January 2016 - ffmpeg-cvslog - lists.ffmpeg.org

build: use a link instead of changing current directory when compiling
by git＠videolan.org 02 Feb '16

02 Feb '16

ffmpeg | branch: master | Andreas Cadhalpun <Andreas.Cadhalpun(a)googlemail.com> | Mon Jan 25 01:42:23 2016 +0100| [b46aae093634271931395d65f422f4b2a23112d3] | committer: Andreas Cadhalpun build: use a link instead of changing current directory when compiling If links don't work, fall back to using the full source path as was previously done. This should fix build failures with MSVC. Reviewed-by: Hendrik Leppkes <h.leppkes(a)gmail.com> Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun(a)googlemail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b46aae09363427193… --- .gitignore | 1 + Makefile | 2 +- common.mak | 7 +++---- configure | 28 +++++++++++++++++++++------- library.mak | 2 +- 5 files changed, 27 insertions(+), 13 deletions(-) diff --git a/.gitignore b/.gitignore index 93b0dca..9fc0ac2 100644 --- a/.gitignore +++ b/.gitignore @@ -64,6 +64,7 @@ /libavcodec/*_tables.h /libavutil/avconfig.h /libavutil/ffversion.h +/src /tests/audiogen /tests/base64 /tests/checkasm/checkasm diff --git a/Makefile b/Makefile index f3bd5f6..e484249 100644 --- a/Makefile +++ b/Makefile @@ -181,7 +181,7 @@ clean:: distclean:: $(RM) $(DISTCLEANSUFFIXES) - $(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h + $(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h src $(RM) -rf doc/examples/pc-uninstalled config: diff --git a/common.mak b/common.mak index 3812149..03b51c5 100644 --- a/common.mak +++ b/common.mak @@ -32,7 +32,7 @@ endif ALLFFLIBS = avcodec avdevice avfilter avformat avresample avutil postproc swscale swresample # NASM requires -I path terminated with / -IFLAGS := -I$(DST_PATH)/ -I$(SRC_PATH)/ +IFLAGS := -I. -I$(SRC_LINK)/ CPPFLAGS := $(IFLAGS) $(CPPFLAGS) CFLAGS += $(ECFLAGS) CCFLAGS = $(CPPFLAGS) $(CFLAGS) @@ -43,12 +43,11 @@ CXXFLAGS += $(CPPFLAGS) $(CFLAGS) YASMFLAGS += $(IFLAGS:%=%/) -Pconfig.asm HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS) -LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)$(DST_PATH)/lib%) $(LDFLAGS) +LDFLAGS := $(ALLFFLIBS:%=$(LD_PATH)lib%) $(LDFLAGS) define COMPILE $(call $(1)DEP,$(1)) - $(Q)cd $(SRC_PATH); if [ -n "$(findstring $(SRC_PATH),$<)" ]; then dest=$(subst $(SRC_PATH)/,,$<); else dest=$(DST_PATH)/$<; fi; \ - $(subst @,,$($(1))) $($(1)FLAGS) $($(1)_DEPFLAGS:$(@:.o=.d)=$(DST_PATH)/$(@:.o=.d)) $($(1)_C) $($(1)_O:$@=$(DST_PATH)/$@) $$dest + $($(1)) $($(1)FLAGS) $($(1)_DEPFLAGS) $($(1)_C) $($(1)_O) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) endef COMPILE_C = $(call COMPILE,CC) diff --git a/configure b/configure index 0685133..57b526f 100755 --- a/configure +++ b/configure @@ -4696,6 +4696,26 @@ case $target_os in ;; esac +# test if creating links works +link_dest=$(mktemp -u $TMPDIR/dest_XXXXXXXX) +link_name=$(mktemp -u $TMPDIR/name_XXXXXXXX) +mkdir "$link_dest" +$ln_s "$link_dest" "$link_name" +touch "$link_dest/test_file" +if [ -e "$link_name/test_file" ]; then + # create link to source path + [ -e src ] && rm src + $ln_s "$source_path" src + source_link=src +else + # creating directory links doesn't work + # fall back to using the full source path + source_link="$source_path" +fi +# cleanup +rm -r "$link_dest" +rm -r "$link_name" + # determine libc flavour probe_libc(){ @@ -6229,12 +6249,6 @@ enabled stripping || strip="echo skipping strip" config_files="$TMPH config.mak doc/config.texi" -if enabled msvc; then - dst_path=$(pwd -W) -else - dst_path=$(pwd) -fi - cat > config.mak <<EOF # Automatically generated by configure - do not modify! ifndef FFMPEG_CONFIG_MAK @@ -6250,10 +6264,10 @@ DOCDIR=\$(DESTDIR)$docdir MANDIR=\$(DESTDIR)$mandir PKGCONFIGDIR=\$(DESTDIR)$pkgconfigdir SRC_PATH=$source_path +SRC_LINK=$source_link ifndef MAIN_MAKEFILE SRC_PATH:=\$(SRC_PATH:.%=..%) endif -DST_PATH=$dst_path CC_IDENT=$cc_ident ARCH=$arch INTRINSICS=$intrinsics diff --git a/library.mak b/library.mak index 0b23a28..3e1082c 100644 --- a/library.mak +++ b/library.mak @@ -28,7 +28,7 @@ $(SUBDIR)x86/%$(DEFAULT_YASMD).asm: $(SUBDIR)x86/%.asm $(SUBDIR)x86/%.o: $(SUBDIR)x86/%$(YASMD).asm $(DEPYASM) $(YASMFLAGS) -I $(<D)/ -M -o $@ $< > $(@:.o=.d) - $(Q)cd $(SRC_PATH); $(subst @,,$(YASM)) $(YASMFLAGS) -I $(<D)/ -o $(DST_PATH)/$@ $(subst $(SRC_PATH)/,,$<) + $(YASM) $(YASMFLAGS) -I $(<D)/ -o $@ $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) -$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@) LIBOBJS := $(OBJS) $(SUBDIR)%.h.o $(TESTOBJS)

3 2

Changelog: add entry on wtvdec performance improvement
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Timothy Gu <timothygu99(a)gmail.com> | Sun Jan 31 15:52:34 2016 -0800| [80075b0149049a131ea0d4e8b96c6c0695e128b7] | committer: Timothy Gu Changelog: add entry on wtvdec performance improvement > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=80075b0149049a131… --- Changelog | 1 + 1 file changed, 1 insertion(+) diff --git a/Changelog b/Changelog index 5b06ff1..04e044e 100644 --- a/Changelog +++ b/Changelog @@ -62,6 +62,7 @@ version <next>: - libaacplus and libvo-aacenc support removed - Cineform HD decoder - new DCA decoder with full support for DTS-HD extensions +- significant performance improvements in Windows Television (WTV) demuxer version 2.8:

1 0

wtvdec: Removed unused variable
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Timothy Gu <timothygu99(a)gmail.com> | Sun Jan 31 15:52:23 2016 -0800| [e5a6dcac47c2f778c69e443aa95b780baeedc46e] | committer: Timothy Gu wtvdec: Removed unused variable > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e5a6dcac47c2f778c… --- libavformat/wtvdec.c | 1 - 1 file changed, 1 deletion(-) diff --git a/libavformat/wtvdec.c b/libavformat/wtvdec.c index 1deb4d9..71deaf0 100644 --- a/libavformat/wtvdec.c +++ b/libavformat/wtvdec.c @@ -1027,7 +1027,6 @@ static int read_header(AVFormatContext *s) if (wtv->nb_index_entries) { pb = wtvfile_open(s, root, root_size, ff_timeline_table_0_entries_Events_le16); if (pb) { - int i; AVIndexEntry *e = wtv->index_entries; AVIndexEntry *e_end = wtv->index_entries + wtv->nb_index_entries - 1; uint64_t last_position = 0;

