll_chsets[0].sampling_frequency;
- }
- /* If downmixing to stereo, don't decode additional channels.
- * FIXME: Using the xch_disable flag for this doesn't seem right. */
- if (!s->xch_disable)
- channels = s->xll_channels;
- }
- }
-
- 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;
- }
-
- /* FIXME: This is an ugly hack, to just revert to the default
- * layout if we have additional channels. Need to convert the XLL
- * channel masks to ffmpeg channel_layout mask. */
- if (av_get_channel_layout_nb_channels(avctx->channel_layout) != avctx->channels)
- avctx->channel_layout = 0;
-
- if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
- 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 / SAMPLES_PER_SUBBAND); i++) {
- int ch;
- unsigned block = upsample ? 512 : 256;
- for (ch = 0; ch < channels; ch++)
- s->samples_chanptr[ch] = samples_flt[ch] + i * block;
- for (; ch < full_channels; ch++)
- s->samples_chanptr[ch] = s->extra_channels[ch - channels] + i * block;
-
- dca_filter_channels(s, i, upsample);
-
- /* 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 / SAMPLES_PER_SUBBAND);
- for (i = 0; i < 2 * s->lfe * 4; i++)
- s->lfe_data[i] = s->lfe_data[i + lfe_samples];
-
- if (s->exss_ext_mask & DCA_EXT_EXSS_XLL) {
- ret = ff_dca_xll_decode_audio(s, frame);
- if (ret < 0)
- return ret;
- }
- /* AVMatrixEncoding
- *
- * DCA_STEREO_TOTAL (Lt/Rt) is equivalent to Dolby Surround */
- ret = ff_side_data_update_matrix_encoding(frame,
- (s->output & ~DCA_LFE) == DCA_STEREO_TOTAL ?
- AV_MATRIX_ENCODING_DOLBY : AV_MATRIX_ENCODING_NONE);
- if (ret < 0)
- return ret;
-
- if ( avctx->profile != FF_PROFILE_DTS_HD_MA
- && avctx->profile != FF_PROFILE_DTS_HD_HRA)
- avctx->bit_rate = s->bit_rate;
- *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();
-
- s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
- if (!s->fdsp)
- return AVERROR(ENOMEM);
-
- 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 > 2 &&
- avctx->request_channel_layout == AV_CH_LAYOUT_STEREO)
- avctx->channels = 2;
-
- 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);
- av_freep(&s->fdsp);
- av_freep(&s->xll_sample_buf);
- av_freep(&s->qmf64_table);
- return 0;
-}
-
-static const AVOption options[] = {
- { "disable_xch", "disable decoding of the XCh extension", offsetof(DCAContext, xch_disable), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
- { "disable_xll", "disable decoding of the XLL extension", offsetof(DCAContext, xll_disable), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM },
- { NULL },
-};
-
-static const AVClass dca_decoder_class = {
- .class_name = "DCA decoder",
- .item_name = av_default_item_name,
- .option = options,
- .version = LIBAVUTIL_VERSION_INT,
- .category = AV_CLASS_CATEGORY_DECODER,
-};
-
-AVCodec ff_dca_decoder = {
- .name = "dca",
- .long_name = NULL_IF_CONFIG_SMALL("DCA (DTS Coherent Acoustics)"),
- .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,
- .capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
- .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
- AV_SAMPLE_FMT_NONE },
- .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles),
- .priv_class = &dca_decoder_class,
-};
diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c
deleted file mode 100644
index 32b149d09c..0000000000
--- a/libavcodec/dcadsp.c
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2004 Gildas Bazin
- * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "config.h"
