[libav-devel] [PATCH] Experimental DCA encoder
From: Alexander E. Patrakov
---
Changelog |1 +
doc/general.texi |2 +-
libavcodec/Makefile|1 +
libavcodec/allcodecs.c |2 +-
libavcodec/dcaenc.c| 587
libavcodec/dcaenc.h| 544
6 files changed, 1135 insertions(+), 2 deletions(-)
create mode 100644 libavcodec/dcaenc.c
create mode 100644 libavcodec/dcaenc.h
diff --git a/Changelog b/Changelog
index 4bc6d31..d446c64 100644
--- a/Changelog
+++ b/Changelog
@@ -86,6 +86,7 @@ version :
- AAC encoding via libvo-aacenc
- AMR-WB encoding via libvo-amrwbenc
- xWMA demuxer
+- Experimental DCA encoder
version 0.6:
diff --git a/doc/general.texi b/doc/general.texi
index 423402c..f290aca 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -602,7 +602,7 @@ following image formats are supported:
@tab Codec used in Delphine Software International games.
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
-@item DCA (DTS Coherent Acoustics) @tab @tab X
+@item DCA (DTS Coherent Acoustics) @tab X @tab X
@item DPCM id RoQ@tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item DPCM Interplay @tab @tab X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 1cdae2c..058c6f9 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -104,6 +104,7 @@ OBJS-$(CONFIG_COOK_DECODER)+= cook.o
OBJS-$(CONFIG_CSCD_DECODER)+= cscd.o
OBJS-$(CONFIG_CYUV_DECODER)+= cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
+OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.o
OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 40a7e23..1d969cd 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -238,7 +238,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
-REGISTER_DECODER (DCA, dca);
+REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_DECODER (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
new file mode 100644
index 000..6d8694f
--- /dev/null
+++ b/libavcodec/dcaenc.c
@@ -0,0 +1,587 @@
+/*
+ * DCA encoder
+ * Copyright (C) 2008 Alexander E. Patrakov
+ * 2010 Benjamin Larsson
+ * 2011 Xiang Wang
+ *
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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/avassert.h"
+#include "libavutil/audioconvert.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "dcaenc.h"
+#include "dcadata.h"
+
+#undef NDEBUG
+
+#define MAX_CHANNELS 6
+#define DCA_SUBBANDS_32 32
+#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_HEADER_SIZE 13
+
+#define DCA_SUBBANDS 32 ///< Subband activity count
+#define QUANTIZER_BITS 16
+#define SUBFRAMES 1
+#define SUBSUBFRAMES 4
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+#define LFE_BITS 8
+#define LFE_INTERPOLATION 64
+#define LFE_PRESENT 2
+#define LFE_MISSING 0
+
+static const int8_t dca_lfe_index[] = {
+1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+{ 0, -1, -1, -1, -1, -1, -1, -1, -1 },
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1 },
+{ 1, 2, 0, -1, -1, -1, -1, -1, -1 },
+{ 0, 1, -1, 2, -1, -1, -1, -1, -1 },
+{ 1, 2, 0, -1, 3, -1, -1, -1, -1 },
+{ 0, 1, -1, 2, 3, -1, -1, -1, -1 },
+{ 1, 2, 0, -1, 3, 4, -1, -1, -1 },
+{ 2, 3, -1, 0, 1, 4, 5, -1, -1 },
+{ 1, 2, 0, -1, 3, 4, 5, -1, -1 },
+{ 0, -1, 4, 5, 2, 3, 1, -1, -1 },
+{ 3, 4, 1, -1, 0, 2, 5, 6, -1 },
+{ 2, 3, -1, 5, 7, 0, 1, 4, 6 },
+{ 3, 4, 1, -1, 0,
Re: [libav-devel] [PATCH] Experimental DCA encoder
On Fri, Apr 15, 2011 at 04:39:48PM +0200, Diego Biurrun wrote: > From: Alexander E. Patrakov Patch updated to apply against HEAD and prettified. Benjamin, rebase any review comments on top of this one :) Diego ___ libav-devel mailing list libav-devel@libav.org https://lists.libav.org/mailman/listinfo/libav-devel
[libav-devel] [PATCH] Experimental DCA encoder
From: Alexander E. Patrakov
---
Changelog |1 +
doc/general.texi |2 +-
libavcodec/Makefile|1 +
libavcodec/allcodecs.c |2 +-
libavcodec/dcaenc.c| 587
libavcodec/dcaenc.h| 544
6 files changed, 1135 insertions(+), 2 deletions(-)
create mode 100644 libavcodec/dcaenc.c
create mode 100644 libavcodec/dcaenc.h
diff --git a/Changelog b/Changelog
index 4bc6d31..d446c64 100644
--- a/Changelog
+++ b/Changelog
@@ -86,6 +86,7 @@ version :
- AAC encoding via libvo-aacenc
- AMR-WB encoding via libvo-amrwbenc
- xWMA demuxer
+- Experimental DCA encoder
version 0.6:
diff --git a/doc/general.texi b/doc/general.texi
index 423402c..f290aca 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -602,7 +602,7 @@ following image formats are supported:
@tab Codec used in Delphine Software International games.