1 0

all: Make header guard names consistent
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Timothy Gu <timothygu99(a)gmail.com> | Thu Jan 28 18:25:13 2016 -0800| [180f9a09588d1c68983692bb5245213956fdde11] | committer: Timothy Gu all: Make header guard names consistent > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=180f9a09588d1c689… --- compat/aix/math.h | 6 +++--- compat/msvcrt/snprintf.h | 6 +++--- compat/os2threads.h | 6 +++--- compat/tms470/math.h | 6 +++--- compat/w32pthreads.h | 6 +++--- libavcodec/aacps.h | 6 +++--- libavcodec/aacps_fixed_tablegen.h | 6 +++--- libavcodec/aacps_tablegen.h | 6 +++--- libavcodec/aacpsdsp.h | 6 +++--- libavcodec/bswapdsp.h | 6 +++--- libavcodec/dxva2.h | 6 +++--- libavcodec/dxva2_internal.h | 6 +++--- libavcodec/g729postfilter.h | 6 +++--- libavcodec/gif.h | 6 +++--- libavcodec/lossless_audiodsp.h | 6 +++--- libavcodec/mips/aacdec_mips.h | 6 +++--- libavcodec/mips/aacsbr_mips.h | 6 +++--- libavcodec/mips/amrwbdec_mips.h | 6 +++--- libavcodec/mips/h264chroma_mips.h | 6 +++--- libavcodec/mips/h264dsp_mips.h | 6 +++--- libavcodec/mips/h264pred_mips.h | 6 +++--- libavcodec/mips/lsp_mips.h | 6 +++--- libavcodec/mips/mpegvideo_mips.h | 6 +++--- libavcodec/motion_est.h | 6 +++--- libavcodec/on2avcdata.h | 6 +++--- libavcodec/tscc2data.h | 6 +++--- libavcodec/vorbis_parser.h | 6 +++--- libavcodec/vorbis_parser_internal.h | 6 +++--- libavcodec/x86/diracdsp_mmx.h | 4 ++-- libavdevice/dshow_capture.h | 6 +++--- libavdevice/opengl_enc_shaders.h | 6 +++--- libavfilter/af_volume.h | 6 +++--- libavfilter/avf_showcqt.h | 4 ++-- libavfilter/framepool.h | 6 +++--- libavfilter/opencl_allkernels.h | 6 +++--- libavfilter/psnr.h | 6 +++--- libavfilter/ssim.h | 6 +++--- libavfilter/stereo3d.h | 6 +++--- libavfilter/vf_hqdn3d.h | 6 +++--- libavutil/mips/libm_mips.h | 6 +++--- libavutil/opencl.h | 6 +++--- libpostproc/version.h | 6 +++--- libswresample/audioconvert.h | 6 +++--- libswresample/swresample_internal.h | 4 ++-- libswresample/version.h | 6 +++--- tests/checkasm/checkasm.h | 4 ++-- 46 files changed, 134 insertions(+), 134 deletions(-) diff --git a/compat/aix/math.h b/compat/aix/math.h index 65a89c4..dee13c8 100644 --- a/compat/aix/math.h +++ b/compat/aix/math.h @@ -19,8 +19,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_COMPAT_AIX_MATH_H -#define FFMPEG_COMPAT_AIX_MATH_H +#ifndef COMPAT_AIX_MATH_H +#define COMPAT_AIX_MATH_H #define class class_in_math_h_causes_problems @@ -28,4 +28,4 @@ #undef class -#endif /* FFMPEG_COMPAT_AIX_MATH_H */ +#endif /* COMPAT_AIX_MATH_H */ diff --git a/compat/msvcrt/snprintf.h b/compat/msvcrt/snprintf.h index f02113c..cd47953 100644 --- a/compat/msvcrt/snprintf.h +++ b/compat/msvcrt/snprintf.h @@ -19,8 +19,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef COMPAT_SNPRINTF_H -#define COMPAT_SNPRINTF_H +#ifndef COMPAT_MSVCRT_SNPRINTF_H +#define COMPAT_MSVCRT_SNPRINTF_H #include <stdarg.h> #include <stdio.h> @@ -35,4 +35,4 @@ int avpriv_vsnprintf(char *s, size_t n, const char *fmt, va_list ap); #define _snprintf avpriv_snprintf #define vsnprintf avpriv_vsnprintf -#endif /* COMPAT_SNPRINTF_H */ +#endif /* COMPAT_MSVCRT_SNPRINTF_H */ diff --git a/compat/os2threads.h b/compat/os2threads.h index f8fb2c8..7c0fe13 100644 --- a/compat/os2threads.h +++ b/compat/os2threads.h @@ -23,8 +23,8 @@ * os2threads to pthreads wrapper */ -#ifndef AVCODEC_OS2PTHREADS_H -#define AVCODEC_OS2PTHREADS_H +#ifndef COMPAT_OS2THREADS_H +#define COMPAT_OS2THREADS_H #define INCL_DOS #include <os2.h> @@ -190,4 +190,4 @@ static av_always_inline int pthread_once(pthread_once_t *once_control, void (*in return 0; } -#endif /* AVCODEC_OS2PTHREADS_H */ +#endif /* COMPAT_OS2THREADS_H */ diff --git a/compat/tms470/math.h b/compat/tms470/math.h index 6234cc5..0a42743 100644 --- a/compat/tms470/math.h +++ b/compat/tms470/math.h @@ -16,8 +16,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_COMPAT_TMS470_MATH_H -#define FFMPEG_COMPAT_TMS470_MATH_H +#ifndef COMPAT_TMS470_MATH_H +#define COMPAT_TMS470_MATH_H #include_next <math.h> @@ -27,4 +27,4 @@ #define INFINITY (*(const float*)((const unsigned []){ 0x7f800000 })) #define NAN (*(const float*)((const unsigned []){ 0x7fc00000 })) -#endif /* FFMPEG_COMPAT_TMS470_MATH_H */ +#endif /* COMPAT_TMS470_MATH_H */ diff --git a/compat/w32pthreads.h b/compat/w32pthreads.h index c327962..4ac2a99 100644 --- a/compat/w32pthreads.h +++ b/compat/w32pthreads.h @@ -26,8 +26,8 @@ * w32threads to pthreads wrapper */ -#ifndef FFMPEG_COMPAT_W32PTHREADS_H -#define FFMPEG_COMPAT_W32PTHREADS_H +#ifndef COMPAT_W32PTHREADS_H +#define COMPAT_W32PTHREADS_H /* Build up a pthread-like API using underlying Windows API. Have only static * methods so as to not conflict with a potentially linked in pthread-win32 @@ -415,4 +415,4 @@ static av_unused void w32thread_init(void) } -#endif /* FFMPEG_COMPAT_W32PTHREADS_H */ +#endif /* COMPAT_W32PTHREADS_H */ diff --git a/libavcodec/aacps.h b/libavcodec/aacps.h index 54f9d99..61edce3 100644 --- a/libavcodec/aacps.h +++ b/libavcodec/aacps.h @@ -19,8 +19,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_PS_H -#define AVCODEC_PS_H +#ifndef AVCODEC_AACPS_H +#define AVCODEC_AACPS_H #include <stdint.h> @@ -83,4 +83,4 @@ void AAC_RENAME(ff_ps_ctx_init)(PSContext *ps); int AAC_RENAME(ff_ps_read_data)(AVCodecContext *avctx, GetBitContext *gb, PSContext *ps, int bits_left); int AAC_RENAME(ff_ps_apply)(AVCodecContext *avctx, PSContext *ps, INTFLOAT L[2][38][64], INTFLOAT R[2][38][64], int top); -#endif /* AVCODEC_PS_H */ +#endif /* AVCODEC_AACPS_H */ diff --git a/libavcodec/aacps_fixed_tablegen.h b/libavcodec/aacps_fixed_tablegen.h index 01f2ece..8b82deb 100644 --- a/libavcodec/aacps_fixed_tablegen.h +++ b/libavcodec/aacps_fixed_tablegen.h @@ -23,8 +23,8 @@ * */ -#ifndef AACPS_FIXED_TABLEGEN_H -#define AACPS_FIXED_TABLEGEN_H +#ifndef AVCODEC_AACPS_FIXED_TABLEGEN_H +#define AVCODEC_AACPS_FIXED_TABLEGEN_H #include <math.h> #include <stdint.h> @@ -400,4 +400,4 @@ static void ps_tableinit(void) } #endif /* CONFIG_HARDCODED_TABLES */ -#endif /* AACPS_FIXED_TABLEGEN_H */ +#endif /* AVCODEC_AACPS_FIXED_TABLEGEN_H */ diff --git a/libavcodec/aacps_tablegen.h b/libavcodec/aacps_tablegen.h index 0bd51cc..0ac4f68 100644 --- a/libavcodec/aacps_tablegen.h +++ b/libavcodec/aacps_tablegen.h @@ -20,8 +20,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AACPS_TABLEGEN_H -#define AACPS_TABLEGEN_H +#ifndef AVCODEC_AACPS_TABLEGEN_H +#define AVCODEC_AACPS_TABLEGEN_H #include <math.h> #include <stdint.h> @@ -214,4 +214,4 @@ static av_cold void ps_tableinit(void) } #endif /* CONFIG_HARDCODED_TABLES */ -#endif /* AACPS_TABLEGEN_H */ +#endif /* AVCODEC_AACPS_TABLEGEN_H */ diff --git a/libavcodec/aacpsdsp.h b/libavcodec/aacpsdsp.h index c194bbe..ad9bbb8 100644 --- a/libavcodec/aacpsdsp.h +++ b/libavcodec/aacpsdsp.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef LIBAVCODEC_AACPSDSP_H -#define LIBAVCODEC_AACPSDSP_H +#ifndef AVCODEC_AACPSDSP_H +#define AVCODEC_AACPSDSP_H #include "aac_defines.h" @@ -54,4 +54,4 @@ void ff_psdsp_init_arm(PSDSPContext *s); void ff_psdsp_init_mips(PSDSPContext *s); void ff_psdsp_init_x86(PSDSPContext *s); -#endif /* LIBAVCODEC_AACPSDSP_H */ +#endif /* AVCODEC_AACPSDSP_H */ diff --git a/libavcodec/bswapdsp.h b/libavcodec/bswapdsp.h index f167d77..4d19092 100644 --- a/libavcodec/bswapdsp.h +++ b/libavcodec/bswapdsp.h @@ -16,8 +16,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_BSWAP_BUF_H -#define AVCODEC_BSWAP_BUF_H +#ifndef AVCODEC_BSWAPDSP_H +#define AVCODEC_BSWAPDSP_H #include <stdint.h> @@ -29,4 +29,4 @@ typedef struct BswapDSPContext { void ff_bswapdsp_init(BswapDSPContext *c); void ff_bswapdsp_init_x86(BswapDSPContext *c); -#endif /* AVCODEC_BSWAP_BUF_H */ +#endif /* AVCODEC_BSWAPDSP_H */ diff --git a/libavcodec/dxva2.h b/libavcodec/dxva2.h index 9e3ab86..22c9399 100644 --- a/libavcodec/dxva2.h +++ b/libavcodec/dxva2.h @@ -20,8 +20,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_DXVA_H -#define AVCODEC_DXVA_H +#ifndef AVCODEC_DXVA2_H +#define AVCODEC_DXVA2_H /** * @file @@ -90,4 +90,4 @@ struct dxva_context { * @} */ -#endif /* AVCODEC_DXVA_H */ +#endif /* AVCODEC_DXVA2_H */ diff --git a/libavcodec/dxva2_internal.h b/libavcodec/dxva2_internal.h index c2499e5..ad89f82 100644 --- a/libavcodec/dxva2_internal.h +++ b/libavcodec/dxva2_internal.h @@ -20,8 +20,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_DXVA_INTERNAL_H -#define AVCODEC_DXVA_INTERNAL_H +#ifndef AVCODEC_DXVA2_INTERNAL_H +#define AVCODEC_DXVA2_INTERNAL_H #define COBJMACROS @@ -118,4 +118,4 @@ int ff_dxva2_common_end_frame(AVCodecContext *, AVFrame *, DECODER_BUFFER_DESC *bs, DECODER_BUFFER_DESC *slice)); -#endif /* AVCODEC_DXVA_INTERNAL_H */ +#endif /* AVCODEC_DXVA2_INTERNAL_H */ diff --git a/libavcodec/g729postfilter.h b/libavcodec/g729postfilter.h index 89e3e40..5c2aaf2 100644 --- a/libavcodec/g729postfilter.h +++ b/libavcodec/g729postfilter.h @@ -18,8 +18,8 @@ * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef FFMPEG_G729POSTFILTER_H -#define FFMPEG_G729POSTFILTER_H +#ifndef AVCODEC_G729POSTFILTER_H +#define AVCODEC_G729POSTFILTER_H #include <stdint.h> #include "audiodsp.h" @@ -113,4 +113,4 @@ void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voici int16_t ff_g729_adaptive_gain_control(int gain_before, int gain_after, int16_t *speech, int subframe_size, int16_t gain_prev); -#endif // FFMPEG_G729POSTFILTER_H +#endif // AVCODEC_G729POSTFILTER_H diff --git a/libavcodec/gif.h b/libavcodec/gif.h index b4cf665..9f35778 100644 --- a/libavcodec/gif.h +++ b/libavcodec/gif.h @@ -26,8 +26,8 @@ * GIF format definitions. */ -#ifndef AVCODEC_GIFDEFS_H -#define AVCODEC_GIFDEFS_H +#ifndef AVCODEC_GIF_H +#define AVCODEC_GIF_H #include <stdint.h> @@ -46,4 +46,4 @@ static const uint8_t gif89a_sig[6] = "GIF89a"; #define GIF_APP_EXT_LABEL 0xff #define NETSCAPE_EXT_STR "NETSCAPE2.0" -#endif /* AVCODEC_GIFDEFS_H */ +#endif /* AVCODEC_GIF_H */ diff --git a/libavcodec/lossless_audiodsp.h b/libavcodec/lossless_audiodsp.h index 9ce2e63..79ca30d 100644 --- a/libavcodec/lossless_audiodsp.h +++ b/libavcodec/lossless_audiodsp.h @@ -20,8 +20,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_LLAUDDSP_H -#define AVCODEC_LLAUDDSP_H +#ifndef AVCODEC_LOSSLESS_AUDIODSP_H +#define AVCODEC_LOSSLESS_AUDIODSP_H #include <stdint.h> @@ -43,4 +43,4 @@ void ff_llauddsp_init_arm(LLAudDSPContext *c); void ff_llauddsp_init_ppc(LLAudDSPContext *c); void ff_llauddsp_init_x86(LLAudDSPContext *c); -#endif /* AVCODEC_LLAUDDSP_H */ +#endif /* AVCODEC_LOSSLESS_AUDIODSP_H */ diff --git a/libavcodec/mips/aacdec_mips.h b/libavcodec/mips/aacdec_mips.h index 054a9fb..c4f6438 100644 --- a/libavcodec/mips/aacdec_mips.h +++ b/libavcodec/mips/aacdec_mips.h @@ -54,8 +54,8 @@ * Reference: libavcodec/aacdec.c */ -#ifndef AVCODEC_MIPS_AACDEC_FLOAT_H -#define AVCODEC_MIPS_AACDEC_FLOAT_H +#ifndef AVCODEC_MIPS_AACDEC_MIPS_H +#define AVCODEC_MIPS_AACDEC_MIPS_H #include "libavcodec/aac.h" #include "libavutil/mips/asmdefs.h" @@ -248,4 +248,4 @@ static inline float *VMUL4S_mips(float *dst, const float *v, unsigned idx, #define VMUL4S VMUL4S_mips #endif /* HAVE_INLINE_ASM && HAVE_MIPSFPU */ -#endif /* AVCODEC_MIPS_AACDEC_FLOAT_H */ +#endif /* AVCODEC_MIPS_AACDEC_MIPS_H */ diff --git a/libavcodec/mips/aacsbr_mips.h b/libavcodec/mips/aacsbr_mips.h index da8389f..13e55d2 100644 --- a/libavcodec/mips/aacsbr_mips.h +++ b/libavcodec/mips/aacsbr_mips.h @@ -51,8 +51,8 @@ * Reference: libavcodec/aacsbr.c */ -#ifndef AVCODEC_MIPS_AACSBR_FLOAT_H -#define AVCODEC_MIPS_AACSBR_FLOAT_H +#ifndef AVCODEC_MIPS_AACSBR_MIPS_H +#define AVCODEC_MIPS_AACSBR_MIPS_H #include "libavcodec/aac.h" #include "libavcodec/sbr.h" @@ -491,4 +491,4 @@ static void sbr_qmf_synthesis_mips(FFTContext *mdct, #endif /* HAVE_MIPSFPU */ #endif /* HAVE_INLINE_ASM */ -#endif /* AVCODEC_MIPS_AACSBR_FLOAT_H */ +#endif /* AVCODEC_MIPS_AACSBR_MIPS_H */ diff --git a/libavcodec/mips/amrwbdec_mips.h b/libavcodec/mips/amrwbdec_mips.h index b3355fb..ec46f95 100644 --- a/libavcodec/mips/amrwbdec_mips.h +++ b/libavcodec/mips/amrwbdec_mips.h @@ -49,8 +49,8 @@ * @file * Reference: libavcodec/amrwbdec.c */ -#ifndef AVCODEC_AMRWBDEC_MIPS_H -#define AVCODEC_AMRWBDEC_MIPS_H +#ifndef AVCODEC_MIPS_AMRWBDEC_MIPS_H +#define AVCODEC_MIPS_AMRWBDEC_MIPS_H #include "config.h" #if HAVE_MIPSFPU && HAVE_INLINE_ASM @@ -59,4 +59,4 @@ void ff_hb_fir_filter_mips(float *out, const float fir_coef[], #define hb_fir_filter ff_hb_fir_filter_mips #endif -#endif /* AVCODEC_AMRWBDEC_MIPS_H */ +#endif /* AVCODEC_MIPS_AMRWBDEC_MIPS_H */ diff --git a/libavcodec/mips/h264chroma_mips.h b/libavcodec/mips/h264chroma_mips.h index 7a373b8..0ef6c74 100644 --- a/libavcodec/mips/h264chroma_mips.h +++ b/libavcodec/mips/h264chroma_mips.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef H264_CHROMA_MIPS_H -#define H264_CHROMA_MIPS_H +#ifndef AVCODEC_MIPS_H264CHROMA_MIPS_H +#define AVCODEC_MIPS_H264CHROMA_MIPS_H #include "libavcodec/h264.h" void ff_put_h264_chroma_mc8_msa(uint8_t *dst, uint8_t *src, int stride, @@ -44,4 +44,4 @@ void ff_put_h264_chroma_mc4_mmi(uint8_t *dst, uint8_t *src, int stride, void ff_avg_h264_chroma_mc4_mmi(uint8_t *dst, uint8_t *src, int stride, int h, int x, int y); -#endif /* H264_CHROMA_MIPS_H */ +#endif /* AVCODEC_MIPS_H264CHROMA_MIPS_H */ diff --git a/libavcodec/mips/h264dsp_mips.h b/libavcodec/mips/h264dsp_mips.h index 3fdbf4f..2fdfd11 100644 --- a/libavcodec/mips/h264dsp_mips.h +++ b/libavcodec/mips/h264dsp_mips.h @@ -19,8 +19,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef H264_DSP_MIPS_H -#define H264_DSP_MIPS_H +#ifndef AVCODEC_MIPS_H264DSP_MIPS_H +#define AVCODEC_MIPS_H264DSP_MIPS_H #include "libavcodec/h264.h" #include "constants.h" @@ -574,4 +574,4 @@ void ff_avg_h264_qpel4_mc23_mmi(uint8_t *dst, const uint8_t *src, void ff_avg_h264_qpel4_mc33_mmi(uint8_t *dst, const uint8_t *src, ptrdiff_t dst_stride); -#endif // #ifndef H264_DSP_MIPS_H +#endif // #ifndef AVCODEC_MIPS_H264DSP_MIPS_H diff --git a/libavcodec/mips/h264pred_mips.h b/libavcodec/mips/h264pred_mips.h index d7d12c5..136e291 100644 --- a/libavcodec/mips/h264pred_mips.h +++ b/libavcodec/mips/h264pred_mips.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef H264_PRED_MIPS_H -#define H264_PRED_MIPS_H +#ifndef AVCODEC_MIPS_H264PRED_MIPS_H +#define AVCODEC_MIPS_H264PRED_MIPS_H #include "constants.h" #include "libavcodec/h264pred.h" @@ -45,4 +45,4 @@ void ff_pred8x8_dc_8_mmi(uint8_t *src, ptrdiff_t stride); void ff_pred8x16_vertical_8_mmi(uint8_t *src, ptrdiff_t stride); void ff_pred8x16_horizontal_8_mmi(uint8_t *src, ptrdiff_t stride); -#endif /* H264_PRED_MIPS_H */ +#endif /* AVCODEC_MIPS_H264PRED_MIPS_H */ diff --git a/libavcodec/mips/lsp_mips.h b/libavcodec/mips/lsp_mips.h index 9d75218..1e899bc 100644 --- a/libavcodec/mips/lsp_mips.h +++ b/libavcodec/mips/lsp_mips.h @@ -51,8 +51,8 @@ * @file * Reference: libavcodec/lsp.c */ -#ifndef AVCODEC_LSP_MIPS_H -#define AVCODEC_LSP_MIPS_H +#ifndef AVCODEC_MIPS_LSP_MIPS_H +#define AVCODEC_MIPS_LSP_MIPS_H #if HAVE_MIPSFPU && HAVE_INLINE_ASM #include "libavutil/mips/asmdefs.h" @@ -108,4 +108,4 @@ static av_always_inline void ff_lsp2polyf_mips(const double *lsp, double *f, int } #define ff_lsp2polyf ff_lsp2polyf_mips #endif /* HAVE_MIPSFPU && HAVE_INLINE_ASM */ -#endif /* AVCODEC_LSP_MIPS_H */ +#endif /* AVCODEC_MIPS_LSP_MIPS_H */ diff --git a/libavcodec/mips/mpegvideo_mips.h b/libavcodec/mips/mpegvideo_mips.h index decacd4c..760d7b3 100644 --- a/libavcodec/mips/mpegvideo_mips.h +++ b/libavcodec/mips/mpegvideo_mips.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef MPEGVIDEO_MIPS_H -#define MPEGVIDEO_MIPS_H +#ifndef AVCODEC_MIPS_MPEGVIDEO_MIPS_H +#define AVCODEC_MIPS_MPEGVIDEO_MIPS_H #include "libavcodec/mpegvideo.h" @@ -35,4 +35,4 @@ void ff_dct_unquantize_mpeg2_intra_mmi(MpegEncContext *s, int16_t *block, int n, int qscale); void ff_denoise_dct_mmi(MpegEncContext *s, int16_t *block); -#endif /* MPEGVIDEO_MIPS_H */ +#endif /* AVCODEC_MIPS_MPEGVIDEO_MIPS_H */ diff --git a/libavcodec/motion_est.h b/libavcodec/motion_est.h index 2d3c02a..3b3a8d7 100644 --- a/libavcodec/motion_est.h +++ b/libavcodec/motion_est.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_MOTIONEST_H -#define AVCODEC_MOTIONEST_H +#ifndef AVCODEC_MOTION_EST_H +#define AVCODEC_MOTION_EST_H #include <stdint.h> @@ -132,4 +132,4 @@ void ff_fix_long_mvs(struct MpegEncContext *s, uint8_t *field_select_table, int field_select, int16_t (*mv_table)[2], int f_code, int type, int truncate); -#endif /* AVCODEC_MOTIONEST_H */ +#endif /* AVCODEC_MOTION_EST_H */ diff --git a/libavcodec/on2avcdata.h b/libavcodec/on2avcdata.h index 7f498e5..95d88e0 100644 --- a/libavcodec/on2avcdata.h +++ b/libavcodec/on2avcdata.h @@ -20,8 +20,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_ON2AVC_DATA_H -#define AVCODEC_ON2AVC_DATA_H +#ifndef AVCODEC_ON2AVCDATA_H +#define AVCODEC_ON2AVCDATA_H #include <stdint.h> @@ -79,4 +79,4 @@ extern const float ff_on2avc_ctab_2[2048]; extern const float ff_on2avc_ctab_3[2048]; extern const float ff_on2avc_ctab_4[2048]; -#endif /* AVCODEC_ON2AVC_DATA_H */ +#endif /* AVCODEC_ON2AVCDATA_H */ diff --git a/libavcodec/tscc2data.h b/libavcodec/tscc2data.h index 4586da7..7806267 100644 --- a/libavcodec/tscc2data.h +++ b/libavcodec/tscc2data.h @@ -19,8 +19,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_TSCC2_DATA_H -#define AVCODEC_TSCC2_DATA_H +#ifndef AVCODEC_TSCC2DATA_H +#define AVCODEC_TSCC2DATA_H #include <stdint.h> @@ -932,4 +932,4 @@ static const uint8_t *tscc2_ac_vlc_bits[NUM_VLC_SETS] = { ac_vlc_descC_bits, }; -#endif /* AVCODEC_TSCC2_DATA_H */ +#endif /* AVCODEC_TSCC2DATA_H */ diff --git a/libavcodec/vorbis_parser.h b/libavcodec/vorbis_parser.h index 06e48bd..81fda3b 100644 --- a/libavcodec/vorbis_parser.h +++ b/libavcodec/vorbis_parser.h @@ -24,8 +24,8 @@ * Determines the duration for each packet. */ -#ifndef AVCODEC_VORBIS_PARSE_H -#define AVCODEC_VORBIS_PARSE_H +#ifndef AVCODEC_VORBIS_PARSER_H +#define AVCODEC_VORBIS_PARSER_H #include <stdint.h> @@ -75,4 +75,4 @@ int av_vorbis_parse_frame(AVVorbisParseContext *s, const uint8_t *buf, void av_vorbis_parse_reset(AVVorbisParseContext *s); -#endif /* AVCODEC_VORBIS_PARSE_H */ +#endif /* AVCODEC_VORBIS_PARSER_H */ diff --git a/libavcodec/vorbis_parser_internal.h b/libavcodec/vorbis_parser_internal.h index 98e4852..691a842 100644 --- a/libavcodec/vorbis_parser_internal.h +++ b/libavcodec/vorbis_parser_internal.h @@ -25,8 +25,8 @@ * Determines the duration for each packet. */ -#ifndef AVCODEC_VORBIS_PARSER_H -#define AVCODEC_VORBIS_PARSER_H +#ifndef AVCODEC_VORBIS_PARSER_INTERNAL_H +#define AVCODEC_VORBIS_PARSER_INTERNAL_H #include "avcodec.h" #include "vorbis_parser.h" @@ -43,4 +43,4 @@ struct AVVorbisParseContext { int prev_mask; ///< bitmask used to get the previous mode flag in each packet }; -#endif /* AVCODEC_VORBIS_PARSER_H */ +#endif /* AVCODEC_VORBIS_PARSER_INTERNAL_H */ diff --git a/libavcodec/x86/diracdsp_mmx.h b/libavcodec/x86/diracdsp_mmx.h index 8985854..d15dc91 100644 --- a/libavcodec/x86/diracdsp_mmx.h +++ b/libavcodec/x86/diracdsp_mmx.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVCODEC_X86_DIRACDSP_H -#define AVCODEC_X86_DIRACDSP_H +#ifndef AVCODEC_X86_DIRACDSP_MMX_H +#define AVCODEC_X86_DIRACDSP_MMX_H #include "libavcodec/diracdsp.h" diff --git a/libavdevice/dshow_capture.h b/libavdevice/dshow_capture.h index b17da10..f26eaf9 100644 --- a/libavdevice/dshow_capture.h +++ b/libavdevice/dshow_capture.h @@ -19,8 +19,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVDEVICE_DSHOW_H -#define AVDEVICE_DSHOW_H +#ifndef AVDEVICE_DSHOW_CAPTURE_H +#define AVDEVICE_DSHOW_CAPTURE_H #define DSHOWDEBUG 0 @@ -349,4 +349,4 @@ HRESULT dshow_try_setup_crossbar_options(ICaptureGraphBuilder2 *graph_builder2, void dshow_show_filter_properties(IBaseFilter *pFilter, AVFormatContext *avctx); -#endif /* AVDEVICE_DSHOW_H */ +#endif /* AVDEVICE_DSHOW_CAPTURE_H */ diff --git a/libavdevice/opengl_enc_shaders.h b/libavdevice/opengl_enc_shaders.h index ed8b3d3..67ee0ae 100644 --- a/libavdevice/opengl_enc_shaders.h +++ b/libavdevice/opengl_enc_shaders.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVDEVICE_OPENGL_SHADERS_H -#define AVDEVICE_OPENGL_SHADERS_H +#ifndef AVDEVICE_OPENGL_ENC_SHADERS_H +#define AVDEVICE_OPENGL_ENC_SHADERS_H #include "libavutil/pixfmt.h" @@ -185,4 +185,4 @@ static const char * const FF_OPENGL_FRAGMENT_SHADER_GRAY = "gl_FragColor = vec4(c, c, c, 1.0);" "}"; -#endif /* AVDEVICE_OPENGL_SHADERS_H */ +#endif /* AVDEVICE_OPENGL_ENC_SHADERS_H */ diff --git a/libavfilter/af_volume.h b/libavfilter/af_volume.h index aff7526..af46e34 100644 --- a/libavfilter/af_volume.h +++ b/libavfilter/af_volume.h @@ -21,8 +21,8 @@ * audio volume filter */ -#ifndef AVFILTER_AF_VOLUME_H -#define AVFILTER_AF_VOLUME_H +#ifndef AVFILTER_VOLUME_H +#define AVFILTER_VOLUME_H #include "libavutil/common.h" #include "libavutil/eval.h" @@ -90,4 +90,4 @@ typedef struct VolumeContext { void ff_volume_init_x86(VolumeContext *vol); -#endif /* AVFILTER_AF_VOLUME_H */ +#endif /* AVFILTER_VOLUME_H */ diff --git a/libavfilter/avf_showcqt.h b/libavfilter/avf_showcqt.h index a018075..a48b2b0 100644 --- a/libavfilter/avf_showcqt.h +++ b/libavfilter/avf_showcqt.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVFILTER_AVF_SHOWCQT_H -#define AVFILTER_AVF_SHOWCQT_H +#ifndef AVFILTER_SHOWCQT_H +#define AVFILTER_SHOWCQT_H #include "libavcodec/avfft.h" #include "avfilter.h" diff --git a/libavfilter/framepool.h b/libavfilter/framepool.h index bb6bb10..2a6c9e8 100644 --- a/libavfilter/framepool.h +++ b/libavfilter/framepool.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVUTIL_FRAME_POOL_H -#define AVUTIL_FRAME_POOL_H +#ifndef AVFILTER_FRAMEPOOL_H +#define AVFILTER_FRAMEPOOL_H #include "libavutil/buffer.h" #include "libavutil/frame.h" @@ -81,4 +81,4 @@ int ff_video_frame_pool_get_config(FFVideoFramePool *pool, AVFrame *ff_video_frame_pool_get(FFVideoFramePool *pool); -#endif /* AVUTIL_FRAME_POOL_H */ +#endif /* AVFILTER_FRAMEPOOL_H */ diff --git a/libavfilter/opencl_allkernels.h b/libavfilter/opencl_allkernels.h index aca02e0..57b650d 100644 --- a/libavfilter/opencl_allkernels.h +++ b/libavfilter/opencl_allkernels.h @@ -18,12 +18,12 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef AVFILTER_OPENCL_ALLKERNEL_H -#define AVFILTER_OPENCL_ALLKERNEL_H +#ifndef AVFILTER_OPENCL_ALLKERNELS_H +#define AVFILTER_OPENCL_ALLKERNELS_H #include "avfilter.h" #include "config.h" void ff_opencl_register_filter_kernel_code_all(void); -#endif /* AVFILTER_OPENCL_ALLKERNEL_H */ +#endif /* AVFILTER_OPENCL_ALLKERNELS_H */ diff --git a/libavfilter/psnr.h b/libavfilter/psnr.h index efe94da..bbc4541 100644 --- a/libavfilter/psnr.h +++ b/libavfilter/psnr.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef LIBAVFILTER_PSNR_H -#define LIBAVFILTER_PSNR_H +#ifndef AVFILTER_PSNR_H +#define AVFILTER_PSNR_H #include <stddef.h> #include <stdint.h> @@ -30,4 +30,4 @@ typedef struct PSNRDSPContext { void ff_psnr_init_x86(PSNRDSPContext *dsp, int bpp); -#endif /* LIBAVFILTER_PSNR_H */ +#endif /* AVFILTER_PSNR_H */ diff --git a/libavfilter/ssim.h b/libavfilter/ssim.h index cd3a6ee..ac0395a 100644 --- a/libavfilter/ssim.h +++ b/libavfilter/ssim.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef LIBAVFILTER_SSIM_H -#define LIBAVFILTER_SSIM_H +#ifndef AVFILTER_SSIM_H +#define AVFILTER_SSIM_H #include <stddef.h> #include <stdint.h> @@ -33,4 +33,4 @@ typedef struct SSIMDSPContext { void ff_ssim_init_x86(SSIMDSPContext *dsp); -#endif /* LIBAVFILTER_SSIM_H */ +#endif /* AVFILTER_SSIM_H */ diff --git a/libavfilter/stereo3d.h b/libavfilter/stereo3d.h index e7a8456..54611d1 100644 --- a/libavfilter/stereo3d.h +++ b/libavfilter/stereo3d.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef LIBAVFILTER_STEREO3D_H -#define LIBAVFILTER_STEREO3D_H +#ifndef AVFILTER_STEREO3D_H +#define AVFILTER_STEREO3D_H #include <stddef.h> #include <stdint.h> @@ -33,4 +33,4 @@ typedef struct Stereo3DDSPContext { void ff_stereo3d_init_x86(Stereo3DDSPContext *dsp); -#endif /* LIBAVFILTER_STEREO3D_H */ +#endif /* AVFILTER_STEREO3D_H */ diff --git a/libavfilter/vf_hqdn3d.h b/libavfilter/vf_hqdn3d.h index be55400..03a79a1 100644 --- a/libavfilter/vf_hqdn3d.h +++ b/libavfilter/vf_hqdn3d.h @@ -20,8 +20,8 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#ifndef AVFILTER_VF_HQDN3D_H -#define AVFILTER_VF_HQDN3D_H +#ifndef AVFILTER_HQDN3D_H +#define AVFILTER_HQDN3D_H #include <stddef.h> #include <stdint.h> @@ -46,4 +46,4 @@ typedef struct HQDN3DContext { void ff_hqdn3d_init_x86(HQDN3DContext *hqdn3d); -#endif /* AVFILTER_VF_HQDN3D_H */ +#endif /* AVFILTER_HQDN3D_H */ diff --git a/libavutil/mips/libm_mips.h b/libavutil/mips/libm_mips.h index 8853bbc..757867b 100644 --- a/libavutil/mips/libm_mips.h +++ b/libavutil/mips/libm_mips.h @@ -50,8 +50,8 @@ * MIPS optimization for some libm functions */ -#ifndef AVUTIL_LIBM_MIPS_H -#define AVUTIL_LIBM_MIPS_H +#ifndef AVUTIL_MIPS_LIBM_MIPS_H +#define AVUTIL_MIPS_LIBM_MIPS_H static av_always_inline av_const long int lrintf_mips(float x) { @@ -70,4 +70,4 @@ static av_always_inline av_const long int lrintf_mips(float x) #define lrintf(x) lrintf_mips(x) #define HAVE_LRINTF 1 -#endif /* AVUTIL_LIBM_MIPS_H */ +#endif /* AVUTIL_MIPS_LIBM_MIPS_H */ diff --git a/libavutil/opencl.h b/libavutil/opencl.h index 168461d..b709927 100644 --- a/libavutil/opencl.h +++ b/libavutil/opencl.h @@ -29,8 +29,8 @@ * change without prior notice. */ -#ifndef LIBAVUTIL_OPENCL_H -#define LIBAVUTIL_OPENCL_H +#ifndef AVUTIL_OPENCL_H +#define AVUTIL_OPENCL_H #define CL_USE_DEPRECATED_OPENCL_1_2_APIS 1 #ifdef __APPLE__ @@ -289,4 +289,4 @@ void av_opencl_uninit(void); int64_t av_opencl_benchmark(AVOpenCLDeviceNode *device, cl_platform_id platform, int64_t (*benchmark)(AVOpenCLExternalEnv *ext_opencl_env)); -#endif /* LIBAVUTIL_OPENCL_H */ +#endif /* AVUTIL_OPENCL_H */ diff --git a/libpostproc/version.h b/libpostproc/version.h index db557a5..8f62582 100644 --- a/libpostproc/version.h +++ b/libpostproc/version.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef POSTPROC_POSTPROCESS_VERSION_H -#define POSTPROC_POSTPROCESS_VERSION_H +#ifndef POSTPROC_VERSION_H +#define POSTPROC_VERSION_H /** * @file @@ -46,4 +46,4 @@ #define FF_API_QP_TYPE (LIBPOSTPROC_VERSION_MAJOR < 55) #endif -#endif /* POSTPROC_POSTPROCESS_VERSION_H */ +#endif /* POSTPROC_VERSION_H */ diff --git a/libswresample/audioconvert.h b/libswresample/audioconvert.h index 2e983df..1ca30c2 100644 --- a/libswresample/audioconvert.h +++ b/libswresample/audioconvert.h @@ -20,8 +20,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef SWR_AUDIOCONVERT_H -#define SWR_AUDIOCONVERT_H +#ifndef SWRESAMPLE_AUDIOCONVERT_H +#define SWRESAMPLE_AUDIOCONVERT_H /** * @file @@ -75,4 +75,4 @@ void swri_audio_convert_free(AudioConvert **ctx); */ int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len); -#endif /* AUDIOCONVERT_H */ +#endif /* SWRESAMPLE_AUDIOCONVERT_H */ diff --git a/libswresample/swresample_internal.h b/libswresample/swresample_internal.h index 5c46dda..5d61de4 100644 --- a/libswresample/swresample_internal.h +++ b/libswresample/swresample_internal.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef SWR_INTERNAL_H -#define SWR_INTERNAL_H +#ifndef SWRESAMPLE_SWRESAMPLE_INTERNAL_H +#define SWRESAMPLE_SWRESAMPLE_INTERNAL_H #include "swresample.h" #include "libavutil/channel_layout.h" diff --git a/libswresample/version.h b/libswresample/version.h index 8f6ecb4..830f00e 100644 --- a/libswresample/version.h +++ b/libswresample/version.h @@ -18,8 +18,8 @@ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#ifndef SWR_VERSION_H -#define SWR_VERSION_H +#ifndef SWRESAMPLE_VERSION_H +#define SWRESAMPLE_VERSION_H /** * @file @@ -42,4 +42,4 @@ #define LIBSWRESAMPLE_IDENT "SwR" AV_STRINGIFY(LIBSWRESAMPLE_VERSION) -#endif /* SWR_VERSION_H */ +#endif /* SWRESAMPLE_VERSION_H */ diff --git a/tests/checkasm/checkasm.h b/tests/checkasm/checkasm.h index 98c0216..5c07f89 100644 --- a/tests/checkasm/checkasm.h +++ b/tests/checkasm/checkasm.h @@ -20,8 +20,8 @@ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. */ -#ifndef CHECKASM_H -#define CHECKASM_H +#ifndef TESTS_CHECKASM_CHECKASM_H +#define TESTS_CHECKASM_CHECKASM_H #include <stdint.h> #include "config.h"