-
-#include "libavutil/attributes.h"
-#include "libavutil/intreadwrite.h"
-
-#include "dcadsp.h"
-#include "dcamath.h"
-
-static void decode_hf_c(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
- const int32_t vq_num[DCA_SUBBANDS],
- const int8_t hf_vq[1024][32], intptr_t vq_offset,
- int32_t scale[DCA_SUBBANDS][2],
- intptr_t start, intptr_t end)
-{
- int i, j;
-
- for (j = start; j < end; j++) {
- const int8_t *ptr = &hf_vq[vq_num[j]][vq_offset];
- for (i = 0; i < 8; i++)
- dst[j][i] = ptr[i] * scale[j][0] + 8 >> 4;
- }
-}
-
-static inline void dca_lfe_fir(float *out, const float *in, const float *coefs,
- int decifactor)
-{
- float *out2 = out + 2 * decifactor - 1;
- int num_coeffs = 256 / decifactor;
- int j, k;
-
- /* One decimated sample generates 2*decifactor interpolated ones */
- for (k = 0; k < decifactor; k++) {
- float v0 = 0.0;
- float v1 = 0.0;
- for (j = 0; j < num_coeffs; j++, coefs++) {
- v0 += in[-j] * *coefs;
- v1 += in[j + 1 - num_coeffs] * *coefs;
- }
- *out++ = v0;
- *out2-- = v1;
- }
-}
-
-static void dca_qmf_32_subbands(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
- SynthFilterContext *synth, FFTContext *imdct,
- float synth_buf_ptr[512],
- int *synth_buf_offset, float synth_buf2[32],
- const float window[512], float *samples_out,
- float raXin[32], float scale)
-{
- int i;
- int subindex;
-
- for (i = sb_act; i < 32; i++)
- 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(&raXin[i], v);
- }
-
- synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
- synth_buf2, window, samples_out, raXin,
- scale);
- samples_out += 32;
- }
-}
-
-static void dequantize_c(int32_t *samples, uint32_t step_size, uint32_t scale)
-{
- int64_t step = (int64_t)step_size * scale;
- int shift, i;
- int32_t step_scale;
-
- if (step > (1 << 23))
- shift = av_log2(step >> 23) + 1;
- else
- shift = 0;
- step_scale = (int32_t)(step >> shift);
-
- for (i = 0; i < SAMPLES_PER_SUBBAND; i++)
- samples[i] = dca_clip23(dca_norm((int64_t)samples[i] * step_scale, 22 - shift));
-}
-
-static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
-{
- dca_lfe_fir(out, in, coefs, 32);
-}
-
-static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
-{
- dca_lfe_fir(out, in, coefs, 64);
-}
-
-av_cold void ff_dcadsp_init(DCADSPContext *s)
-{
- s->lfe_fir[0] = dca_lfe_fir0_c;
- s->lfe_fir[1] = dca_lfe_fir1_c;
- s->qmf_32_subbands = dca_qmf_32_subbands;
- s->decode_hf = decode_hf_c;
- s->dequantize = dequantize_c;
-
- if (ARCH_AARCH64)
- ff_dcadsp_init_aarch64(s);
- if (ARCH_ARM)
- ff_dcadsp_init_arm(s);
- if (ARCH_X86)
- ff_dcadsp_init_x86(s);
-}
diff --git a/libavcodec/dcadsp.h b/libavcodec/dcadsp.h
deleted file mode 100644
index 8c8db854a4..0000000000
--- a/libavcodec/dcadsp.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_DCADSP_H
-#define AVCODEC_DCADSP_H
-
-#include "avfft.h"
-#include "synth_filter.h"
-
-#define DCA_SUBBANDS_X96K 64
-#define DCA_SUBBANDS 64
-#define SAMPLES_PER_SUBBAND 8 // number of samples per subband per subsubframe
-
-
-typedef struct DCADSPContext {
- void (*lfe_fir[2])(float *out, const float *in, const float *coefs);
- void (*qmf_32_subbands)(float samples_in[DCA_SUBBANDS][SAMPLES_PER_SUBBAND], int sb_act,
- SynthFilterContext *synth, FFTContext *imdct,
- float synth_buf_ptr[512],
- int *synth_buf_offset, float synth_buf2[32],
- const float window[512], float *samples_out,
- float raXin[32], float scale);