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
-@item DCA (DTS Coherent Acoustics) @tab @tab X
+@item DCA (DTS Coherent Acoustics) @tab X @tab X
@item DPCM id RoQ@tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item DPCM Interplay @tab @tab X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 1cdae2c..058c6f9 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -104,6 +104,7 @@ OBJS-$(CONFIG_COOK_DECODER)+= cook.o
OBJS-$(CONFIG_CSCD_DECODER)+= cscd.o
OBJS-$(CONFIG_CYUV_DECODER)+= cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
+OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.o
OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 40a7e23..1d969cd 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -238,7 +238,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
-REGISTER_DECODER (DCA, dca);
+REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_DECODER (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
new file mode 100644
index 000..4734524
--- /dev/null
+++ b/libavcodec/dcaenc.c
@@ -0,0 +1,587 @@
+/*
+ * DCA encoder
+ * Copyright (C) 2008 Alexander E. Patrakov
+ * 2010 Benjamin Larsson
+ * 2011 Xiang Wang
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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/avassert.h"
+#include "libavutil/audioconvert.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "dcaenc.h"
+#include "dcadata.h"
+
+#undef NDEBUG
+
+#define MAX_CHANNELS 6
+#define DCA_SUBBANDS_32 32
+#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_HEADER_SIZE 13
+
+#define DCA_SUBBANDS 32 ///< Subband activity count
+#define QUANTIZER_BITS 16
+#define SUBFRAMES 1
+#define SUBSUBFRAMES 4
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+#define LFE_BITS 8
+#define LFE_INTERPOLATION 64
+#define LFE_PRESENT 2
+#define LFE_MISSING 0
+
+static const int8_t dca_lfe_index[] = {
+1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, 2, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, 3, -1, -1, -1, -1},
+{ 0, 1, -1, 2, 3, -1, -1, -1, -1},
+{ 1, 2, 0, -1, 3, 4, -1, -1, -1},
+{ 2, 3, -1, 0, 1, 4, 5, -1, -1},
+{ 1, 2, 0, -1, 3, 4, 5, -1, -1},
+{ 0, -1, 4, 5, 2, 3, 1, -1, -1},
+{ 3, 4, 1, -1, 0, 2, 5, 6, -1},
+{ 2, 3, -1, 5, 7, 0, 1, 4, 6},
+{ 3, 4, 1, -1, 0, 2, 5, 7, 6},
+
Re: [libav-devel] [PATCH] Experimental DCA encoder
On Fri, Apr 08, 2011 at 01:33:21AM +0200, Alexander E. Patrakov wrote:
> ---
> Changelog |1 +
> doc/general.texi |2 +-
> libavcodec/Makefile|1 +
> libavcodec/allcodecs.c |2 +-
> libavcodec/dcaenc.c| 587
>
> libavcodec/dcaenc.h| 544
> 6 files changed, 1135 insertions(+), 2 deletions(-)
> create mode 100644 libavcodec/dcaenc.c
> create mode 100644 libavcodec/dcaenc.h
>
[...]
> diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
> new file mode 100644
> index 000..670cc3e
> --- /dev/null
> +++ b/libavcodec/dcaenc.c
> @@ -0,0 +1,587 @@
> +/*
> + * DCA encoder
> + * Copyright (C) 2008 Alexander E. Patrakov
> + * 2010 Benjamin Larsson
> + * 2011 Xiang Wang
> + * This file is part of Libav.
> + *
> + * Libav 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.
> + *
> + * Libav 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 Libav; 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/avassert.h"
> +#include "libavutil/audioconvert.h"
> +#include "avcodec.h"
> +#include "get_bits.h"
> +#include "put_bits.h"
> +#include "dcaenc.h"
> +#include "dcadata.h"
> +
> +#undef NDEBUG
> +
> +#define MAX_CHANNELS 6
> +#define DCA_SUBBANDS_32 32
> +#define DCA_MAX_FRAME_SIZE 16383
> +#define DCA_HEADER_SIZE 13
> +
> +#define DCA_SUBBANDS 32 ///< Subband activity count
> +#define QUANTIZER_BITS 16
> +#define SUBFRAMES 1
> +#define SUBSUBFRAMES 4
> +#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
> +#define LFE_BITS 8
> +#define LFE_INTERPOLATION 64
> +#define LFE_PRESENT 2
> +#define LFE_MISSING 0
have you considered sharing some of these with decoder?