1 0

wtv: Speed up wtv index creation
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | popcornmix <popcornmix(a)gmail.com> | Fri Jan 29 20:27:00 2016 +0000| [def56677e58de9aaa516a738b6469747662ac372] | committer: Michael Niedermayer wtv: Speed up wtv index creation The index creation is O(N^2) with number of entries (typically thousands). On a Pi this can take more than 60 seconds to execute for a recording of a few hours. By replacing with an O(N) loop, this takes virtually zero time Liked-by: Paul B Mahol <onemda(a)gmail.com> Reviewed-by: Peter Ross <pross(a)xvid.org> Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=def56677e58de9aaa… --- libavformat/wtvdec.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/libavformat/wtvdec.c b/libavformat/wtvdec.c index e8f6196..1deb4d9 100644 --- a/libavformat/wtvdec.c +++ b/libavformat/wtvdec.c @@ -1028,21 +1028,23 @@ static int read_header(AVFormatContext *s) pb = wtvfile_open(s, root, root_size, ff_timeline_table_0_entries_Events_le16); if (pb) { int i; + AVIndexEntry *e = wtv->index_entries; + AVIndexEntry *e_end = wtv->index_entries + wtv->nb_index_entries - 1; + uint64_t last_position = 0; while (1) { uint64_t frame_nb = avio_rl64(pb); uint64_t position = avio_rl64(pb); + while (frame_nb > e->size && e <= e_end) { + e->pos = last_position; + e++; + } if (avio_feof(pb)) break; - for (i = wtv->nb_index_entries - 1; i >= 0; i--) { - AVIndexEntry *e = wtv->index_entries + i; - if (frame_nb > e->size) - break; - if (position > e->pos) - e->pos = position; - } + last_position = position; } + e_end->pos = last_position; wtvfile_close(pb); - st->duration = wtv->index_entries[wtv->nb_index_entries - 1].timestamp; + st->duration = e_end->timestamp; } } }