- void (*decode_hf)(int32_t dst[DCA_SUBBANDS][SAMPLES_PER_SUBBAND],
- const int32_t vq_num[DCA_SUBBANDS],
- const int8_t hf_vq[1024][32], intptr_t vq_offset,
- int32_t scale[DCA_SUBBANDS][2],
- intptr_t start, intptr_t end);
- void (*dequantize)(int32_t *samples, uint32_t step_size, uint32_t scale);
-} DCADSPContext;
-
-void ff_dcadsp_init(DCADSPContext *s);
-void ff_dcadsp_init_aarch64(DCADSPContext *s);
-void ff_dcadsp_init_arm(DCADSPContext *s);
-void ff_dcadsp_init_x86(DCADSPContext *s);
-
-#endif /* AVCODEC_DCADSP_H */
diff --git a/libavcodec/dcamath.h b/libavcodec/dcamath.h
deleted file mode 100644
index a8a41427f7..0000000000
--- a/libavcodec/dcamath.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#ifndef AVCODEC_DCAMATH_H
-#define AVCODEC_DCAMATH_H
-
-#include "libavutil/common.h"
-
-
-// clip a signed integer into the (-2^23), (2^23-1) range
-static inline int dca_clip23(int a)
-{
- return av_clip_intp2(a, 23);
-}
-
-static inline int32_t dca_norm(int64_t a, int bits)
-{
- if (bits > 0)
- return (int32_t)((a + (INT64_C(1) << (bits - 1))) >> bits);
- else
- return (int32_t)a;
-}
-
-static inline int64_t dca_round(int64_t a, int bits)
-{
- if (bits > 0)
- return (a + (INT64_C(1) << (bits - 1))) & ~((INT64_C(1) << bits) - 1);
- else
- return a;
-}
-
-#endif /* AVCODEC_DCAMATH_H */
diff --git a/libavcodec/x86/Makefile b/libavcodec/x86/Makefile
index bcb42332a0..eec98cb7a0 100644
--- a/libavcodec/x86/Makefile
+++ b/libavcodec/x86/Makefile
@@ -44,8 +44,7 @@ OBJS-$(CONFIG_ADPCM_G722_ENCODER) += x86/g722dsp_init.o
OBJS-$(CONFIG_ALAC_DECODER) += x86/alacdsp_init.o
OBJS-$(CONFIG_APNG_DECODER) += x86/pngdsp_init.o
OBJS-$(CONFIG_CAVS_DECODER) += x86/cavsdsp.o
-OBJS-$(CONFIG_DCA_DECODER) += x86/dcadsp_init.o \
- x86/synth_filter_init.o
+#OBJS-$(CONFIG_DCA_DECODER) += x86/synth_filter_init.o
OBJS-$(CONFIG_DNXHD_ENCODER) += x86/dnxhdenc_init.o
OBJS-$(CONFIG_HEVC_DECODER) += x86/hevcdsp_init.o
OBJS-$(CONFIG_JPEG2000_DECODER) += x86/jpeg2000dsp_init.o
@@ -133,8 +132,7 @@ YASM-OBJS-$(CONFIG_ADPCM_G722_DECODER) += x86/g722dsp.o
YASM-OBJS-$(CONFIG_ADPCM_G722_ENCODER) += x86/g722dsp.o
YASM-OBJS-$(CONFIG_ALAC_DECODER) += x86/alacdsp.o
YASM-OBJS-$(CONFIG_APNG_DECODER) += x86/pngdsp.o
-YASM-OBJS-$(CONFIG_DCA_DECODER) += x86/dcadsp.o \
- x86/synth_filter.o
+#YASM-OBJS-$(CONFIG_DCA_DECODER) += x86/synth_filter.o
YASM-OBJS-$(CONFIG_DIRAC_DECODER) += x86/diracdsp_mmx.o x86/diracdsp_yasm.o \
x86/dwt_yasm.o
YASM-OBJS-$(CONFIG_DNXHD_ENCODER) += x86/dnxhdenc.o
diff --git a/libavcodec/x86/dcadsp.asm b/libavcodec/x86/dcadsp.asm
deleted file mode 100644
index 55e73bcc29..0000000000
--- a/libavcodec/x86/dcadsp.asm
+++ /dev/null
@@ -1,123 +0,0 @@
-;******************************************************************************
-;* SSE-optimized functions for the DCA decoder
-;* Copyright (C) 2012-2014 Christophe Gisquet <christophe.gisquet@gmail.com>
-;*
-;* This file is part of FFmpeg.
-;*
-;* FFmpeg is free software; you can redistribute it and/or
-;* modify it under the terms of the GNU Lesser General Public
-;* License as published by the Free Software Foundation; either
-;* version 2.1 of the License, or (at your option) any later version.
-;*
-;* FFmpeg is distributed in the hope that it will be useful,
-;* but WITHOUT ANY WARRANTY; without even the implied warranty of
-;* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
-;* Lesser General Public License for more details.