> +static const int8_t dca_lfe_index[] = {
> +1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
> +};
this table too
> +static const int8_t dca_channel_reorder_lfe[][9] = {
> +{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 1, 2, 0, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, 2, -1, -1, -1, -1, -1},
> +{ 1, 2, 0, -1, 3, -1, -1, -1, -1},
> +{ 0, 1, -1, 2, 3, -1, -1, -1, -1},
> +{ 1, 2, 0, -1, 3, 4, -1, -1, -1},
> +{ 2, 3, -1, 0, 1, 4, 5, -1, -1},
> +{ 1, 2, 0, -1, 3, 4, 5, -1, -1},
> +{ 0, -1, 4, 5, 2, 3, 1, -1, -1},
> +{ 3, 4, 1, -1, 0, 2, 5, 6, -1},
> +{ 2, 3, -1, 5, 7, 0, 1, 4, 6},
> +{ 3, 4, 1, -1, 0, 2, 5, 7, 6},
> +};
> +
> +static const int8_t dca_channel_reorder_nolfe[][9] = {
> +{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
> +{ 1, 2, 0, -1, -1, -1, -1, -1, -1},
> +{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
> +{ 1, 2, 0, 3, -1, -1, -1, -1, -1},
> +{ 0, 1, 2, 3, -1, -1, -1, -1, -1},
> +{ 1, 2, 0, 3, 4, -1, -1, -1, -1},
> +{ 2, 3, 0, 1, 4, 5, -1, -1, -1},
> +{ 1, 2, 0, 3, 4, 5, -1, -1, -1},
> +{ 0, 4, 5, 2, 3, 1, -1, -1, -1},
> +{ 3, 4, 1, 0, 2, 5, 6, -1, -1},
> +{ 2, 3, 5, 7, 0, 1, 4, 6, -1},
> +{ 3, 4, 1, 0, 2, 5, 7, 6, -1},
> +};
please add a not on _how_ these tables remap channels (and why they do it in
different way from decoder)
> +typedef struct {
> +PutBitContext pb;
> +int32_t history[MAX_CHANNELS][512]; /* This is a circular buffer */
> +int start[MAX_CHANNELS];
> +int frame_size;
> +int prim_channels;
> +int lfe_channel;
> +int sample_rate_code;
> +int scale_factor[MAX_CHANNELS][DCA_SUBBANDS_32];
> +int lfe_scale_factor;
> +int lfe_data[SUBFRAMES*SUBSUBFRAMES*4];
> +
> +int a_mode; ///< audio channels arrangement
> +int num_channel;
> +int lfe_state;
> +int lfe_offset;
> +const int8_t* channel_order_tab;///< channel reordering table, lfe
> and non lfe
> +
> +int32_t pcm[FFMAX(LFE_INTERPOLATION, DCA_SUBBANDS_32)];
> +int32_t subband[PCM_SAMPLES][MAX_CHANNELS][DCA_SUBBANDS_32]; /*
> [sample][channel][subband] */
> +} DCAContext;
> +
> +static i
Re: [libav-devel] [PATCH] Experimental DCA encoder v2
Ping MvH Benjamin Larsson ___ libav-devel mailing list libav-devel@libav.org https://lists.libav.org/mailman/listinfo/libav-devel
[libav-devel] [PATCH] Experimental DCA encoder v2
Fixed some things but not others.
MvH
Benjamin Larsson
>From 20a2180193d0fcfaaf71fb1d9f3bb4cd90a1ce0f Mon Sep 17 00:00:00 2001
From: Alexander E. Patrakov
Date: Fri, 8 Apr 2011 01:33:21 +0200
Subject: [PATCH] Experimental DCA encoder
---
Changelog |1 +
doc/general.texi |2 +-
libavcodec/Makefile|1 +
libavcodec/allcodecs.c |2 +-
libavcodec/dcaenc.c| 587
libavcodec/dcaenc.h| 544
6 files changed, 1135 insertions(+), 2 deletions(-)
create mode 100644 libavcodec/dcaenc.c
create mode 100644 libavcodec/dcaenc.h
diff --git a/Changelog b/Changelog
index ec09c28..045c3f5 100644
--- a/Changelog
+++ b/Changelog
@@ -83,6 +83,7 @@ version :
- Linux framebuffer input device added
- Chronomaster DFA decoder
- Mobotix MxPEG decoder
+- Experimental DCA encoder
version 0.6:
diff --git a/doc/general.texi b/doc/general.texi
index 5d66e42..2eeb609 100644
--- a/doc/general.texi
+++ b/doc/general.texi
@@ -599,7 +599,7 @@ following image formats are supported:
@tab Codec used in Delphine Software International games.
@item COOK @tab @tab X
@tab All versions except 5.1 are supported.
-@item DCA (DTS Coherent Acoustics) @tab @tab X
+@item DCA (DTS Coherent Acoustics) @tab X @tab X
@item DPCM id RoQ@tab X @tab X
@tab Used in Quake III, Jedi Knight 2, other computer games.
@item DPCM Interplay @tab @tab X
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 837f7e2..195469d 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -104,6 +104,7 @@ OBJS-$(CONFIG_COOK_DECODER)+= cook.o
OBJS-$(CONFIG_CSCD_DECODER)+= cscd.o
OBJS-$(CONFIG_CYUV_DECODER)+= cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
+OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.o
OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 7636392..9530358 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -238,7 +238,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
-REGISTER_DECODER (DCA, dca);
+REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_DECODER (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
new file mode 100644
index 000..670cc3e
--- /dev/null
+++ b/libavcodec/dcaenc.c
@@ -0,0 +1,587 @@
+/*
+ * DCA encoder
+ * Copyright (C) 2008 Alexander E. Patrakov
+ * 2010 Benjamin Larsson
+ * 2011 Xiang Wang
+ * This file is part of Libav.