1 0

vf_phase: Reduce the scope of several variables
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Timothy Gu <timothygu99(a)gmail.com> | Sat Jan 30 10:00:19 2016 -0800| [014e4e4499fd5615340d0001f2a0f02382d53718] | committer: Timothy Gu vf_phase: Reduce the scope of several variables Reviewed-by: Paul B Mahol <onemda(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=014e4e4499fd56153… --- libavfilter/vf_phase.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/libavfilter/vf_phase.c b/libavfilter/vf_phase.c index 7249388..fadeb62 100644 --- a/libavfilter/vf_phase.c +++ b/libavfilter/vf_phase.c @@ -116,14 +116,7 @@ static int config_input(AVFilterLink *inlink) */ static enum PhaseMode analyze_plane(void *ctx, enum PhaseMode mode, AVFrame *old, AVFrame *new) { - double bdiff, tdiff, pdiff, scale; - const int ns = new->linesize[0]; - const int os = old->linesize[0]; - const uint8_t *nptr = new->data[0]; - const uint8_t *optr = old->data[0]; - const int h = new->height; - const int w = new->width; - int bdif, tdif, pdif; + double bdiff, tdiff, pdiff; if (mode == AUTO) { mode = new->interlaced_frame ? new->top_field_first ? @@ -136,6 +129,15 @@ static enum PhaseMode analyze_plane(void *ctx, enum PhaseMode mode, AVFrame *old if (mode <= BOTTOM_FIRST) { bdiff = pdiff = tdiff = 65536.0; } else { + const int ns = new->linesize[0]; + const int os = old->linesize[0]; + const uint8_t *nptr = new->data[0]; + const uint8_t *optr = old->data[0]; + const int h = new->height; + const int w = new->width; + int bdif, tdif, pdif; + double scale; + int top = 0, t; const uint8_t *rend, *end = nptr + (h - 2) * ns;

1 0

avfilter/avf_showfreqs: properly handle pts
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Paul B Mahol <onemda(a)gmail.com> | Sun Jan 31 21:38:49 2016 +0100| [4ab4793c155e66ce7845831a5761c2bc0d04b035] | committer: Paul B Mahol avfilter/avf_showfreqs: properly handle pts Signed-off-by: Paul B Mahol <onemda(a)gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=4ab4793c155e66ce7… --- libavfilter/avf_showfreqs.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/libavfilter/avf_showfreqs.c b/libavfilter/avf_showfreqs.c index 6323887..af6e4c2 100644 --- a/libavfilter/avf_showfreqs.c +++ b/libavfilter/avf_showfreqs.c @@ -154,6 +154,15 @@ static int query_formats(AVFilterContext *ctx) return 0; } +static av_cold int init(AVFilterContext *ctx) +{ + ShowFreqsContext *s = ctx->priv; + + s->pts = AV_NOPTS_VALUE; + + return 0; +} + static int config_output(AVFilterLink *outlink) { AVFilterContext *ctx = outlink->src; @@ -423,8 +432,12 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFilterContext *ctx = inlink->dst; ShowFreqsContext *s = ctx->priv; AVFrame *fin = NULL; + int consumed = 0; int ret = 0; + if (s->pts == AV_NOPTS_VALUE) + s->pts = in->pts - av_audio_fifo_size(s->fifo); + av_audio_fifo_write(s->fifo, (void **)in->extended_data, in->nb_samples); while (av_audio_fifo_size(s->fifo) >= s->win_size) { fin = ff_get_audio_buffer(inlink, s->win_size); @@ -433,8 +446,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) goto fail; } - fin->pts = s->pts; - s->pts += s->hop_size; + fin->pts = s->pts + consumed; + consumed += s->hop_size; ret = av_audio_fifo_peek(s->fifo, (void **)fin->extended_data, s->win_size); if (ret < 0) goto fail; @@ -447,6 +460,7 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) } fail: + s->pts = AV_NOPTS_VALUE; av_frame_free(&fin); av_frame_free(&in); return ret; @@ -491,6 +505,7 @@ static const AVFilterPad showfreqs_outputs[] = { AVFilter ff_avf_showfreqs = { .name = "showfreqs", .description = NULL_IF_CONFIG_SMALL("Convert input audio to a frequencies video output."), + .init = init, .uninit = uninit, .query_formats = query_formats, .priv_size = sizeof(ShowFreqsContext),

1 0

msvc: Fix libx264 linking
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Henrik Gramner <henrik(a)gramner.com> | Wed Jan 27 22:41:43 2016 +0100| [58313f2dda0a01add2d3d3d33f0b9eadfee1ff05] | committer: Henrik Gramner msvc: Fix libx264 linking > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=58313f2dda0a01add… --- configure | 1 + 1 file changed, 1 insertion(+) diff --git a/configure b/configure index d7029c3..44ecfc8 100755 --- a/configure +++ b/configure @@ -3547,6 +3547,7 @@ msvc_common_flags(){ -lz) echo zlib.lib ;; -lavifil32) echo vfw32.lib ;; -lavicap32) echo vfw32.lib user32.lib ;; + -lx264) echo libx264.lib ;; -l*) echo ${flag#-l}.lib ;; -LARGEADDRESSAWARE) echo $flag ;; -L*) echo -libpath:${flag#-L} ;;

1 0

avcodec/flacenc: Fix prediction_order parameter
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | Michael Niedermayer <michael(a)niedermayer.cc> | Sun Jan 31 20:04:17 2016 +0100| [3c8e95ab5da5a1e5ada0379020911d99efe48534] | committer: Michael Niedermayer avcodec/flacenc: Fix prediction_order parameter Fixes regression Signed-off-by: Michael Niedermayer <michael(a)niedermayer.cc> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=3c8e95ab5da5a1e5a… --- libavcodec/flacenc.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/libavcodec/flacenc.c b/libavcodec/flacenc.c index 6fed138..a91ed19 100644 --- a/libavcodec/flacenc.c +++ b/libavcodec/flacenc.c @@ -315,8 +315,10 @@ static av_cold int flac_encode_init(AVCodecContext *avctx) FF_LPC_TYPE_LEVINSON, FF_LPC_TYPE_LEVINSON, FF_LPC_TYPE_LEVINSON, FF_LPC_TYPE_LEVINSON})[level]; - s->options.min_prediction_order = ((int[]){ 2, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1})[level]; - s->options.max_prediction_order = ((int[]){ 3, 4, 4, 6, 8, 8, 8, 8, 12, 12, 12, 32, 32})[level]; + if (s->options.min_prediction_order < 0) + s->options.min_prediction_order = ((int[]){ 2, 0, 0, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1})[level]; + if (s->options.max_prediction_order < 0) + s->options.max_prediction_order = ((int[]){ 3, 4, 4, 6, 8, 8, 8, 8, 12, 12, 12, 32, 32})[level]; if (s->options.prediction_order_method < 0) s->options.prediction_order_method = ((int[]){ ORDER_METHOD_EST, ORDER_METHOD_EST, ORDER_METHOD_EST,