-;*
-;* You should have received a copy of the GNU Lesser General Public
-;* License along with FFmpeg; if not, write to the Free Software
-;* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
-;******************************************************************************
-
-%include "libavutil/x86/x86util.asm"
-
-SECTION_RODATA
-pf_inv16: times 4 dd 0x3D800000 ; 1/16
-
-SECTION .text
-
-; %1=v0/v1 %2=in1 %3=in2
-%macro FIR_LOOP 2-3
-.loop%1:
-%define va m1
-%define vb m2
-%if %1
-%define OFFSET 0
-%else
-%define OFFSET NUM_COEF*count
-%endif
-; for v0, incrementing and for v1, decrementing
- mova va, [cf0q + OFFSET]
- mova vb, [cf0q + OFFSET + 4*NUM_COEF]
-%if %0 == 3
- mova m4, [cf0q + OFFSET + mmsize]
- mova m0, [cf0q + OFFSET + 4*NUM_COEF + mmsize]
-%endif
- mulps va, %2
- mulps vb, %2
-%if %0 == 3
-%if cpuflag(fma3)
- fmaddps va, m4, %3, va
- fmaddps vb, m0, %3, vb
-%else
- mulps m4, %3
- mulps m0, %3
- addps va, m4
- addps vb, m0
-%endif
-%endif
- ; va = va1 va2 va3 va4
- ; vb = vb1 vb2 vb3 vb4
-%if %1
- SWAP va, vb
-%endif
- mova m4, va
- unpcklps va, vb ; va3 vb3 va4 vb4
- unpckhps m4, vb ; va1 vb1 va2 vb2
- addps m4, va ; va1+3 vb1+3 va2+4 vb2+4
- movhlps vb, m4 ; va1+3 vb1+3
- addps vb, m4 ; va0..4 vb0..4
- movlps [outq + count], vb
-%if %1
- sub cf0q, 8*NUM_COEF
-%endif
- add count, 8
- jl .loop%1
-%endmacro
-
-; void dca_lfe_fir(float *out, float *in, float *coefs)
-%macro DCA_LFE_FIR 1
-cglobal dca_lfe_fir%1, 3,3,6-%1, out, in, cf0
-%define IN1 m3
-%define IN2 m5
-%define count inq
-%define NUM_COEF 4*(2-%1)
-%define NUM_OUT 32*(%1+1)
-
- movu IN1, [inq + 4 - 1*mmsize]
- shufps IN1, IN1, q0123
-%if %1 == 0
- movu IN2, [inq + 4 - 2*mmsize]
- shufps IN2, IN2, q0123
-%endif
-
- mov count, -4*NUM_OUT
- add cf0q, 4*NUM_COEF*NUM_OUT
- add outq, 4*NUM_OUT
- ; compute v0 first
-%if %1 == 0
- FIR_LOOP 0, IN1, IN2
-%else
- FIR_LOOP 0, IN1
-%endif
- shufps IN1, IN1, q0123
- mov count, -4*NUM_OUT
- ; cf1 already correctly positioned
- add outq, 4*NUM_OUT ; outq now at out2
- sub cf0q, 8*NUM_COEF
-%if %1 == 0
- shufps IN2, IN2, q0123
- FIR_LOOP 1, IN2, IN1
-%else
- FIR_LOOP 1, IN1
-%endif
- RET
-%endmacro
-
-INIT_XMM sse
-DCA_LFE_FIR 0
-DCA_LFE_FIR 1
-%if HAVE_FMA3_EXTERNAL
-INIT_XMM fma3
-DCA_LFE_FIR 0
-%endif
diff --git a/libavcodec/x86/dcadsp_init.c b/libavcodec/x86/dcadsp_init.c
deleted file mode 100644
index c27c045d1d..0000000000
--- a/libavcodec/x86/dcadsp_init.c
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2012-2014 Christophe Gisquet <christophe.gisquet@gmail.com>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-#include "libavutil/attributes.h"
-#include "libavutil/cpu.h"
-#include "libavutil/x86/cpu.h"
-#include "libavcodec/dcadsp.h"
-
-void ff_dca_lfe_fir0_sse(float *out, const float *in, const float *coefs);
-void ff_dca_lfe_fir1_sse(float *out, const float *in, const float *coefs);
-void ff_dca_lfe_fir0_fma3(float *out, const float *in, const float *coefs);
-
-av_cold void ff_dcadsp_init_x86(DCADSPContext *s)
-{
- int cpu_flags = av_get_cpu_flags();
-
- if (EXTERNAL_SSE(cpu_flags)) {
- s->lfe_fir[0] = ff_dca_lfe_fir0_sse;