+ *
+ * Libav 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.
+ *
+ * Libav 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 Libav; 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/avassert.h"
+#include "libavutil/audioconvert.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "dcaenc.h"
+#include "dcadata.h"
+
+#undef NDEBUG
+
+#define MAX_CHANNELS 6
+#define DCA_SUBBANDS_32 32
+#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_HEADER_SIZE 13
+
+#define DCA_SUBBANDS 32 ///< Subband activity count
+#define QUANTIZER_BITS 16
+#define SUBFRAMES 1
+#define SUBSUBFRAMES 4
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+#define LFE_BITS 8
+#define LFE_INTERPOLATION 64
+#define LFE_PRESENT 2
+#define LFE_MISSING 0
+
+static const int8_t dca_lfe_index[] = {
+1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, 2, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, 3, -1, -1, -1, -1},
+{ 0, 1, -1, 2, 3, -1, -1, -1, -1},
+{ 1, 2, 0, -1, 3, 4, -1, -1, -1},
+{ 2, 3, -1, 0, 1, 4, 5, -1, -1},
+
Re: [libav-devel] [PATCH] Experimental DCA encoder
"Ronald S. Bultje" writes:
> Hi,
>
> 2011/4/7 Benjamin Larsson :
>> +static inline int32_t mul32(int32_t a, int32_t b)
>> +{
>> +/* on >=i686, gcc compiles this into a single "imull" instruction */
>> +int64_t r = (int64_t)a * b;
>> +/* round the result before truncating - improves accuracy */
>> +return (r + 0x8000) >> 32;
>> +}
>
> I believe this is what the MUL64() macro is for, then it works on
> non-x86 archs also, but that doesn't round...
ARM has a rounding version of that instruction FWIW.
--
Måns Rullgård
m...@mansr.com
___
libav-devel mailing list
libav-devel@libav.org
https://lists.libav.org/mailman/listinfo/libav-devel
Re: [libav-devel] [PATCH] Experimental DCA encoder
On Fri, Apr 08, 2011 at 07:53:32AM -0400, Ronald S. Bultje wrote: > Hi, [...] > > Ah, that's what the fixme is for I guess. Sounds ok to me then. > (Didn't Mans just implement fixed-point FFT or something like that? Or > was that "just" MDCT?) MDCT is often implemented via FFT. P.S. And if somebody gets crazy idea of implementing DTS-MA he should implement bitexact fixedpoint DCT32 they use inside QMF (and which is not given in DTS specs IIRC). ___ libav-devel mailing list libav-devel@libav.org https://lists.libav.org/mailman/listinfo/libav-devel
Re: [libav-devel] [PATCH] Experimental DCA encoder
Hi,
On Fri, Apr 8, 2011 at 5:02 AM, Benjamin Larsson wrote:
> On 04/08/2011 03:58 AM, Ronald S. Bultje wrote:
>> 2011/4/7 Benjamin Larsson :
>>> +static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
>>> int channel)
>>> +{
>>> + int band, i, j, k;
>>> + int32_t resp;
>>> + int32_t accum[DCA_SUBBANDS_32];
>>> +
>>> + add_new_samples(c, in, DCA_SUBBANDS_32, channel);
>>> +
>>> + /* Calculate the dot product of the signal with the (possibly inverted)
>>> + reference decoder's response to this vector:
>>> + (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
>>> + so that -1.0 cancels 1.0 from the previous step */
>>> +
>>> + memset(accum,0,sizeof(accum));
>>> +
>>> + for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
>>> + accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] +=
>>> mul32(c->history[channel][i], UnQMF[j]);
>>> + for (i = 0; i < c->start[channel]; k++, j++, i++)
>>> + accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] +=
>>> mul32(c->history[channel][i], UnQMF[j]);
>>> +
>>> + resp = 0;
>>> + /* TODO: implement FFT instead of this naive calculation */
>>> + for (band = 0; band < DCA_SUBBANDS_32; band++) {
>>> + for (j = 0; j < 32; j++)
>>> + resp += mul32(accum[j], band_delta_factor(band, j));
>>> +
>>> + out[band] = (band & 2) ? (-resp) : resp;
>>> + }
>>> +}
>>
>> I'll probably sound very naive, but what does this piece of the code do?
>
> Decomposes the signal into bands. Time domain -> frequency domain.
Ah, that's what the fixme is for I guess. Sounds ok to me then.
(Didn't Mans just implement fixed-point FFT or something like that? Or
was that "just" MDCT?)
