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.
author: foo86 2016-01-16 11:07:08 +0300
committer: Hendrik Leppkes 2016-01-31 17:09:38 +0100
commit: 46089967722f74e794865a044f5f682f26628802 (patch)
tree: b4ca91d42d3eb0da3229d217323565738c101f87
parent: b552f3afa2a76142c9aa87a89e31e75423b4cd3b (diff)
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
author	foo86	2016-01-16 11:07:08 +0300
committer	Hendrik Leppkes	2016-01-31 17:09:38 +0100
commit	46089967722f74e794865a044f5f682f26628802 (patch)
tree	b4ca91d42d3eb0da3229d217323565738c101f87
parent	b552f3afa2a76142c9aa87a89e31e75423b4cd3b (diff)