- s->lfe_fir[1] = ff_dca_lfe_fir1_sse;
- }
-
- if (EXTERNAL_FMA3(cpu_flags)) {
- s->lfe_fir[0] = ff_dca_lfe_fir0_fma3;
- }
-}
diff --git a/tests/checkasm/Makefile b/tests/checkasm/Makefile
index 301c2e2f1d..14a11d64c3 100644
--- a/tests/checkasm/Makefile
+++ b/tests/checkasm/Makefile
@@ -1,7 +1,7 @@
# libavcodec tests
AVCODECOBJS-$(CONFIG_ALAC_DECODER) += alacdsp.o
AVCODECOBJS-$(CONFIG_BSWAPDSP) += bswapdsp.o
-AVCODECOBJS-$(CONFIG_DCA_DECODER) += dcadsp.o synth_filter.o
+#AVCODECOBJS-$(CONFIG_DCA_DECODER) += synth_filter.o
AVCODECOBJS-$(CONFIG_FLACDSP) += flacdsp.o
AVCODECOBJS-$(CONFIG_FMTCONVERT) += fmtconvert.o
AVCODECOBJS-$(CONFIG_H264PRED) += h264pred.o
diff --git a/tests/checkasm/checkasm.c b/tests/checkasm/checkasm.c
index dd37649ba7..f7d1331317 100644
--- a/tests/checkasm/checkasm.c
+++ b/tests/checkasm/checkasm.c
@@ -71,10 +71,9 @@ static const struct {
#if CONFIG_BSWAPDSP
{ "bswapdsp", checkasm_check_bswapdsp },
#endif
- #if CONFIG_DCA_DECODER
- { "dcadsp", checkasm_check_dcadsp },
+/* #if CONFIG_DCA_DECODER
{ "synth_filter", checkasm_check_synth_filter },
- #endif
+ #endif*/
#if CONFIG_FLACDSP
{ "flacdsp", checkasm_check_flacdsp },
#endif
diff --git a/tests/checkasm/checkasm.h b/tests/checkasm/checkasm.h
index 21000232d3..98c0216464 100644
--- a/tests/checkasm/checkasm.h
+++ b/tests/checkasm/checkasm.h
@@ -32,7 +32,6 @@
void checkasm_check_alacdsp(void);
void checkasm_check_bswapdsp(void);
-void checkasm_check_dcadsp(void);
void checkasm_check_flacdsp(void);
void checkasm_check_fmtconvert(void);
void checkasm_check_h264pred(void);
diff --git a/tests/checkasm/dcadsp.c b/tests/checkasm/dcadsp.c
deleted file mode 100644
index 5c7ff6f2d1..0000000000
--- a/tests/checkasm/dcadsp.c
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * Copyright (c) 2015 Janne Grunau
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with FFmpeg; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <math.h>
-#include <string.h>
-#include <stdlib.h>
-
-#include "libavutil/internal.h"
-#include "libavutil/intfloat.h"
-#include "libavcodec/dca.h"
-#include "libavcodec/dcadsp.h"
-#include "libavcodec/dcadata.h"
-
-#include "checkasm.h"
-
-#define randomize_lfe_fir(size) \
- do { \
- int i; \
- for (i = 0; i < size; i++) { \
- float f = (float)rnd() / (UINT_MAX >> 1) - 1.0f; \
- in[i] = f; \
- } \
- for (i = 0; i < 256; i++) { \
- float f = (float)rnd() / (UINT_MAX >> 1) - 1.0f; \
- coeffs[i] = f; \
- } \
- } while (0)
-
-#define check_lfe_fir(decifactor, eps) \
- do { \
- LOCAL_ALIGNED_16(float, in, [256 / decifactor]); \
- LOCAL_ALIGNED_16(float, out0, [decifactor * 2]); \
- LOCAL_ALIGNED_16(float, out1, [decifactor * 2]); \
- LOCAL_ALIGNED_16(float, coeffs, [256]); \
- int i; \
- const float * in_ptr = in + (256 / decifactor) - 1; \
- declare_func(void, float *out, const float *in, const float *coeffs); \
- /* repeat the test several times */ \
- for (i = 0; i < 32; i++) { \
- int j; \
- memset(out0, 0, sizeof(*out0) * 2 * decifactor); \
- memset(out1, 0xFF, sizeof(*out1) * 2 * decifactor); \
- randomize_lfe_fir(256 / decifactor); \