Ronald
___
libav-devel mailing list
libav-devel@libav.org
https://lists.libav.org/mailman/listinfo/libav-devel
Re: [libav-devel] [PATCH] Experimental DCA encoder
On Fri, Apr 08, 2011 at 01:41:37AM +0200, Benjamin Larsson wrote: > [...] Please make sure this compiles with --disable-everything --enable-encoder=dca. Diego ___ libav-devel mailing list libav-devel@libav.org https://lists.libav.org/mailman/listinfo/libav-devel
Re: [libav-devel] [PATCH] Experimental DCA encoder
On Fri, Apr 08, 2011 at 01:41:37AM +0200, Benjamin Larsson wrote:
>
> ---
> libavcodec/Makefile|1 +
> libavcodec/allcodecs.c |2 +-
> libavcodec/dcaenc.c| 581
>
> libavcodec/dcaenc.h| 544 +
> 4 files changed, 1127 insertions(+), 1 deletions(-)
changelog, docs update, minor bump
> --- /dev/null
> +++ b/libavcodec/dcaenc.c
> @@ -0,0 +1,581 @@
> +/*
> + * DCA encoder
> + * Copyright (C) 2008 Alexander E. Patrakov
> + * 2010 Benjamin Larsson
> + * 2011 Xiang Wang
> + * This file is part of Libav.
> + *
> + * 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
That still contains some FFmpeg references, please fix.
> +static void add_new_samples(DCAContext *c, const int32_t *in, int count, int
> channel){
{ on next line, long line
> +static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
> int channel)
long line
> +int band, i, j, k;
> +int32_t resp;
> +int32_t accum[DCA_SUBBANDS_32];
> +
> +memset(accum,0,sizeof(accum));
Just initialize to 0
> +static int32_t lfe_fir_64i[512];
> +static int lfe_downsample(DCAContext *c, int32_t in[LFE_INTERPOLATION]){
{ on next line
> +static void init_lfe_fir(void){
ditto
> +static int initialized;
> +int i;
> +if(initialized)
if (
> +for(i=0; i<512; i++)
for (
.. and please give those operators some room to breathe ..
> +static void put_primary_audio_header(DCAContext *c)
> +{
> +/* From dca.c */
> +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 };
That comment makes me suspicious - are these duplicated?
> +/* Subband activity count */
> +for (ch=0; chprim_channels; ch++) {
> +
> +/* High frequency VQ start subband */
> +for (ch=0; chprim_channels; ch++) {
> +
> +/* Joint intensity coding index: 0, 0 */
> +for (ch=0; chprim_channels; ch++) {
> +
> +/* Transient mode codebook: A4, A4 (arbitrary) */
> +for (ch=0; chprim_channels; ch++) {
> +
> +/* Scale factor code book: 7 bit linear, 7-bit sqrt table (for each
> channel) */
> +for (ch=0; chprim_channels; ch++) {
> +
> +/* Bit allocation quantizer select: linear 5-bit */
> +for (ch=0; chprim_channels; ch++) {
> +
> +/* Quantization index codebook select: dummy data
> + to avoid transmission of scale factor adjustment */
> +for (i=1; i<11; i++) {
> +for (ch=0; chprim_channels; ch++) {
.. more operators in need of elbow room .. :)
There are more instances below, same for 'if(' and 'for(', { on the wrong
line and excessively long lines that could be shortened easily. Fixing
would be appreciated.
You could also drop (or not) some {} around if/for blocks.
> +/**
> + * 8-23 bits quantization
> + * @param sample
> + * @param bits
> + */
> +static inline uint32_t quantize(int32_t sample, int bits)
These Doxygen parameter comments are pretty useless.
> +switch(avctx->channel_layout) {
> + case AV_CH_LAYOUT_STEREO: c->a_mode = 2; c->num_channel = 2; break;
switch (
> + case AV_CH_LAYOUT_5POINT0: c->a_mode = 9; c->num_channel = 9; break;
> + case AV_CH_LAYOUT_5POINT1: c->a_mode = 9; c->num_channel = 9; break;
> + case AV_CH_LAYOUT_5POINT0_BACK: c->a_mode = 9; c->num_channel = 9;
> break;
> + case AV_CH_LAYOUT_5POINT1_BACK: c->a_mode = 9; c->num_channel = 9;
> break;
> + default:
> +av_log(avctx, AV_LOG_ERROR, "Only stereo, 5.1, 5.0, 5.0back and
> 5.0front channel layouts supported at the moment!\n");
> +return AVERROR_PATCHWELCOME;
> +}
Indent the case statements at the same depth as the switch and fix that
instance of 2-space indentation.
> +AVCodec ff_dca_encoder = {
> +.name = "dca",
> +.type = CODEC_TYPE_AUDIO,
> +.id = CODEC_ID_DTS,
> +.priv_data_size = sizeof(DCAContext),
> +.init = DCA_encode_init,
> +.encode = DCA_encode_frame,
> +.capabilities = CODEC_CAP_EXPERIMENTAL,
> +.sample_fmts = (const enum
> AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
> +NULL,
> +NULL,
> +};
pointless trailing NULLs, long_name missing
> --- /dev/null
> +++ b
Re: [libav-devel] [PATCH] Experimental DCA encoder
On 04/08/2011 03:58 AM, Ronald S. Bultje wrote:
> Hi,
>
> 2011/4/7 Benjamin Larsson :
>> +static inline int32_t mul32(int32_t a, int32_t b)
>> +{
>> +/* on >=i686, gcc compiles this into a single "imull" instruction */
>> +int64_t r = (int64_t)a * b;
>> +/* round the result before truncating - improves accuracy */
>> +return (r + 0x8000) >> 32;
>> +}
> I believe this is what the MUL64() macro is for, then it works on
> non-x86 archs also, but that doesn't round...