1 0

avcodec/dca: add new decoder based on libdcadec
by git＠videolan.org 31 Jan '16

31 Jan '16

ffmpeg | branch: master | foo86 <foobaz86(a)gmail.com> | Sat Jan 16 11:54:38 2016 +0300| [ae5b2c52501d5009fe712334428138a9b758849b] | committer: Hendrik Leppkes avcodec/dca: add new decoder based on libdcadec > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ae5b2c52501d5009f… --- Changelog | 1 + configure | 1 + libavcodec/Makefile | 3 + libavcodec/aarch64/Makefile | 4 +- libavcodec/allcodecs.c | 2 +- libavcodec/arm/Makefile | 6 +- libavcodec/dca_core.c | 2603 +++++++++++++++++++++++++++++++++++++++++++ libavcodec/dca_core.h | 206 ++++ libavcodec/dca_exss.c | 514 +++++++++ libavcodec/dca_exss.h | 92 ++ libavcodec/dca_xll.c | 1499 +++++++++++++++++++++++++ libavcodec/dca_xll.h | 149 +++ libavcodec/dcadec.c | 417 +++++++ libavcodec/dcadec.h | 80 ++ libavcodec/dcadsp.c | 413 +++++++ libavcodec/dcadsp.h | 91 ++ libavcodec/version.h | 2 +- libavcodec/x86/Makefile | 4 +- tests/checkasm/Makefile | 2 +- tests/checkasm/checkasm.c | 4 +- tests/fate/acodec.mak | 4 +- tests/fate/audio.mak | 4 +- 22 files changed, 6085 insertions(+), 16 deletions(-) diff --git a/Changelog b/Changelog index 2d7ad06..5b06ff1 100644 --- a/Changelog +++ b/Changelog @@ -61,6 +61,7 @@ version <next>: - support for dvaudio in wav and avi - libaacplus and libvo-aacenc support removed - Cineform HD decoder +- new DCA decoder with full support for DTS-HD extensions version 2.8: diff --git a/configure b/configure index 66e1139..d7029c3 100755 --- a/configure +++ b/configure @@ -2271,6 +2271,7 @@ comfortnoise_encoder_select="lpc" cook_decoder_select="audiodsp mdct sinewin" cscd_decoder_select="lzo" cscd_decoder_suggest="zlib" +dca_decoder_select="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 1ad2e93..a89fb11 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -222,6 +222,9 @@ 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-$(CONFIG_DCA_DECODER) += dcadec.o dca.o dcadata.o \ + dca_core.o dca_exss.o dca_xll.o \ + dcadsp.o dcadct.o synth_filter.o OBJS-$(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 803f55b..fd89035 100644 --- a/libavcodec/aarch64/Makefile +++ b/libavcodec/aarch64/Makefile @@ -1,4 +1,4 @@ -#OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_init.o +OBJS-$(CONFIG_DCA_DECODER) += aarch64/synth_filter_init.o OBJS-$(CONFIG_FFT) += aarch64/fft_init_aarch64.o OBJS-$(CONFIG_FMTCONVERT) += aarch64/fmtconvert_init.o OBJS-$(CONFIG_H264CHROMA) += aarch64/h264chroma_init_aarch64.o @@ -17,7 +17,7 @@ OBJS-$(CONFIG_VORBIS_DECODER) += aarch64/vorbisdsp_init.o ARMV8-OBJS-$(CONFIG_VIDEODSP) += aarch64/videodsp.o -#NEON-OBJS-$(CONFIG_DCA_DECODER) += 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/allcodecs.c b/libavcodec/allcodecs.c index b174729..c7c1af5 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_ENCODER(DCA, dca); + REGISTER_ENCDEC (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 b2f5a5a..179c403 100644 --- a/libavcodec/arm/Makefile +++ b/libavcodec/arm/Makefile @@ -36,7 +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-$(CONFIG_DCA_DECODER) += arm/synth_filter_init_arm.o +OBJS-$(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-$(CONFIG_RV40_DECODER) += arm/rv40dsp_init_arm.o @@ -87,7 +87,7 @@ VFP-OBJS-$(CONFIG_FMTCONVERT) += arm/fmtconvert_vfp.o VFP-OBJS-$(CONFIG_MDCT) += arm/mdct_vfp.o # decoders/encoders -#VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_vfp.o +VFP-OBJS-$(CONFIG_DCA_DECODER) += arm/synth_filter_vfp.o # NEON optimizations @@ -126,7 +126,7 @@ NEON-OBJS-$(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-$(CONFIG_DCA_DECODER) += arm/synth_filter_neon.o +NEON-OBJS-$(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/dca_core.c b/libavcodec/dca_core.c new file mode 100644 index 0000000..94f0f3d --- /dev/null +++ b/libavcodec/dca_core.c @@ -0,0 +1,2603 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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 "dcadec.h" +#include "dcadata.h" +#include "dcahuff.h" +#include "dcamath.h" +#include "dca_syncwords.h" + +#if ARCH_ARM +#include "arm/dca.h" +#endif + +enum HeaderType { + HEADER_CORE, + HEADER_XCH, + HEADER_XXCH +}; + +enum AudioMode { + AMODE_MONO, // Mode 0: A (mono) + AMODE_MONO_DUAL, // Mode 1: A + B (dual mono) + AMODE_STEREO, // Mode 2: L + R (stereo) + AMODE_STEREO_SUMDIFF, // Mode 3: (L+R) + (L-R) (sum-diff) + AMODE_STEREO_TOTAL, // Mode 4: LT + RT (left and right total) + AMODE_3F, // Mode 5: C + L + R + AMODE_2F1R, // Mode 6: L + R + S + AMODE_3F1R, // Mode 7: C + L + R + S + AMODE_2F2R, // Mode 8: L + R + SL + SR + AMODE_3F2R, // Mode 9: C + L + R + SL + SR + + AMODE_COUNT +}; + +enum ExtAudioType { + EXT_AUDIO_XCH = 0, + EXT_AUDIO_X96 = 2, + EXT_AUDIO_XXCH = 6 +}; + +enum LFEFlag { + LFE_FLAG_NONE, + LFE_FLAG_128, + LFE_FLAG_64, + LFE_FLAG_INVALID +}; + +static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = { + { DCA_SPEAKER_C, -1, -1, -1, -1 }, + { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 }, + { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 }, + { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 }, + { DCA_SPEAKER_L, DCA_SPEAKER_R, -1, -1, -1 }, + { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , -1, -1 }, + { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Cs, -1, -1 }, + { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , DCA_SPEAKER_Cs, -1 }, + { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs, -1 }, + { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs } +}; + +static const uint8_t audio_mode_ch_mask[AMODE_COUNT] = { + DCA_SPEAKER_LAYOUT_MONO, + DCA_SPEAKER_LAYOUT_STEREO, + DCA_SPEAKER_LAYOUT_STEREO, + DCA_SPEAKER_LAYOUT_STEREO, + DCA_SPEAKER_LAYOUT_STEREO, + DCA_SPEAKER_LAYOUT_3_0, + DCA_SPEAKER_LAYOUT_2_1, + DCA_SPEAKER_LAYOUT_3_1, + DCA_SPEAKER_LAYOUT_2_2, + DCA_SPEAKER_LAYOUT_5POINT0 +}; + +static const uint8_t block_code_nbits[7] = { + 7, 10, 12, 13, 15, 17, 19 +}; + +static const uint8_t quant_index_sel_nbits[DCA_CODE_BOOKS] = { + 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 +}; + +static const uint8_t quant_index_group_size[DCA_CODE_BOOKS] = { + 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 +}; + +typedef struct DCAVLC { + int offset; ///< Code values offset + int max_depth; ///< Parameter for get_vlc2() + VLC vlc[7]; ///< Actual codes +} DCAVLC; + +static DCAVLC vlc_bit_allocation; +static DCAVLC vlc_transition_mode; +static DCAVLC vlc_scale_factor; +static DCAVLC vlc_quant_index[DCA_CODE_BOOKS]; + +static av_cold void dca_init_vlcs(void) +{ + static VLC_TYPE dca_table[23622][2]; + static int vlcs_initialized = 0; + int i, j, k; + + if (vlcs_initialized) + return; + +#define DCA_INIT_VLC(vlc, a, b, c, d) \ + do { \ + vlc.table = &dca_table[ff_dca_vlc_offs[k]]; \ + vlc.table_allocated = ff_dca_vlc_offs[k + 1] - ff_dca_vlc_offs[k]; \ + init_vlc(&vlc, a, b, c, 1, 1, d, 2, 2, INIT_VLC_USE_NEW_STATIC); \ + } while (0) + + vlc_bit_allocation.offset = 1; + vlc_bit_allocation.max_depth = 2; + for (i = 0, k = 0; i < 5; i++, k++) + DCA_INIT_VLC(vlc_bit_allocation.vlc[i], bitalloc_12_vlc_bits[i], 12, + bitalloc_12_bits[i], bitalloc_12_codes[i]); + + vlc_scale_factor.offset = -64; + vlc_scale_factor.max_depth = 2; + for (i = 0; i < 5; i++, k++) + DCA_INIT_VLC(vlc_scale_factor.vlc[i], SCALES_VLC_BITS, 129, + scales_bits[i], scales_codes[i]); + + vlc_transition_mode.offset = 0; + vlc_transition_mode.max_depth = 1; + for (i = 0; i < 4; i++, k++) + DCA_INIT_VLC(vlc_transition_mode.vlc[i], tmode_vlc_bits[i], 4, + tmode_bits[i], tmode_codes[i]); + + for (i = 0; i < DCA_CODE_BOOKS; i++) { + vlc_quant_index[i].offset = bitalloc_offsets[i]; + vlc_quant_index[i].max_depth = 1 + (i > 4); + for (j = 0; j < quant_index_group_size[i]; j++, k++) + DCA_INIT_VLC(vlc_quant_index[i].vlc[j], bitalloc_maxbits[i][j], + bitalloc_sizes[i], bitalloc_bits[i][j], bitalloc_codes[i][j]); + } + + vlcs_initialized = 1; +} + +static int get_vlc(GetBitContext *s, DCAVLC *v, int i) +{ + return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset; +} + +static void get_array(GetBitContext *s, int32_t *array, int size, int n) +{ + int i; + + for (i = 0; i < size; i++) + array[i] = get_sbits(s, n); +} + +// 5.3.1 - Bit stream header +static int parse_frame_header(DCACoreDecoder *s) +{ + int normal_frame, pcmr_index; + + // Frame type + normal_frame = get_bits1(&s->gb); + + // Deficit sample count + if (get_bits(&s->gb, 5) != DCA_PCMBLOCK_SAMPLES - 1) { + av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n"); + return normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME; + } + + // CRC present flag + s->crc_present = get_bits1(&s->gb); + + // Number of PCM sample blocks + s->npcmblocks = get_bits(&s->gb, 7) + 1; + if (s->npcmblocks & (DCA_SUBBAND_SAMPLES - 1)) { + av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", s->npcmblocks); + return (s->npcmblocks < 6 || normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME; + } + + // Primary frame byte size + s->frame_size = get_bits(&s->gb, 14) + 1; + if (s->frame_size < 96) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", s->frame_size); + return AVERROR_INVALIDDATA; + } + + // Audio channel arrangement + s->audio_mode = get_bits(&s->gb, 6); + if (s->audio_mode >= AMODE_COUNT) { + av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", s->audio_mode); + return AVERROR_PATCHWELCOME; + } + + // Core audio sampling frequency + s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)]; + if (!s->sample_rate) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n"); + return AVERROR_INVALIDDATA; + } + + // Transmission bit rate + s->bit_rate = ff_dca_bit_rates[get_bits(&s->gb, 5)]; + + // Reserved field + skip_bits1(&s->gb); + + // Embedded dynamic range flag + s->drc_present = get_bits1(&s->gb); + + // Embedded time stamp flag + s->ts_present = get_bits1(&s->gb); + + // Auxiliary data flag + s->aux_present = get_bits1(&s->gb); + + // HDCD mastering flag + skip_bits1(&s->gb); + + // Extension audio descriptor flag + s->ext_audio_type = get_bits(&s->gb, 3); + + // Extended coding flag + s->ext_audio_present = get_bits1(&s->gb); + + // Audio sync word insertion flag + s->sync_ssf = get_bits1(&s->gb); + + // Low frequency effects flag + s->lfe_present = get_bits(&s->gb, 2); + if (s->lfe_present == LFE_FLAG_INVALID) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n"); + return AVERROR_INVALIDDATA; + } + + // Predictor history flag switch + s->predictor_history = get_bits1(&s->gb); + + // Header CRC check bytes + if (s->crc_present) + skip_bits(&s->gb, 16); + + // Multirate interpolator switch + s->filter_perfect = get_bits1(&s->gb); + + // Encoder software revision + skip_bits(&s->gb, 4); + + // Copy history + skip_bits(&s->gb, 2); + + // Source PCM resolution + s->source_pcm_res = ff_dca_bits_per_sample[pcmr_index = get_bits(&s->gb, 3)]; + if (!s->source_pcm_res) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n"); + return AVERROR_INVALIDDATA; + } + s->es_format = pcmr_index & 1; + + // Front sum/difference flag + s->sumdiff_front = get_bits1(&s->gb); + + // Surround sum/difference flag + s->sumdiff_surround = get_bits1(&s->gb); + + // Dialog normalization / unspecified + skip_bits(&s->gb, 4); + + return 0; +} + +// 5.3.2 - Primary audio coding header +static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xch_base) +{ + int n, ch, nchannels, header_size = 0, header_pos = get_bits_count(&s->gb); + unsigned int mask, index; + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + switch (header) { + case HEADER_CORE: + // Number of subframes + s->nsubframes = get_bits(&s->gb, 4) + 1; + + // Number of primary audio channels + s->nchannels = get_bits(&s->gb, 3) + 1; + if (s->nchannels != ff_dca_channels[s->audio_mode]) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid number of primary audio channels (%d) for audio channel arrangement (%d)\n", s->nchannels, s->audio_mode); + return AVERROR_INVALIDDATA; + } + av_assert1(s->nchannels <= DCA_CHANNELS - 2); + + s->ch_mask = audio_mode_ch_mask[s->audio_mode]; + + // Add LFE channel if present + if (s->lfe_present) + s->ch_mask |= DCA_SPEAKER_MASK_LFE1; + break; + + case HEADER_XCH: + s->nchannels = ff_dca_channels[s->audio_mode] + 1; + av_assert1(s->nchannels <= DCA_CHANNELS - 1); + s->ch_mask |= DCA_SPEAKER_MASK_Cs; + break; + + case HEADER_XXCH: + // Channel set header length + header_size = get_bits(&s->gb, 7) + 1; + + // Check CRC + if (s->xxch_crc_present + && (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH channel set header checksum\n"); + return AVERROR_INVALIDDATA; + } + + // Number of channels in a channel set + nchannels = get_bits(&s->gb, 3) + 1; + if (nchannels > DCA_XXCH_CHANNELS_MAX) { + avpriv_request_sample(s->avctx, "%d XXCH channels", nchannels); + return AVERROR_PATCHWELCOME; + } + s->nchannels = ff_dca_channels[s->audio_mode] + nchannels; + av_assert1(s->nchannels <= DCA_CHANNELS); + + // Loudspeaker layout mask + mask = get_bits_long(&s->gb, s->xxch_mask_nbits - DCA_SPEAKER_Cs); + s->xxch_spkr_mask = mask << DCA_SPEAKER_Cs; + + if (av_popcount(s->xxch_spkr_mask) != nchannels) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH speaker layout mask (%#x)\n", s->xxch_spkr_mask); + return AVERROR_INVALIDDATA; + } + + if (s->xxch_core_mask & s->xxch_spkr_mask) { + av_log(s->avctx, AV_LOG_ERROR, "XXCH speaker layout mask (%#x) overlaps with core (%#x)\n", s->xxch_spkr_mask, s->xxch_core_mask); + return AVERROR_INVALIDDATA; + } + + // Combine core and XXCH masks together + s->ch_mask = s->xxch_core_mask | s->xxch_spkr_mask; + + // Downmix coefficients present in stream + if (get_bits1(&s->gb)) { + int *coeff_ptr = s->xxch_dmix_coeff; + + // Downmix already performed by encoder + s->xxch_dmix_embedded = get_bits1(&s->gb); + + // Downmix scale factor + index = get_bits(&s->gb, 6) * 4 - FF_DCA_DMIXTABLE_OFFSET - 3; + if (index >= FF_DCA_INV_DMIXTABLE_SIZE) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix scale index (%d)\n", index); + return AVERROR_INVALIDDATA; + } + s->xxch_dmix_scale_inv = ff_dca_inv_dmixtable[index]; + + // Downmix channel mapping mask + for (ch = 0; ch < nchannels; ch++) { + mask = get_bits_long(&s->gb, s->xxch_mask_nbits); + if ((mask & s->xxch_core_mask) != mask) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix channel mapping mask (%#x)\n", mask); + return AVERROR_INVALIDDATA; + } + s->xxch_dmix_mask[ch] = mask; + } + + // Downmix coefficients + for (ch = 0; ch < nchannels; ch++) { + for (n = 0; n < s->xxch_mask_nbits; n++) { + if (s->xxch_dmix_mask[ch] & (1U << n)) { + int code = get_bits(&s->gb, 7); + int sign = (code >> 6) - 1; + if (code &= 63) { + index = code * 4 - 3; + if (index >= FF_DCA_DMIXTABLE_SIZE) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix coefficient index (%d)\n", index); + return AVERROR_INVALIDDATA; + } + *coeff_ptr++ = (ff_dca_dmixtable[index] ^ sign) - sign; + } else { + *coeff_ptr++ = 0; + } + } + } + } + } else { + s->xxch_dmix_embedded = 0; + } + + break; + } + + // Subband activity count + for (ch = xch_base; ch < s->nchannels; ch++) { + s->nsubbands[ch] = get_bits(&s->gb, 5) + 2; + if (s->nsubbands[ch] > DCA_SUBBANDS) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid subband activity count\n"); + return AVERROR_INVALIDDATA; + } + } + + // High frequency VQ start subband + for (ch = xch_base; ch < s->nchannels; ch++) + s->subband_vq_start[ch] = get_bits(&s->gb, 5) + 1; + + // Joint intensity coding index + for (ch = xch_base; ch < s->nchannels; ch++) { + if ((n = get_bits(&s->gb, 3)) && header == HEADER_XXCH) + n += xch_base - 1; + if (n > s->nchannels) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid joint intensity coding index\n"); + return AVERROR_INVALIDDATA; + } + s->joint_intensity_index[ch] = n; + } + + // Transient mode code book + for (ch = xch_base; ch < s->nchannels; ch++) + s->transition_mode_sel[ch] = get_bits(&s->gb, 2); + + // Scale factor code book + for (ch = xch_base; ch < s->nchannels; ch++) { + s->scale_factor_sel[ch] = get_bits(&s->gb, 3); + if (s->scale_factor_sel[ch] == 7) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor code book\n"); + return AVERROR_INVALIDDATA; + } + } + + // Bit allocation quantizer select + for (ch = xch_base; ch < s->nchannels; ch++) { + s->bit_allocation_sel[ch] = get_bits(&s->gb, 3); + if (s->bit_allocation_sel[ch] == 7) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation quantizer select\n"); + return AVERROR_INVALIDDATA; + } + } + + // Quantization index codebook select + for (n = 0; n < DCA_CODE_BOOKS; n++) + for (ch = xch_base; ch < s->nchannels; ch++) + s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]); + + // Scale factor adjustment index + for (n = 0; n < DCA_CODE_BOOKS; n++) + for (ch = xch_base; ch < s->nchannels; ch++) + if (s->quant_index_sel[ch][n] < quant_index_group_size[n]) + s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)]; + + if (header == HEADER_XXCH) { + // Reserved + // Byte align + // CRC16 of channel set header + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set header\n"); + return AVERROR_INVALIDDATA; + } + } else { + // Audio header CRC check word + if (s->crc_present) + skip_bits(&s->gb, 16); + } + + return 0; +} + +static inline int parse_scale(DCACoreDecoder *s, int *scale_index, int sel) +{ + const uint32_t *scale_table; + unsigned int scale_size; + + // Select the root square table + if (sel > 5) { + scale_table = ff_dca_scale_factor_quant7; + scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); + } else { + scale_table = ff_dca_scale_factor_quant6; + scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6); + } + + // If Huffman code was used, the difference of scales was encoded + if (sel < 5) + *scale_index += get_vlc(&s->gb, &vlc_scale_factor, sel); + else + *scale_index = get_bits(&s->gb, sel + 1); + + // Look up scale factor