- call_ref(out0, in_ptr, coeffs); \
- call_new(out1, in_ptr, coeffs); \
- for (j = 0; j < 2 * decifactor; j++) { \
- if (!float_near_abs_eps(out0[j], out1[j], eps)) { \
- if (0) { \
- union av_intfloat32 x, y; x.f = out0[j]; y.f = out1[j]; \
- fprintf(stderr, "%3d: %11g (0x%08x); %11g (0x%08x)\n", \
- j, x.f, x.i, y.f, y.i); \
- } \
- fail(); \
- break; \
- } \
- } \
- bench_new(out1, in_ptr, coeffs); \
- } \
- } while (0)
-
-void checkasm_check_dcadsp(void)
-{
- DCADSPContext c;
-
- ff_dcadsp_init(&c);
-
- /* values are limited to {-8, 8} so absolute epsilon is good enough */
- if (check_func(c.lfe_fir[0], "dca_lfe_fir0"))
- check_lfe_fir(32, 1.0e-6f);
-
- if (check_func(c.lfe_fir[1], "dca_lfe_fir1"))
- check_lfe_fir(64, 1.0e-6f);
-
- report("dcadsp");
-}
diff --git a/tests/fate/acodec.mak b/tests/fate/acodec.mak
index e0f23208e2..62b1bc1f09 100644
--- a/tests/fate/acodec.mak
+++ b/tests/fate/acodec.mak
@@ -99,14 +99,14 @@ FATE_ACODEC-$(call ENCDEC, ALAC, MOV) += fate-acodec-alac
fate-acodec-alac: FMT = mov
fate-acodec-alac: CODEC = alac -compression_level 1
-FATE_ACODEC-$(call ENCDEC, DCA, DTS) += fate-acodec-dca
+#FATE_ACODEC-$(call ENCDEC, DCA, DTS) += fate-acodec-dca
fate-acodec-dca: tests/data/asynth-44100-2.wav
fate-acodec-dca: SRC = tests/data/asynth-44100-2.wav
fate-acodec-dca: CMD = md5 -i $(TARGET_PATH)/$(SRC) -c:a dca -strict -2 -f dts -flags +bitexact
fate-acodec-dca: CMP = oneline
fate-acodec-dca: REF = 7ffdefdf47069289990755c79387cc90
-FATE_ACODEC-$(call ENCDEC, DCA, WAV) += fate-acodec-dca2
+#FATE_ACODEC-$(call ENCDEC, DCA, WAV) += fate-acodec-dca2
fate-acodec-dca2: CMD = enc_dec_pcm dts wav s16le $(SRC) -c:a dca -strict -2 -flags +bitexact
fate-acodec-dca2: REF = $(SRC)
fate-acodec-dca2: CMP = stddev
diff --git a/tests/fate/audio.mak b/tests/fate/audio.mak
index 493bb8ce43..686b7dfad7 100644
--- a/tests/fate/audio.mak
+++ b/tests/fate/audio.mak
@@ -21,12 +21,7 @@ fate-dca-core: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts.ts
fate-dca-core: CMP = oneoff
fate-dca-core: REF = $(SAMPLES)/dts/dts.pcm
-FATE_DCA-$(CONFIG_DTS_DEMUXER) += fate-dca-xll
-fate-dca-xll: CMD = pcm -disable_xll 0 -i $(TARGET_SAMPLES)/dts/master_audio_7.1_24bit.dts
-fate-dca-xll: CMP = oneoff
-fate-dca-xll: REF = $(SAMPLES)/dts/master_audio_7.1_24bit_2.pcm
-
-FATE_SAMPLES_AUDIO-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
+#FATE_SAMPLES_AUDIO-$(CONFIG_DCA_DECODER) += $(FATE_DCA-yes)
fate-dca: $(FATE_DCA-yes)
FATE_SAMPLES_AUDIO-$(call DEMDEC, DSICIN, DSICINAUDIO) += fate-delphine-cin-audio
@@ -36,7 +31,7 @@ FATE_SAMPLES_AUDIO-$(call DEMDEC, DSS, DSS_SP) += fate-dss-lp fate-dss-sp
fate-dss-lp: CMD = framecrc -i $(TARGET_SAMPLES)/dss/lp.dss -frames 30
fate-dss-sp: CMD = framecrc -i $(TARGET_SAMPLES)/dss/sp.dss -frames 30
-FATE_SAMPLES_AUDIO-$(call DEMDEC, DTS, DCA) += fate-dts_es
+#FATE_SAMPLES_AUDIO-$(call DEMDEC, DTS, DCA) += fate-dts_es
fate-dts_es: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts_es.dts
fate-dts_es: CMP = oneoff
fate-dts_es: REF = $(SAMPLES)/dts/dts_es_2.pcm
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-28 20:03:01 +0000