Ok
>> +static int32_t cos_table[128];
> [..]
>> +static void qmf_init(void)
> There should probably be a static inited = 0; that you set after
> succesful init, since this function only has to run once, not every
> time the codec re-inits.
Ok
>> +int i;
>> +int32_t c[17], s[17];
>> +s[0] = 0; /* sin(index * PI / 64) * 0x7fff */
>> +c[0] = 0x7fff; /* cos(index * PI / 64) * 0x7fff */
>> +
>> +for (i = 1; i <= 16; i++) {
>> +s[i] = 2 * (mul32(c[i-1], 105372028) + mul32(s[i-1], 2144896908));
>> +c[i] = 2 * (mul32(c[i-1], 2144896908) - mul32(s[i-1], 105372028));
>> +}
>> +
>> +for (i = 0; i < 16; i++) {
>> +cos_table[i] = c[i] >> 3; /* so that the output doesn't overflow */
>> +cos_table[i+16] = s[16-i] >> 3;
>> +cos_table[i+32] = -s[i] >> 3;
>> +cos_table[i+48] = -c[16-i] >> 3;
>> +cos_table[i+64] = -c[i] >> 3;
>> +cos_table[i+80] = -s[16-i] >> 3;
>> +cos_table[i+96] = s[i] >> 3;ut
>> +cos_table[i+112] = c[16-i] >> 3;
>> +}
> For the cos-table, can you use libavcodec/sinewin.h functions? If not,
> does this look like a proper int-implementation of the same thing that
> we might want to share? E.g. does acenc_fixed.c have fixed-point cos
> table generation also?
>
> (I know, this is minor, ignore for now, not critical.)
The costable is custom. And there is a float based transform in the
decoder already.
>> +static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
>> int channel)
>> +{
>> +int band, i, j, k;
>> +int32_t resp;
>> +int32_t accum[DCA_SUBBANDS_32];
>> +
>> +add_new_samples(c, in, DCA_SUBBANDS_32, channel);
>> +
>> +/* Calculate the dot product of the signal with the (possibly inverted)
>> + reference decoder's response to this vector:
>> + (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
>> + so that -1.0 cancels 1.0 from the previous step */
>> +
>> +memset(accum,0,sizeof(accum));
>> +
>> +for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
>> +accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] +=
>> mul32(c->history[channel][i], UnQMF[j]);
>> +for (i = 0; i < c->start[channel]; k++, j++, i++)
>> +accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] +=
>> mul32(c->history[channel][i], UnQMF[j]);
>> +
>> +resp = 0;
>> +/* TODO: implement FFT instead of this naive calculation */
>> +for (band = 0; band < DCA_SUBBANDS_32; band++) {
>> +for (j = 0; j < 32; j++)
>> +resp += mul32(accum[j], band_delta_factor(band, j));
>> +
>> +out[band] = (band & 2) ? (-resp) : resp;
>> +}
>> +}
> I'll probably sound very naive, but what does this piece of the code do?
Decomposes the signal into bands. Time domain -> frequency domain.
>> +static void init_lfe_fir(void){
>> +static int initialized;
>> +int i;
>> +if(initialized)
>> +return;
>> +for(i=0; i<512; i++)
>> +lfe_fir_64i[i] = lfe_fir_64[i] * (1<<25); //float -> int32_t
>> +initialized = 1;
>> +}
> This kind of stuff, along with cos tables, FFT etc, raises the
> question why this wasn't implemented in float... Realistically, if
> !CONFIG_SMALL, the table should be hardcoded.
Don't really know but I'd like to keep the fixed-point code path for reg
test purposes.
> The bitstream stuff I didn't review because I'm not very familiar with it...
>
>> +static int DCA_encode_init(AVCodecContext *avctx) {
> [..]
>> +for(i=0; i<16; i++){
>> +if(dca_sample_rates[i] == avctx->sample_rate)
>> +break;
>> +}
>> +if(i==16){
>> +av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported\n",
>> avctx->sample_rate);
>> +return -1;
>> +}
>> +c->sample_rate_code = i;
> That is pretty unhelpful. What samplerates _are_ supported? Better
> yet, how would the user accomplish the resampling within ffmpeg?
It's the same as with every other libav encoder. But I guess it could
print all the 16 supported sample rates.
>> diff --git a/libavcodec/dcaenc.h b/libavcodec/dcaenc.h
> [..]
>> +/* This is a scaled version of the response of the reference decoder to
>> + this vector of subband samples: ( 1.0 0.0 0.0 ... 0.0 )
>> + */
>> +
>> +static const int32_t UnQMF[512] = {
> ??? What is that?
The decomposition filter.