from the root square table + if ((unsigned int)*scale_index >= scale_size) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor index\n"); + return AVERROR_INVALIDDATA; + } + + return scale_table[*scale_index]; +} + +static inline int parse_joint_scale(DCACoreDecoder *s, int sel) +{ + int scale_index; + + // Absolute value was encoded even when Huffman code was used + if (sel < 5) + scale_index = get_vlc(&s->gb, &vlc_scale_factor, sel); + else + scale_index = get_bits(&s->gb, sel + 1); + + // Bias by 64 + scale_index += 64; + + // Look up joint scale factor + if ((unsigned int)scale_index >= FF_ARRAY_ELEMS(ff_dca_joint_scale_factors)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor index\n"); + return AVERROR_INVALIDDATA; + } + + return ff_dca_joint_scale_factors[scale_index]; +} + +// 5.4.1 - Primary audio coding side information +static int parse_subframe_header(DCACoreDecoder *s, int sf, + enum HeaderType header, int xch_base) +{ + int ch, band, ret; + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + if (header == HEADER_CORE) { + // Subsubframe count + s->nsubsubframes[sf] = get_bits(&s->gb, 2) + 1; + + // Partial subsubframe sample count + skip_bits(&s->gb, 3); + } + + // Prediction mode + for (ch = xch_base; ch < s->nchannels; ch++) + for (band = 0; band < s->nsubbands[ch]; band++) + s->prediction_mode[ch][band] = get_bits1(&s->gb); + + // Prediction coefficients VQ address + for (ch = xch_base; ch < s->nchannels; ch++) + for (band = 0; band < s->nsubbands[ch]; band++) + if (s->prediction_mode[ch][band]) + s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12); + + // Bit allocation index + for (ch = xch_base; ch < s->nchannels; ch++) { + int sel = s->bit_allocation_sel[ch]; + + for (band = 0; band < s->subband_vq_start[ch]; band++) { + int abits; + + if (sel < 5) + abits = get_vlc(&s->gb, &vlc_bit_allocation, sel); + else + abits = get_bits(&s->gb, sel - 1); + + if (abits > DCA_ABITS_MAX) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n"); + return AVERROR_INVALIDDATA; + } + + s->bit_allocation[ch][band] = abits; + } + } + + // Transition mode + for (ch = xch_base; ch < s->nchannels; ch++) { + // Clear transition mode for all subbands + memset(s->transition_mode[sf][ch], 0, sizeof(s->transition_mode[0][0])); + + // Transient possible only if more than one subsubframe + if (s->nsubsubframes[sf] > 1) { + int sel = s->transition_mode_sel[ch]; + for (band = 0; band < s->subband_vq_start[ch]; band++) + if (s->bit_allocation[ch][band]) + s->transition_mode[sf][ch][band] = get_vlc(&s->gb, &vlc_transition_mode, sel); + } + } + + // Scale factors + for (ch = xch_base; ch < s->nchannels; ch++) { + int sel = s->scale_factor_sel[ch]; + int scale_index = 0; + + // Extract scales for subbands up to VQ + for (band = 0; band < s->subband_vq_start[ch]; band++) { + if (s->bit_allocation[ch][band]) { + if ((ret = parse_scale(s, &scale_index, sel)) < 0) + return ret; + s->scale_factors[ch][band][0] = ret; + if (s->transition_mode[sf][ch][band]) { + if ((ret = parse_scale(s, &scale_index, sel)) < 0) + return ret; + s->scale_factors[ch][band][1] = ret; + } + } else { + s->scale_factors[ch][band][0] = 0; + } + } + + // High frequency VQ subbands + for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++) { + if ((ret = parse_scale(s, &scale_index, sel)) < 0) + return ret; + s->scale_factors[ch][band][0] = ret; + } + } + + // Joint subband codebook select + for (ch = xch_base; ch < s->nchannels; ch++) { + if (s->joint_intensity_index[ch]) { + s->joint_scale_sel[ch] = get_bits(&s->gb, 3); + if (s->joint_scale_sel[ch] == 7) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor code book\n"); + return AVERROR_INVALIDDATA; + } + } + } + + // Scale factors for joint subband coding + for (ch = xch_base; ch < s->nchannels; ch++) { + int src_ch = s->joint_intensity_index[ch] - 1; + if (src_ch >= 0) { + int sel = s->joint_scale_sel[ch]; + for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) { + if ((ret = parse_joint_scale(s, sel)) < 0) + return ret; + s->joint_scale_factors[ch][band] = ret; + } + } + } + + // Dynamic range coefficient + if (s->drc_present && header == HEADER_CORE) + skip_bits(&s->gb, 8); + + // Side information CRC check word + if (s->crc_present) + skip_bits(&s->gb, 16); + + return 0; +} + +#ifndef decode_blockcodes +static inline int decode_blockcodes(int code1, int code2, int levels, int32_t *audio) +{ + int offset = (levels - 1) / 2; + int n, div; + + for (n = 0; n < DCA_SUBBAND_SAMPLES / 2; n++) { + div = FASTDIV(code1, levels); + audio[n] = code1 - div * levels - offset; + code1 = div; + } + for (; n < DCA_SUBBAND_SAMPLES; n++) { + div = FASTDIV(code2, levels); + audio[n] = code2 - div * levels - offset; + code2 = div; + } + + return code1 | code2; +} +#endif + +static inline int parse_block_codes(DCACoreDecoder *s, int32_t *audio, int abits) +{ + // Extract block code indices from the bit stream + int code1 = get_bits(&s->gb, block_code_nbits[abits - 1]); + int code2 = get_bits(&s->gb, block_code_nbits[abits - 1]); + int levels = ff_dca_quant_levels[abits]; + + // Look up samples from the block code book + if (decode_blockcodes(code1, code2, levels, audio)) { + av_log(s->avctx, AV_LOG_ERROR, "Failed to decode block code(s)\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static inline int parse_huffman_codes(DCACoreDecoder *s, int32_t *audio, int abits, int sel) +{ + int i; + + // Extract Huffman codes from the bit stream + for (i = 0; i < DCA_SUBBAND_SAMPLES; i++) + audio[i] = get_vlc(&s->gb, &vlc_quant_index[abits - 1], sel); + + return 1; +} + +static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, int ch) +{ + av_assert1(abits >= 0 && abits <= DCA_ABITS_MAX); + + if (abits == 0) { + // No bits allocated + memset(audio, 0, DCA_SUBBAND_SAMPLES * sizeof(*audio)); + return 0; + } + + if (abits <= DCA_CODE_BOOKS) { + int sel = s->quant_index_sel[ch][abits - 1]; + if (sel < quant_index_group_size[abits - 1]) { + // Huffman codes + return parse_huffman_codes(s, audio, abits, sel); + } + if (abits <= 7) { + // Block codes + return parse_block_codes(s, audio, abits); + } + } + + // No further encoding + get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3); + return 0; +} + +static inline void dequantize(int32_t *output, const int32_t *input, + int32_t step_size, int32_t scale, int residual) +{ + // Account for quantizer step size + int64_t step_scale = (int64_t)step_size * scale; + int n, shift = 0; + + // Limit scale factor resolution to 22 bits + if (step_scale > (1 << 23)) { + shift = av_log2(step_scale >> 23) + 1; + step_scale >>= shift; + } + + // Scale the samples + if (residual) { + for (n = 0; n < DCA_SUBBAND_SAMPLES; n++) + output[n] += clip23(norm__(input[n] * step_scale, 22 - shift)); + } else { + for (n = 0; n < DCA_SUBBAND_SAMPLES; n++) + output[n] = clip23(norm__(input[n] * step_scale, 22 - shift)); + } +} + +static inline void inverse_adpcm(int32_t **subband_samples, + const int16_t *vq_index, + const int8_t *prediction_mode, + int sb_start, int sb_end, + int ofs, int len) +{ + int i, j, k; + + for (i = sb_start; i < sb_end; i++) { + if (prediction_mode[i]) { + const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]]; + int32_t *ptr = subband_samples[i] + ofs; + for (j = 0; j < len; j++) { + int64_t err = 0; + for (k = 0; k < DCA_ADPCM_COEFFS; k++) + err += (int64_t)ptr[j - k - 1] * coeff[k]; + ptr[j] = clip23(ptr[j] + clip23(norm13(err))); + } + } + } +} + +// 5.5 - Primary audio data arrays +static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType header, + int xch_base, int *sub_pos, int *lfe_pos) +{ + int32_t audio[16], scale; + int n, ssf, ofs, ch, band; + + // Check number of subband samples in this subframe + int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES; + if (*sub_pos + nsamples > s->npcmblocks) { + av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n"); + return AVERROR_INVALIDDATA; + } + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + // VQ encoded subbands + for (ch = xch_base; ch < s->nchannels; ch++) { + int32_t vq_index[DCA_SUBBANDS]; + + for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++) + // Extract the VQ address from the bit stream + vq_index[band] = get_bits(&s->gb, 10); + + if (s->subband_vq_start[ch] < s->nsubbands[ch]) { + s->dcadsp->decode_hf(s->subband_samples[ch], vq_index, + ff_dca_high_freq_vq, s->scale_factors[ch], + s->subband_vq_start[ch], s->nsubbands[ch], + *sub_pos, nsamples); + } + } + + // Low frequency effect data + if (s->lfe_present && header == HEADER_CORE) { + unsigned int index; + + // Determine number of LFE samples in this subframe + int nlfesamples = 2 * s->lfe_present * s->nsubsubframes[sf]; + av_assert1((unsigned int)nlfesamples <= FF_ARRAY_ELEMS(audio)); + + // Extract LFE samples from the bit stream + get_array(&s->gb, audio, nlfesamples, 8); + + // Extract scale factor index from the bit stream + index = get_bits(&s->gb, 8); + if (index >= FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE scale factor index\n"); + return AVERROR_INVALIDDATA; + } + + // Look up the 7-bit root square quantization table + scale = ff_dca_scale_factor_quant7[index]; + + // Account for quantizer step size which is 0.035 + scale = mul23(4697620 /* 0.035 * (1 << 27) */, scale); + + // Scale and take the LFE samples + for (n = 0, ofs = *lfe_pos; n < nlfesamples; n++, ofs++) + s->lfe_samples[ofs] = clip23(audio[n] * scale >> 4); + + // Advance LFE sample pointer for the next subframe + *lfe_pos = ofs; + } + + // Audio data + for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) { + for (ch = xch_base; ch < s->nchannels; ch++) { + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + // Not high frequency VQ subbands + for (band = 0; band < s->subband_vq_start[ch]; band++) { + int ret, trans_ssf, abits = s->bit_allocation[ch][band]; + int32_t step_size; + + // Extract bits from the bit stream + if ((ret = extract_audio(s, audio, abits, ch)) < 0) + return ret; + + // Select quantization step size table and look up + // quantization step size + if (s->bit_rate == 3) + step_size = ff_dca_lossless_quant[abits]; + else + step_size = ff_dca_lossy_quant[abits]; + + // Identify transient location + trans_ssf = s->transition_mode[sf][ch][band]; + + // Determine proper scale factor + if (trans_ssf == 0 || ssf < trans_ssf) + scale = s->scale_factors[ch][band][0]; + else + scale = s->scale_factors[ch][band][1]; + + // Adjust scale factor when SEL indicates Huffman code + if (ret > 0) { + int64_t adj = s->scale_factor_adj[ch][abits - 1]; + scale = clip23(adj * scale >> 22); + } + + dequantize(s->subband_samples[ch][band] + ofs, + audio, step_size, scale, 0); + } + } + + // DSYNC + if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) { + av_log(s->avctx, AV_LOG_ERROR, "DSYNC check failed\n"); + return AVERROR_INVALIDDATA; + } + + ofs += DCA_SUBBAND_SAMPLES; + } + + // Inverse ADPCM + for (ch = xch_base; ch < s->nchannels; ch++) { + inverse_adpcm(s->subband_samples[ch], s->prediction_vq_index[ch], + s->prediction_mode[ch], 0, s->nsubbands[ch], + *sub_pos, nsamples); + } + + // Joint subband coding + for (ch = xch_base; ch < s->nchannels; ch++) { + int src_ch = s->joint_intensity_index[ch] - 1; + if (src_ch >= 0) { + s->dcadsp->decode_joint(s->subband_samples[ch], s->subband_samples[src_ch], + s->joint_scale_factors[ch], s->nsubbands[ch], + s->nsubbands[src_ch], *sub_pos, nsamples); + } + } + + // Advance subband sample pointer for the next subframe + *sub_pos = ofs; + return 0; +} + +static void erase_adpcm_history(DCACoreDecoder *s) +{ + int ch, band; + + // Erase ADPCM history from previous frame if + // predictor history switch was disabled + for (ch = 0; ch < DCA_CHANNELS; ch++) + for (band = 0; band < DCA_SUBBANDS; band++) + AV_ZERO128(s->subband_samples[ch][band] - DCA_ADPCM_COEFFS); +} + +static int alloc_sample_buffer(DCACoreDecoder *s) +{ + int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks; + int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS; + int nlfesamples = DCA_LFE_HISTORY + s->npcmblocks / 2; + unsigned int size = s->subband_size; + int ch, band; + + // Reallocate subband sample buffer + av_fast_mallocz(&s->subband_buffer, &s->subband_size, + (nframesamples + nlfesamples) * sizeof(int32_t)); + if (!s->subband_buffer) + return AVERROR(ENOMEM); + + if (size != s->subband_size) { + for (ch = 0; ch < DCA_CHANNELS; ch++) + for (band = 0; band < DCA_SUBBANDS; band++) + s->subband_samples[ch][band] = s->subband_buffer + + (ch * DCA_SUBBANDS + band) * nchsamples + DCA_ADPCM_COEFFS; + s->lfe_samples = s->subband_buffer + nframesamples; + } + + if (!s->predictor_history) + erase_adpcm_history(s); + + return 0; +} + +static int parse_frame_data(DCACoreDecoder *s, enum HeaderType header, int xch_base) +{ + int sf, ch, ret, band, sub_pos, lfe_pos; + + if ((ret = parse_coding_header(s, header, xch_base)) < 0) + return ret; + + for (sf = 0, sub_pos = 0, lfe_pos = DCA_LFE_HISTORY; sf < s->nsubframes; sf++) { + if ((ret = parse_subframe_header(s, sf, header, xch_base)) < 0) + return ret; + if ((ret = parse_subframe_audio(s, sf, header, xch_base, &sub_pos, &lfe_pos)) < 0) + return ret; + } + + for (ch = xch_base; ch < s->nchannels; ch++) { + // Determine number of active subbands for this channel + int nsubbands = s->nsubbands[ch]; + if (s->joint_intensity_index[ch]) + nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]); + + // Update history for ADPCM + for (band = 0; band < nsubbands; band++) { + int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS; + AV_COPY128(samples, samples + s->npcmblocks); + } + + // Clear inactive subbands + for (; band < DCA_SUBBANDS; band++) { + int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS; + memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t)); + } + } + + return 0; +} + +static int parse_xch_frame(DCACoreDecoder *s) +{ + int ret; + + if (s->ch_mask & DCA_SPEAKER_MASK_Cs) { + av_log(s->avctx, AV_LOG_ERROR, "XCH with Cs speaker already present\n"); + return AVERROR_INVALIDDATA; + } + + if ((ret = parse_frame_data(s, HEADER_XCH, s->nchannels)) < 0) + return ret; + + // Seek to the end of core frame, don't trust XCH frame size + if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XCH frame\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int parse_xxch_frame(DCACoreDecoder *s) +{ + int xxch_nchsets, xxch_frame_size; + int ret, mask, header_size, header_pos = get_bits_count(&s->gb); + + // XXCH sync word + if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XXCH) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH sync word\n"); + return AVERROR_INVALIDDATA; + } + + // XXCH frame header length + header_size = get_bits(&s->gb, 6) + 1; + + // Check XXCH frame header CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH frame header checksum\n"); + return AVERROR_INVALIDDATA; + } + + // CRC presence flag for channel set header + s->xxch_crc_present = get_bits1(&s->gb); + + // Number of bits for loudspeaker mask + s->xxch_mask_nbits = get_bits(&s->gb, 5) + 1; + if (s->xxch_mask_nbits <= DCA_SPEAKER_Cs) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XXCH speaker mask (%d)\n", s->xxch_mask_nbits); + return AVERROR_INVALIDDATA; + } + + // Number of channel sets + xxch_nchsets = get_bits(&s->gb, 2) + 1; + if (xxch_nchsets > 1) { + avpriv_request_sample(s->avctx, "%d XXCH channel sets", xxch_nchsets); + return AVERROR_PATCHWELCOME; + } + + // Channel set 0 data byte size + xxch_frame_size = get_bits(&s->gb, 14) + 1; + + // Core loudspeaker activity mask + s->xxch_core_mask = get_bits_long(&s->gb, s->xxch_mask_nbits); + + // Validate the core mask + mask = s->ch_mask; + + if ((mask & DCA_SPEAKER_MASK_Ls) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss)) + mask = (mask & ~DCA_SPEAKER_MASK_Ls) | DCA_SPEAKER_MASK_Lss; + + if ((mask & DCA_SPEAKER_MASK_Rs) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss)) + mask = (mask & ~DCA_SPEAKER_MASK_Rs) | DCA_SPEAKER_MASK_Rss; + + if (mask != s->xxch_core_mask) { + av_log(s->avctx, AV_LOG_ERROR, "XXCH core speaker activity mask (%#x) disagrees with core (%#x)\n", s->xxch_core_mask, mask); + return AVERROR_INVALIDDATA; + } + + // Reserved + // Byte align + // CRC16 of XXCH frame header + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH frame header\n"); + return AVERROR_INVALIDDATA; + } + + // Parse XXCH channel set 0 + if ((ret = parse_frame_data(s, HEADER_XXCH, s->nchannels)) < 0) + return ret; + + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8 + xxch_frame_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchannels, + int *xbr_nsubbands, int xbr_transition_mode, int sf, int *sub_pos) +{ + int xbr_nabits[DCA_CHANNELS]; + int xbr_bit_allocation[DCA_CHANNELS][DCA_SUBBANDS]; + int xbr_scale_nbits[DCA_CHANNELS]; + int32_t xbr_scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2]; + int ssf, ch, band, ofs; + + // Check number of subband samples in this subframe + if (*sub_pos + s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES > s->npcmblocks) { + av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n"); + return AVERROR_INVALIDDATA; + } + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + // Number of bits for XBR bit allocation index + for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) + xbr_nabits[ch] = get_bits(&s->gb, 2) + 2; + + // XBR bit allocation index + for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) { + for (band = 0; band < xbr_nsubbands[ch]; band++) { + xbr_bit_allocation[ch][band] = get_bits(&s->gb, xbr_nabits[ch]); + if (xbr_bit_allocation[ch][band] > DCA_ABITS_MAX) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR bit allocation index\n"); + return AVERROR_INVALIDDATA; + } + } + } + + // Number of bits for scale indices + for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) { + xbr_scale_nbits[ch] = get_bits(&s->gb, 3); + if (!xbr_scale_nbits[ch]) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XBR scale factor index\n"); + return AVERROR_INVALIDDATA; + } + } + + // XBR scale factors + for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) { + const uint32_t *scale_table; + int scale_size; + + // Select the root square table + if (s->scale_factor_sel[ch] > 5) { + scale_table = ff_dca_scale_factor_quant7; + scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7); + } else { + scale_table = ff_dca_scale_factor_quant6; + scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6); + } + + // Parse scale factor indices and look up scale factors from the root + // square table + for (band = 0; band < xbr_nsubbands[ch]; band++) { + if (xbr_bit_allocation[ch][band]) { + int scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]); + if (scale_index >= scale_size) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n"); + return AVERROR_INVALIDDATA; + } + xbr_scale_factors[ch][band][0] = scale_table[scale_index]; + if (xbr_transition_mode && s->transition_mode[sf][ch][band]) { + scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]); + if (scale_index >= scale_size) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n"); + return AVERROR_INVALIDDATA; + } + xbr_scale_factors[ch][band][1] = scale_table[scale_index]; + } + } + } + } + + // Audio data + for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) { + for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) { + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + for (band = 0; band < xbr_nsubbands[ch]; band++) { + int ret, trans_ssf, abits = xbr_bit_allocation[ch][band]; + int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale; + + // Extract bits from the bit stream + if (abits > 7) { + // No further encoding + get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3); + } else if (abits > 0) { + // Block codes + if ((ret = parse_block_codes(s, audio, abits)) < 0) + return ret; + } else { + // No bits allocated + continue; + } + + // Look up quantization step size + step_size = ff_dca_lossless_quant[abits]; + + // Identify transient location + if (xbr_transition_mode) + trans_ssf = s->transition_mode[sf][ch][band]; + else + trans_ssf = 0; + + // Determine proper scale factor + if (trans_ssf == 0 || ssf < trans_ssf) + scale = xbr_scale_factors[ch][band][0]; + else + scale = xbr_scale_factors[ch][band][1]; + + dequantize(s->subband_samples[ch][band] + ofs, + audio, step_size, scale, 1); + } + } + + // DSYNC + if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) { + av_log(s->avctx, AV_LOG_ERROR, "XBR-DSYNC check failed\n"); + return AVERROR_INVALIDDATA; + } + + ofs += DCA_SUBBAND_SAMPLES; + } + + // Advance subband sample pointer for the next subframe + *sub_pos = ofs; + return 0; +} + +static int parse_xbr_frame(DCACoreDecoder *s) +{ + int xbr_frame_size[DCA_EXSS_CHSETS_MAX]; + int xbr_nchannels[DCA_EXSS_CHSETS_MAX]; + int xbr_nsubbands[DCA_EXSS_CHSETS_MAX * DCA_EXSS_CHANNELS_MAX]; + int xbr_nchsets, xbr_transition_mode, xbr_band_nbits, xbr_base_ch; + int i, ch1, ch2, ret, header_size, header_pos = get_bits_count(&s->gb); + + // XBR sync word + if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XBR) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR sync word\n"); + return AVERROR_INVALIDDATA; + } + + // XBR frame header length + header_size = get_bits(&s->gb, 6) + 1; + + // Check XBR frame header CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR frame header checksum\n"); + return AVERROR_INVALIDDATA; + } + + // Number of channel sets + xbr_nchsets = get_bits(&s->gb, 2) + 1; + + // Channel set data byte size + for (i = 0; i < xbr_nchsets; i++) + xbr_frame_size[i] = get_bits(&s->gb, 14) + 1; + + // Transition mode flag + xbr_transition_mode = get_bits1(&s->gb); + + // Channel set headers + for (i = 0, ch2 = 0; i < xbr_nchsets; i++) { + xbr_nchannels[i] = get_bits(&s->gb, 3) + 1; + xbr_band_nbits = get_bits(&s->gb, 2) + 5; + for (ch1 = 0; ch1 < xbr_nchannels[i]; ch1++, ch2++) { + xbr_nsubbands[ch2] = get_bits(&s->gb, xbr_band_nbits) + 1; + if (xbr_nsubbands[ch2] > DCA_SUBBANDS) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid number of active XBR subbands (%d)\n", xbr_nsubbands[ch2]); + return AVERROR_INVALIDDATA; + } + } + } + + // Reserved + // Byte align + // CRC16 of XBR frame header + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR frame header\n"); + return AVERROR_INVALIDDATA; + } + + // Channel set data + for (i = 0, xbr_base_ch = 0; i < xbr_nchsets; i++) { + header_pos = get_bits_count(&s->gb); + + if (xbr_base_ch + xbr_nchannels[i] <= s->nchannels) { + int sf, sub_pos; + + for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) { + if ((ret = parse_xbr_subframe(s, xbr_base_ch, + xbr_base_ch + xbr_nchannels[i], + xbr_nsubbands, xbr_transition_mode, + sf, &sub_pos)) < 0) + return ret; + } + } + + xbr_base_ch += xbr_nchannels[i]; + + if (ff_dca_seek_bits(&s->gb, header_pos + xbr_frame_size[i] * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR channel set\n"); + return AVERROR_INVALIDDATA; + } + } + + return 0; +} + +// Modified ISO/IEC 9899 linear congruential generator +// Returns pseudorandom integer in range [-2^30, 2^30 - 1] +static int rand_x96(DCACoreDecoder *s) +{ + s->x96_rand = 1103515245U * s->x96_rand + 12345U; + return (s->x96_rand & 0x7fffffff) - 0x40000000; +} + +static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int *sub_pos) +{ + int n, ssf, ch, band, ofs; + + // Check number of subband samples in this subframe + int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES; + if (*sub_pos + nsamples > s->npcmblocks) { + av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n"); + return AVERROR_INVALIDDATA; + } + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + // VQ encoded or unallocated subbands + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) { + // Get the sample pointer and scale factor + int32_t *samples = s->x96_subband_samples[ch][band] + *sub_pos; + int32_t scale = s->scale_factors[ch][band >> 1][band & 1]; + + switch (s->bit_allocation[ch][band]) { + case 0: // No bits allocated for subband + if (scale <= 1) + memset(samples, 0, nsamples * sizeof(int32_t)); + else for (n = 0; n < nsamples; n++) + // Generate scaled random samples + samples[n] = mul31(rand_x96(s), scale); + break; + + case 1: // VQ encoded subband + for (ssf = 0; ssf < (s->nsubsubframes[sf] + 1) / 2; ssf++) { + // Extract the VQ address from the bit stream and look up + // the VQ code book for up to 16 subband samples + const int8_t *vq_samples = ff_dca_high_freq_vq[get_bits(&s->gb, 10)]; + // Scale and take the samples + for (n = 0; n < FFMIN(nsamples - ssf * 16, 16); n++) + *samples++ = clip23(vq_samples[n] * scale + (1 << 3) >> 4); + } + break; + } + } + } + + // Audio data + for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) { + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) { + int ret, abits = s->bit_allocation[ch][band] - 1; + int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale; + + // Not VQ encoded or unallocated subbands + if (abits < 1) + continue; + + // Extract bits from the bit stream + if ((ret = extract_audio(s, audio, abits, ch)) < 0) + return ret; + + // Select quantization step size table and look up quantization + // step size + if (s->bit_rate == 3) + step_size = ff_dca_lossless_quant[abits]; + else + step_size = ff_dca_lossy_quant[abits]; + + // Get the scale factor + scale = s->scale_factors[ch][band >> 1][band & 1]; + + dequantize(s->x96_subband_samples[ch][band] + ofs, + audio, step_size, scale, 0); + } + } + + // DSYNC + if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) { + av_log(s->avctx, AV_LOG_ERROR, "X96-DSYNC check failed\n"); + return AVERROR_INVALIDDATA; + } + + ofs += DCA_SUBBAND_SAMPLES; + } + + // Inverse ADPCM + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + inverse_adpcm(s->x96_subband_samples[ch], s->prediction_vq_index[ch], + s->prediction_mode[ch], s->x96_subband_start, s->nsubbands[ch], + *sub_pos, nsamples); + } + + // Joint subband coding + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + int src_ch = s->joint_intensity_index[ch] - 1; + if (src_ch >= 0) { + s->dcadsp->decode_joint(s->x96_subband_samples[ch], s->x96_subband_samples[src_ch], + s->joint_scale_factors[ch], s->nsubbands[ch], + s->nsubbands[src_ch], *sub_pos, nsamples); + } + } + + // Advance subband sample pointer for the next subframe + *sub_pos = ofs; + return 0; +} + +static void erase_x96_adpcm_history(DCACoreDecoder *s) +{ + int ch, band; + + // Erase ADPCM history from previous frame if + // predictor history switch was disabled + for (ch = 0; ch < DCA_CHANNELS; ch++) + for (band = 0; band < DCA_SUBBANDS_X96; band++) + AV_ZERO128(s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS); +} + +static int alloc_x96_sample_buffer(DCACoreDecoder *s) +{ + int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks; + int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS_X96; + unsigned int size = s->x96_subband_size; + int ch, band; + + // Reallocate subband sample buffer + av_fast_mallocz(&s->x96_subband_buffer, &s->x96_subband_size, + nframesamples * sizeof(int32_t)); + if (!s->x96_subband_buffer) + return AVERROR(ENOMEM); + + if (size != s->x96_subband_size) { + for (ch = 0; ch < DCA_CHANNELS; ch++) + for (band = 0; band < DCA_SUBBANDS_X96; band++) + s->x96_subband_samples[ch][band] = s->x96_subband_buffer + + (ch * DCA_SUBBANDS_X96 + band) * nchsamples + DCA_ADPCM_COEFFS; + } + + if (!s->predictor_history) + erase_x96_adpcm_history(s); + + return 0; +} + +static int parse_x96_subframe_header(DCACoreDecoder *s, int xch_base) +{ + int ch, band, ret; + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + // Prediction mode + for (ch = xch_base; ch < s->x96_nchannels; ch++) + for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) + s->prediction_mode[ch][band] = get_bits1(&s->gb); + + // Prediction coefficients VQ address + for (ch = xch_base; ch < s->x96_nchannels; ch++) + for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) + if (s->prediction_mode[ch][band]) + s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12); + + // Bit allocation index + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + int sel = s->bit_allocation_sel[ch]; + int abits = 0; + + for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) { + // If Huffman code was used, the difference of abits was encoded + if (sel < 7) + abits += get_vlc(&s->gb, &vlc_quant_index[5 + 2 * s->x96_high_res], sel); + else + abits = get_bits(&s->gb, 3 + s->x96_high_res); + + if (abits < 0 || abits > 7 + 8 * s->x96_high_res) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 bit allocation index\n"); + return AVERROR_INVALIDDATA; + } + + s->bit_allocation[ch][band] = abits; + } + } + + // Scale factors + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + int sel = s->scale_factor_sel[ch]; + int scale_index = 0; + + // Extract scales for subbands which are transmitted even for + // unallocated subbands + for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) { + if ((ret = parse_scale(s, &scale_index, sel)) < 0) + return ret; + s->scale_factors[ch][band >> 1][band & 1] = ret; + } + } + + // Joint subband codebook select + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + if (s->joint_intensity_index[ch]) { + s->joint_scale_sel[ch] = get_bits(&s->gb, 3); + if (s->joint_scale_sel[ch] == 7) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint scale factor code book\n"); + return AVERROR_INVALIDDATA; + } + } + } + + // Scale factors for joint subband coding + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + int src_ch = s->joint_intensity_index[ch] - 1; + if (src_ch >= 0) { + int sel = s->joint_scale_sel[ch]; + for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) { + if ((ret = parse_joint_scale(s, sel)) < 0) + return ret; + s->joint_scale_factors[ch][band] = ret; + } + } + } + + // Side information CRC check word + if (s->crc_present) + skip_bits(&s->gb, 16); + + return 0; +} + +static int parse_x96_coding_header(DCACoreDecoder *s, int exss, int xch_base) +{ + int n, ch, header_size = 0, header_pos = get_bits_count(&s->gb); + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + if (exss) { + // Channel set header length + header_size = get_bits(&s->gb, 7) + 1; + + // Check CRC + if (s->x96_crc_present + && (s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 channel set header checksum\n"); + return AVERROR_INVALIDDATA; + } + } + + // High resolution flag + s->x96_high_res = get_bits1(&s->gb); + + // First encoded subband + if (s->x96_rev_no < 8) { + s->x96_subband_start = get_bits(&s->gb, 5); + if (s->x96_subband_start > 27) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband start index (%d)\n", s->x96_subband_start); + return AVERROR_INVALIDDATA; + } + } else { + s->x96_subband_start = DCA_SUBBANDS; + } + + // Subband activity count + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + s->nsubbands[ch] = get_bits(&s->gb, 6) + 1; + if (s->nsubbands[ch] < DCA_SUBBANDS) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband activity count (%d)\n", s->nsubbands[ch]); + return AVERROR_INVALIDDATA; + } + } + + // Joint intensity coding index + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + if ((n = get_bits(&s->gb, 3)) && xch_base) + n += xch_base - 1; + if (n > s->x96_nchannels) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint intensity coding index\n"); + return AVERROR_INVALIDDATA; + } + s->joint_intensity_index[ch] = n; + } + + // Scale factor code book + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + s->scale_factor_sel[ch] = get_bits(&s->gb, 3); + if (s->scale_factor_sel[ch] >= 6) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 scale factor code book\n"); + return AVERROR_INVALIDDATA; + } + } + + // Bit allocation quantizer select + for (ch = xch_base; ch < s->x96_nchannels; ch++) + s->bit_allocation_sel[ch] = get_bits(&s->gb, 3); + + // Quantization index codebook select + for (n = 0; n < 6 + 4 * s->x96_high_res; n++) + for (ch = xch_base; ch < s->x96_nchannels; ch++) + s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]); + + if (exss) { + // Reserved + // Byte align + // CRC16 of channel set header + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set header\n"); + return AVERROR_INVALIDDATA; + } + } else { + if (s->crc_present) + skip_bits(&s->gb, 16); + } + + return 0; +} + +static int parse_x96_frame_data(DCACoreDecoder *s, int exss, int xch_base) +{ + int sf, ch, ret, band, sub_pos; + + if ((ret = parse_x96_coding_header(s, exss, xch_base)) < 0) + return ret; + + for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) { + if ((ret = parse_x96_subframe_header(s, xch_base)) < 0) + return ret; + if ((ret = parse_x96_subframe_audio(s, sf, xch_base, &sub_pos)) < 0) + return ret; + } + + for (ch = xch_base; ch < s->x96_nchannels; ch++) { + // Determine number of active subbands for this channel + int nsubbands = s->nsubbands[ch]; + if (s->joint_intensity_index[ch]) + nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]); + + // Update history for ADPCM and clear inactive subbands + for (band = 0; band < DCA_SUBBANDS_X96; band++) { + int32_t *samples = s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS; + if (band >= s->x96_subband_start && band < nsubbands) + AV_COPY128(samples, samples + s->npcmblocks); + else + memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t)); + } + } + + return 0; +} + +static int parse_x96_frame(DCACoreDecoder *s) +{ + int ret; + + // Revision number + s->x96_rev_no = get_bits(&s->gb, 4); + if (s->x96_rev_no < 1 || s->x96_rev_no > 8) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no); + return AVERROR_INVALIDDATA; + } + + s->x96_crc_present = 0; + s->x96_nchannels = s->nchannels; + + if ((ret = alloc_x96_sample_buffer(s)) < 0) + return ret; + + if ((ret = parse_x96_frame_data(s, 0, 0)) < 0) + return ret; + + // Seek to the end of core frame + if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int parse_x96_frame_exss(DCACoreDecoder *s) +{ + int x96_frame_size[DCA_EXSS_CHSETS_MAX]; + int x96_nchannels[DCA_EXSS_CHSETS_MAX]; + int x96_nchsets, x96_base_ch; + int i, ret, header_size, header_pos = get_bits_count(&s->gb); + + // X96 sync word + if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_X96) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 sync word\n"); + return AVERROR_INVALIDDATA; + } + + // X96 frame header length + header_size = get_bits(&s->gb, 6) + 1; + + // Check X96 frame header CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, header_pos + 32, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 frame header checksum\n"); + return AVERROR_INVALIDDATA; + } + + // Revision number + s->x96_rev_no = get_bits(&s->gb, 4); + if (s->x96_rev_no < 1 || s->x96_rev_no > 8) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no); + return AVERROR_INVALIDDATA; + } + + // CRC presence flag for channel set header + s->x96_crc_present = get_bits1(&s->gb); + + // Number of channel sets + x96_nchsets = get_bits(&s->gb, 2) + 1; + + // Channel set data byte size + for (i = 0; i < x96_nchsets; i++) + x96_frame_size[i] = get_bits(&s->gb, 12) + 1; + + // Number of channels in channel set + for (i = 0; i < x96_nchsets; i++) + x96_nchannels[i] = get_bits(&s->gb, 3) + 1; + + // Reserved + // Byte align + // CRC16 of X96 frame header + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame header\n"); + return AVERROR_INVALIDDATA; + } + + if ((ret = alloc_x96_sample_buffer(s)) < 0) + return ret; + + // Channel set data + for (i = 0, x96_base_ch = 0; i < x96_nchsets; i++) { + header_pos = get_bits_count(&s->gb); + + if (x96_base_ch + x96_nchannels[i] <= s->nchannels) { + s->x96_nchannels = x96_base_ch + x96_nchannels[i]; + if ((ret = parse_x96_frame_data(s, 