> Ronald
MvH
Benjamin Larsson
___
libav-devel mailing list
libav-devel@libav.org
https:
Re: [libav-devel] [PATCH] Experimental DCA encoder
Hi,
2011/4/7 Benjamin Larsson :
> +static inline int32_t mul32(int32_t a, int32_t b)
> +{
> +/* on >=i686, gcc compiles this into a single "imull" instruction */
> +int64_t r = (int64_t)a * b;
> +/* round the result before truncating - improves accuracy */
> +return (r + 0x8000) >> 32;
> +}
I believe this is what the MUL64() macro is for, then it works on
non-x86 archs also, but that doesn't round...
> +static int32_t cos_table[128];
[..]
> +static void qmf_init(void)
There should probably be a static inited = 0; that you set after
succesful init, since this function only has to run once, not every
time the codec re-inits.
> +int i;
> +int32_t c[17], s[17];
> +s[0] = 0; /* sin(index * PI / 64) * 0x7fff */
> +c[0] = 0x7fff; /* cos(index * PI / 64) * 0x7fff */
> +
> +for (i = 1; i <= 16; i++) {
> +s[i] = 2 * (mul32(c[i-1], 105372028) + mul32(s[i-1], 2144896908));
> +c[i] = 2 * (mul32(c[i-1], 2144896908) - mul32(s[i-1], 105372028));
> +}
> +
> +for (i = 0; i < 16; i++) {
> +cos_table[i] = c[i] >> 3; /* so that the output doesn't overflow */
> +cos_table[i+16] = s[16-i] >> 3;
> +cos_table[i+32] = -s[i] >> 3;
> +cos_table[i+48] = -c[16-i] >> 3;
> +cos_table[i+64] = -c[i] >> 3;
> +cos_table[i+80] = -s[16-i] >> 3;
> +cos_table[i+96] = s[i] >> 3;
> +cos_table[i+112] = c[16-i] >> 3;
> +}
For the cos-table, can you use libavcodec/sinewin.h functions? If not,
does this look like a proper int-implementation of the same thing that
we might want to share? E.g. does acenc_fixed.c have fixed-point cos
table generation also?
(I know, this is minor, ignore for now, not critical.)
> +static void qmf_decompose(DCAContext *c, int32_t in[32], int32_t out[32],
> int channel)
> +{
> +int band, i, j, k;
> +int32_t resp;
> +int32_t accum[DCA_SUBBANDS_32];
> +
> +add_new_samples(c, in, DCA_SUBBANDS_32, channel);
> +
> +/* Calculate the dot product of the signal with the (possibly inverted)
> + reference decoder's response to this vector:
> + (0.0, 0.0, ..., 0.0, -1.0, 1.0, 0.0, ..., 0.0)
> + so that -1.0 cancels 1.0 from the previous step */
> +
> +memset(accum,0,sizeof(accum));
> +
> +for (k = 48, j = 0, i = c->start[channel]; i < 512; k++, j++, i++)
> +accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] +=
> mul32(c->history[channel][i], UnQMF[j]);
> +for (i = 0; i < c->start[channel]; k++, j++, i++)
> +accum[(k & 32) ? (31 - (k & 31)) : (k & 31)] +=
> mul32(c->history[channel][i], UnQMF[j]);
> +
> +resp = 0;
> +/* TODO: implement FFT instead of this naive calculation */
> +for (band = 0; band < DCA_SUBBANDS_32; band++) {
> +for (j = 0; j < 32; j++)
> +resp += mul32(accum[j], band_delta_factor(band, j));
> +
> +out[band] = (band & 2) ? (-resp) : resp;
> +}
> +}
I'll probably sound very naive, but what does this piece of the code do?
> +static void init_lfe_fir(void){
> +static int initialized;
> +int i;
> +if(initialized)
> +return;
> +for(i=0; i<512; i++)
> +lfe_fir_64i[i] = lfe_fir_64[i] * (1<<25); //float -> int32_t
> +initialized = 1;
> +}
This kind of stuff, along with cos tables, FFT etc, raises the
question why this wasn't implemented in float... Realistically, if
!CONFIG_SMALL, the table should be hardcoded.
The bitstream stuff I didn't review because I'm not very familiar with it...
> +static int DCA_encode_init(AVCodecContext *avctx) {
[..]
> +for(i=0; i<16; i++){
> +if(dca_sample_rates[i] == avctx->sample_rate)
> +break;
> +}
> +if(i==16){
> +av_log(avctx, AV_LOG_ERROR, "Sample rate %iHz not supported\n",
> avctx->sample_rate);
> +return -1;
> +}
> +c->sample_rate_code = i;
That is pretty unhelpful. What samplerates _are_ supported? Better
yet, how would the user accomplish the resampling within ffmpeg?
> diff --git a/libavcodec/dcaenc.h b/libavcodec/dcaenc.h
[..]
> +/* This is a scaled version of the response of the reference decoder to
> + this vector of subband samples: ( 1.0 0.0 0.0 ... 0.0 )
> + */
> +
> +static const int32_t UnQMF[512] = {
??? What is that?
Ronald
___
libav-devel mailing list
libav-devel@libav.org
https://lists.libav.org/mailman/listinfo/libav-devel
[libav-devel] [PATCH] Experimental DCA encoder
MvH
Benjamin Larsson
>From 72e6371e0c98e344275173452cec98940a33ef39 Mon Sep 17 00:00:00 2001
From: Alexander E. Patrakov
Date: Fri, 8 Apr 2011 01:33:21 +0200
Subject: [PATCH] Experimental DCA encoder
---
libavcodec/Makefile|1 +
libavcodec/allcodecs.c |2 +-
libavcodec/dcaenc.c| 581
libavcodec/dcaenc.h| 544 +
4 files changed, 1127 insertions(+), 1 deletions(-)
create mode 100644 libavcodec/dcaenc.c
create mode 100644 libavcodec/dcaenc.h
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 837f7e2..195469d 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -104,6 +104,7 @@ OBJS-$(CONFIG_COOK_DECODER)+= cook.o
OBJS-$(CONFIG_CSCD_DECODER)+= cscd.o
OBJS-$(CONFIG_CYUV_DECODER)+= cyuv.o
OBJS-$(CONFIG_DCA_DECODER) += dca.o synth_filter.o dcadsp.o
+OBJS-$(CONFIG_DCA_ENCODER) += dcaenc.o
OBJS-$(CONFIG_DFA_DECODER) += dfa.o
OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o \
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index 7636392..9530358 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -238,7 +238,7 @@ void avcodec_register_all(void)
REGISTER_DECODER (BINKAUDIO_DCT, binkaudio_dct);
REGISTER_DECODER (BINKAUDIO_RDFT, binkaudio_rdft);
REGISTER_DECODER (COOK, cook);
-REGISTER_DECODER (DCA, dca);
+REGISTER_ENCDEC (DCA, dca);
REGISTER_DECODER (DSICINAUDIO, dsicinaudio);
REGISTER_DECODER (EAC3, eac3);
REGISTER_ENCDEC (FLAC, flac);
diff --git a/libavcodec/dcaenc.c b/libavcodec/dcaenc.c
new file mode 100644
index 000..0362485
--- /dev/null
+++ b/libavcodec/dcaenc.c
@@ -0,0 +1,581 @@
+/*
+ * DCA encoder
+ * Copyright (C) 2008 Alexander E. Patrakov
+ * 2010 Benjamin Larsson
+ * 2011 Xiang Wang
+ * This file is part of Libav.
+ *
+ * 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/avassert.h"
+#include "libavutil/audioconvert.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "put_bits.h"
+#include "dcaenc.h"
+#include "dcadata.h"
+
+#undef NDEBUG
+
+#define MAX_CHANNELS 6
+#define DCA_SUBBANDS_32 32
+#define DCA_MAX_FRAME_SIZE 16383
+#define DCA_HEADER_SIZE 13
+
+#define DCA_SUBBANDS 32 ///< Subband activity count
+#define QUANTIZER_BITS 16
+#define SUBFRAMES 1
+#define SUBSUBFRAMES 4
+#define PCM_SAMPLES (SUBFRAMES*SUBSUBFRAMES*8)
+#define LFE_BITS 8
+#define LFE_INTERPOLATION 64
+#define LFE_PRESENT 2
+#define LFE_MISSING 0
+
+static const int8_t dca_lfe_index[] = {
+1,2,2,2,2,3,2,3,2,3,2,3,1,3,2,3
+};
+
+static const int8_t dca_channel_reorder_lfe[][9] = {
+{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, 2, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, 3, -1, -1, -1, -1},
+{ 0, 1, -1, 2, 3, -1, -1, -1, -1},
+{ 1, 2, 0, -1, 3, 4, -1, -1, -1},
+{ 2, 3, -1, 0, 1, 4, 5, -1, -1},
+{ 1, 2, 0, -1, 3, 4, 5, -1, -1},
+{ 0, -1, 4, 5, 2, 3, 1, -1, -1},
+{ 3, 4, 1, -1, 0, 2, 5, 6, -1},
+{ 2, 3, -1, 5, 7, 0, 1, 4, 6},
+{ 3, 4, 1, -1, 0, 2, 5, 7, 6},
+};
+
+static const int8_t dca_channel_reorder_nolfe[][9] = {
+{ 0, -1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 0, 1, -1, -1, -1, -1, -1, -1, -1},
+{ 1, 2, 0, -1, -1, -1, -1, -1, -1},
+{ 0, 1, 2, -1, -1, -1, -1, -1, -1},
+{ 1, 2, 0, 3, -1, -1, -1, -1, -1},
+{ 0, 1, 2, 3, -1, -1, -1, -1, -1},
+{ 1, 2, 0, 3, 4, -1, -1, -1, -1},
+{ 2, 3, 0, 1, 4, 5, -1, -1, -1},
+{ 1, 2, 0, 3, 4, 5, -1, -1, -1},
+{ 0, 4, 5, 2, 3, 1, -1, -1, -1},
+{ 3, 4, 1, 0, 2, 5, 6, -1, -1},
+{ 2, 3, 5, 7, 0, 1, 4, 6, -1},
+{ 3, 4, 1, 0, 2, 5, 7, 6, -1},
+};
+
+type