1, x96_base_ch)) < 0) + return ret; + } + + x96_base_ch += x96_nchannels[i]; + + if (ff_dca_seek_bits(&s->gb, header_pos + x96_frame_size[i] * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set\n"); + return AVERROR_INVALIDDATA; + } + } + + return 0; +} + +static int parse_aux_data(DCACoreDecoder *s) +{ + int aux_pos; + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + // Auxiliary data byte count (can't be trusted) + skip_bits(&s->gb, 6); + + // 4-byte align + skip_bits_long(&s->gb, -get_bits_count(&s->gb) & 31); + + // Auxiliary data sync word + if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_REV1AUX) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data sync word\n"); + return AVERROR_INVALIDDATA; + } + + aux_pos = get_bits_count(&s->gb); + + // Auxiliary decode time stamp flag + if (get_bits1(&s->gb)) + skip_bits_long(&s->gb, 47); + + // Auxiliary dynamic downmix flag + if (s->prim_dmix_embedded = get_bits1(&s->gb)) { + int i, m, n; + + // Auxiliary primary channel downmix type + s->prim_dmix_type = get_bits(&s->gb, 3); + if (s->prim_dmix_type >= DCA_DMIX_TYPE_COUNT) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid primary channel set downmix type\n"); + return AVERROR_INVALIDDATA; + } + + // Size of downmix coefficients matrix + m = ff_dca_dmix_primary_nch[s->prim_dmix_type]; + n = ff_dca_channels[s->audio_mode] + !!s->lfe_present; + + // Dynamic downmix code coefficients + for (i = 0; i < m * n; i++) { + int code = get_bits(&s->gb, 9); + int sign = (code >> 8) - 1; + unsigned int index = code & 0xff; + if (index >= FF_DCA_DMIXTABLE_SIZE) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid downmix coefficient index\n"); + return AVERROR_INVALIDDATA; + } + s->prim_dmix_coeff[i] = (ff_dca_dmixtable[index] ^ sign) - sign; + } + } + + // Byte align + skip_bits(&s->gb, -get_bits_count(&s->gb) & 7); + + // CRC16 of auxiliary data + skip_bits(&s->gb, 16); + + // Check CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, aux_pos, get_bits_count(&s->gb))) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data checksum\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int parse_optional_info(DCACoreDecoder *s) +{ + DCAContext *dca = s->avctx->priv_data; + int ret = -1; + + // Time code stamp + if (s->ts_present) + skip_bits_long(&s->gb, 32); + + // Auxiliary data + if (s->aux_present && (ret = parse_aux_data(s)) < 0 + && (s->avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + + if (ret < 0) + s->prim_dmix_embedded = 0; + + // Core extensions + if (s->ext_audio_present && !dca->core_only) { + int sync_pos = FFMIN(s->frame_size / 4, s->gb.size_in_bits / 32) - 1; + int last_pos = get_bits_count(&s->gb) / 32; + int size, dist; + + // Search for extension sync words aligned on 4-byte boundary. Search + // must be done backwards from the end of core frame to work around + // sync word aliasing issues. + switch (s->ext_audio_type) { + case EXT_AUDIO_XCH: + if (dca->request_channel_layout) + break; + + // The distance between XCH sync word and end of the core frame + // must be equal to XCH frame size. Off by one error is allowed for + // compatibility with legacy bitstreams. Minimum XCH frame size is + // 96 bytes. AMODE and PCHS are further checked to reduce + // probability of alias sync detection. + for (; sync_pos >= last_pos; sync_pos--) { + if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XCH) { + s->gb.index = (sync_pos + 1) * 32; + size = get_bits(&s->gb, 10) + 1; + dist = s->frame_size - sync_pos * 4; + if (size >= 96 + && (size == dist || size - 1 == dist) + && get_bits(&s->gb, 7) == 0x08) { + s->xch_pos = get_bits_count(&s->gb); + break; + } + } + } + + if (s->avctx->err_recognition & AV_EF_EXPLODE) { + av_log(s->avctx, AV_LOG_ERROR, "XCH sync word not found\n"); + return AVERROR_INVALIDDATA; + } + break; + + case EXT_AUDIO_X96: + // The distance between X96 sync word and end of the core frame + // must be equal to X96 frame size. Minimum X96 frame size is 96 + // bytes. + for (; sync_pos >= last_pos; sync_pos--) { + if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_X96) { + s->gb.index = (sync_pos + 1) * 32; + size = get_bits(&s->gb, 12) + 1; + dist = s->frame_size - sync_pos * 4; + if (size >= 96 && size == dist) { + s->x96_pos = get_bits_count(&s->gb); + break; + } + } + } + + if (s->avctx->err_recognition & AV_EF_EXPLODE) { + av_log(s->avctx, AV_LOG_ERROR, "X96 sync word not found\n"); + return AVERROR_INVALIDDATA; + } + break; + + case EXT_AUDIO_XXCH: + if (dca->request_channel_layout) + break; + + // XXCH frame header CRC must be valid. Minimum XXCH frame header + // size is 11 bytes. + for (; sync_pos >= last_pos; sync_pos--) { + if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XXCH) { + s->gb.index = (sync_pos + 1) * 32; + size = get_bits(&s->gb, 6) + 1; + if (size >= 11 && + !ff_dca_check_crc(&s->gb, (sync_pos + 1) * 32, + sync_pos * 32 + size * 8)) { + s->xxch_pos = sync_pos * 32; + break; + } + } + } + + if (s->avctx->err_recognition & AV_EF_EXPLODE) { + av_log(s->avctx, AV_LOG_ERROR, "XXCH sync word not found\n"); + return AVERROR_INVALIDDATA; + } + break; + } + } + + return 0; +} + +int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size) +{ + int ret; + + s->ext_audio_mask = 0; + s->xch_pos = s->xxch_pos = s->x96_pos = 0; + + if ((ret = init_get_bits8(&s->gb, data, size)) < 0) + return ret; + + skip_bits_long(&s->gb, 32); + if ((ret = parse_frame_header(s)) < 0) + return ret; + if ((ret = alloc_sample_buffer(s)) < 0) + return ret; + if ((ret = parse_frame_data(s, HEADER_CORE, 0)) < 0) + return ret; + if ((ret = parse_optional_info(s)) < 0) + return ret; + + // Workaround for DTS in WAV + if (s->frame_size > size && s->frame_size < size + 4) { + av_log(s->avctx, AV_LOG_DEBUG, "Working around excessive core frame size (%d > %d)\n", s->frame_size, size); + s->frame_size = size; + } + + if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of core frame\n"); + if (s->avctx->err_recognition & AV_EF_EXPLODE) + return AVERROR_INVALIDDATA; + } + + return 0; +} + +int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset) +{ + AVCodecContext *avctx = s->avctx; + DCAContext *dca = avctx->priv_data; + GetBitContext gb = s->gb; + int exss_mask = asset ? asset->extension_mask : 0; + int ret = 0, ext = 0; + + // Parse (X)XCH unless downmixing + if (!dca->request_channel_layout) { + if (exss_mask & DCA_EXSS_XXCH) { + if ((ret = init_get_bits8(&s->gb, data + asset->xxch_offset, asset->xxch_size)) < 0) + return ret; + ret = parse_xxch_frame(s); + ext = DCA_EXSS_XXCH; + } else if (s->xxch_pos) { + s->gb.index = s->xxch_pos; + ret = parse_xxch_frame(s); + ext = DCA_CSS_XXCH; + } else if (s->xch_pos) { + s->gb.index = s->xch_pos; + ret = parse_xch_frame(s); + ext = DCA_CSS_XCH; + } + + // Revert to primary channel set in case (X)XCH parsing fails + if (ret < 0) { + if (avctx->err_recognition & AV_EF_EXPLODE) + return ret; + s->nchannels = ff_dca_channels[s->audio_mode]; + s->ch_mask = audio_mode_ch_mask[s->audio_mode]; + if (s->lfe_present) + s->ch_mask |= DCA_SPEAKER_MASK_LFE1; + } else { + s->ext_audio_mask |= ext; + } + } + + // Parse XBR + if (exss_mask & DCA_EXSS_XBR) { + if ((ret = init_get_bits8(&s->gb, data + asset->xbr_offset, asset->xbr_size)) < 0) + return ret; + if ((ret = parse_xbr_frame(s)) < 0) { + if (avctx->err_recognition & AV_EF_EXPLODE) + return ret; + } else { + s->ext_audio_mask |= DCA_EXSS_XBR; + } + } + + // Parse X96 unless decoding XLL + if (!(dca->packet & DCA_PACKET_XLL)) { + if (exss_mask & DCA_EXSS_X96) { + if ((ret = init_get_bits8(&s->gb, data + asset->x96_offset, asset->x96_size)) < 0) + return ret; + if ((ret = parse_x96_frame_exss(s)) < 0) { + if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + } else { + s->ext_audio_mask |= DCA_EXSS_X96; + } + } else if (s->x96_pos) { + s->gb = gb; + s->gb.index = s->x96_pos; + if ((ret = parse_x96_frame(s)) < 0) { + if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + } else { + s->ext_audio_mask |= DCA_CSS_X96; + } + } + } + + return 0; +} + +static int map_prm_ch_to_spkr(DCACoreDecoder *s, int ch) +{ + int pos, spkr; + + // Try to map this channel to core first + pos = ff_dca_channels[s->audio_mode]; + if (ch < pos) { + spkr = prm_ch_to_spkr_map[s->audio_mode][ch]; + if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) { + if (s->xxch_core_mask & (1U << spkr)) + return spkr; + if (spkr == DCA_SPEAKER_Ls && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss)) + return DCA_SPEAKER_Lss; + if (spkr == DCA_SPEAKER_Rs && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss)) + return DCA_SPEAKER_Rss; + return -1; + } + return spkr; + } + + // Then XCH + if ((s->ext_audio_mask & DCA_CSS_XCH) && ch == pos) + return DCA_SPEAKER_Cs; + + // Then XXCH + if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) { + for (spkr = DCA_SPEAKER_Cs; spkr < s->xxch_mask_nbits; spkr++) + if (s->xxch_spkr_mask & (1U << spkr)) + if (pos++ == ch) + return spkr; + } + + // No mapping + return -1; +} + +static void erase_dsp_history(DCACoreDecoder *s) +{ + memset(s->dcadsp_data, 0, sizeof(s->dcadsp_data)); + s->output_history_lfe_fixed = 0; + s->output_history_lfe_float = 0; +} + +static void set_filter_mode(DCACoreDecoder *s, int mode) +{ + if (s->filter_mode != mode) { + erase_dsp_history(s); + s->filter_mode = mode; + } +} + +int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth) +{ + int n, ch, spkr, nsamples, x96_nchannels = 0; + const int32_t *filter_coeff; + int32_t *ptr; + + // Externally set x96_synth flag implies that X96 synthesis should be + // enabled, yet actual X96 subband data should be discarded. This is a + // special case for lossless residual decoder that ignores X96 data if + // present. + if (!x96_synth && (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96))) { + x96_nchannels = s->x96_nchannels; + x96_synth = 1; + } + if (x96_synth < 0) + x96_synth = 0; + + s->output_rate = s->sample_rate << x96_synth; + s->npcmsamples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth; + + // Reallocate PCM output buffer + av_fast_malloc(&s->output_buffer, &s->output_size, + nsamples * av_popcount(s->ch_mask) * sizeof(int32_t)); + if (!s->output_buffer) + return AVERROR(ENOMEM); + + ptr = (int32_t *)s->output_buffer; + for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) { + if (s->ch_mask & (1U << spkr)) { + s->output_samples[spkr] = ptr; + ptr += nsamples; + } else { + s->output_samples[spkr] = NULL; + } + } + + // Handle change of filtering mode + set_filter_mode(s, x96_synth | DCA_FILTER_MODE_FIXED); + + // Select filter + if (x96_synth) + filter_coeff = ff_dca_fir_64bands_fixed; + else if (s->filter_perfect) + filter_coeff = ff_dca_fir_32bands_perfect_fixed; + else + filter_coeff = ff_dca_fir_32bands_nonperfect_fixed; + + // Filter primary channels + for (ch = 0; ch < s->nchannels; ch++) { + // Map this primary channel to speaker + spkr = map_prm_ch_to_spkr(s, ch); + if (spkr < 0) + return AVERROR(EINVAL); + + // Filter bank reconstruction + s->dcadsp->sub_qmf_fixed[x96_synth]( + &s->synth, + &s->dcadct, + s->output_samples[spkr], + s->subband_samples[ch], + ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL, + s->dcadsp_data[ch].u.fix.hist1, + &s->dcadsp_data[ch].offset, + s->dcadsp_data[ch].u.fix.hist2, + filter_coeff, + s->npcmblocks); + } + + // Filter LFE channel + if (s->lfe_present) { + int32_t *samples = s->output_samples[DCA_SPEAKER_LFE1]; + int nlfesamples = s->npcmblocks >> 1; + + // Check LFF + if (s->lfe_present == LFE_FLAG_128) { + av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n"); + return AVERROR(EINVAL); + } + + // Offset intermediate buffer for X96 + if (x96_synth) + samples += nsamples / 2; + + // Interpolate LFE channel + s->dcadsp->lfe_fir_fixed(samples, s->lfe_samples + DCA_LFE_HISTORY, + ff_dca_lfe_fir_64_fixed, s->npcmblocks); + + if (x96_synth) { + // Filter 96 kHz oversampled LFE PCM to attenuate high frequency + // (47.6 - 48.0 kHz) components of interpolation image + s->dcadsp->lfe_x96_fixed(s->output_samples[DCA_SPEAKER_LFE1], + samples, &s->output_history_lfe_fixed, + nsamples / 2); + + } + + // Update LFE history + for (n = DCA_LFE_HISTORY - 1; n >= 0; n--) + s->lfe_samples[n] = s->lfe_samples[nlfesamples + n]; + } + + return 0; +} + +static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame) +{ + AVCodecContext *avctx = s->avctx; + DCAContext *dca = avctx->priv_data; + int i, n, ch, ret, spkr, nsamples; + + // Don't filter twice when falling back from XLL + if (!(dca->packet & DCA_PACKET_XLL) && (ret = ff_dca_core_filter_fixed(s, 0)) < 0) + return ret; + + avctx->sample_rate = s->output_rate; + avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + avctx->bits_per_raw_sample = 24; + + frame->nb_samples = nsamples = s->npcmsamples; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + // Undo embedded XCH downmix + if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH) + && s->audio_mode >= AMODE_2F2R) { + s->dcadsp->dmix_sub_xch(s->output_samples[DCA_SPEAKER_Ls], + s->output_samples[DCA_SPEAKER_Rs], + s->output_samples[DCA_SPEAKER_Cs], + nsamples); + + } + + // Undo embedded XXCH downmix + if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) + && s->xxch_dmix_embedded) { + int scale_inv = s->xxch_dmix_scale_inv; + int *coeff_ptr = s->xxch_dmix_coeff; + int xch_base = ff_dca_channels[s->audio_mode]; + av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX); + + // Undo embedded core downmix pre-scaling + for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) { + if (s->xxch_core_mask & (1U << spkr)) { + s->dcadsp->dmix_scale_inv(s->output_samples[spkr], + scale_inv, nsamples); + } + } + + // Undo downmix + for (ch = xch_base; ch < s->nchannels; ch++) { + int src_spkr = map_prm_ch_to_spkr(s, ch); + if (src_spkr < 0) + return AVERROR(EINVAL); + for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) { + if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) { + int coeff = mul16(*coeff_ptr++, scale_inv); + if (coeff) { + s->dcadsp->dmix_sub(s->output_samples[spkr ], + s->output_samples[src_spkr], + coeff, nsamples); + } + } + } + } + } + + if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) { + // Front sum/difference decoding + if ((s->sumdiff_front && s->audio_mode > AMODE_MONO) + || s->audio_mode == AMODE_STEREO_SUMDIFF) { + s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_L], + s->output_samples[DCA_SPEAKER_R], + nsamples); + } + + // Surround sum/difference decoding + if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) { + s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls], + s->output_samples[DCA_SPEAKER_Rs], + nsamples); + } + } + + // Downmix primary channel set to stereo + if (s->request_mask != s->ch_mask) { + ff_dca_downmix_to_stereo_fixed(s->dcadsp, + s->output_samples, + s->prim_dmix_coeff, + nsamples, s->ch_mask); + } + + for (i = 0; i < avctx->channels; i++) { + int32_t *samples = s->output_samples[s->ch_remap[i]]; + int32_t *plane = (int32_t *)frame->extended_data[i]; + for (n = 0; n < nsamples; n++) + plane[n] = clip23(samples[n]) * (1 << 8); + } + + return 0; +} + +static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame) +{ + AVCodecContext *avctx = s->avctx; + int x96_nchannels = 0, x96_synth = 0; + int i, n, ch, ret, spkr, nsamples, nchannels; + float *output_samples[DCA_SPEAKER_COUNT] = { NULL }, *ptr; + const float *filter_coeff; + + if (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96)) { + x96_nchannels = s->x96_nchannels; + x96_synth = 1; + } + + avctx->sample_rate = s->sample_rate << x96_synth; + avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; + avctx->bits_per_raw_sample = 0; + + frame->nb_samples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth; + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + // Build reverse speaker to channel mapping + for (i = 0; i < avctx->channels; i++) + output_samples[s->ch_remap[i]] = (float *)frame->extended_data[i]; + + // Allocate space for extra channels + nchannels = av_popcount(s->ch_mask) - avctx->channels; + if (nchannels > 0) { + av_fast_malloc(&s->output_buffer, &s->output_size, + nsamples * nchannels * sizeof(float)); + if (!s->output_buffer) + return AVERROR(ENOMEM); + + ptr = (float *)s->output_buffer; + for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) { + if (!(s->ch_mask & (1U << spkr))) + continue; + if (output_samples[spkr]) + continue; + output_samples[spkr] = ptr; + ptr += nsamples; + } + } + + // Handle change of filtering mode + set_filter_mode(s, x96_synth); + + // Select filter + if (x96_synth) + filter_coeff = ff_dca_fir_64bands; + else if (s->filter_perfect) + filter_coeff = ff_dca_fir_32bands_perfect; + else + filter_coeff = ff_dca_fir_32bands_nonperfect; + + // Filter primary channels + for (ch = 0; ch < s->nchannels; ch++) { + // Map this primary channel to speaker + spkr = map_prm_ch_to_spkr(s, ch); + if (spkr < 0) + return AVERROR(EINVAL); + + // Filter bank reconstruction + s->dcadsp->sub_qmf_float[x96_synth]( + &s->synth, + &s->imdct[x96_synth], + output_samples[spkr], + s->subband_samples[ch], + ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL, + s->dcadsp_data[ch].u.flt.hist1, + &s->dcadsp_data[ch].offset, + s->dcadsp_data[ch].u.flt.hist2, + filter_coeff, + s->npcmblocks, + 1.0f / (1 << (17 - x96_synth))); + } + + // Filter LFE channel + if (s->lfe_present) { + int dec_select = (s->lfe_present == LFE_FLAG_128); + float *samples = output_samples[DCA_SPEAKER_LFE1]; + int nlfesamples = s->npcmblocks >> (dec_select + 1); + + // Offset intermediate buffer for X96 + if (x96_synth) + samples += nsamples / 2; + + // Select filter + if (dec_select) + filter_coeff = ff_dca_lfe_fir_128; + else + filter_coeff = ff_dca_lfe_fir_64; + + // Interpolate LFE channel + s->dcadsp->lfe_fir_float[dec_select]( + samples, s->lfe_samples + DCA_LFE_HISTORY, + filter_coeff, s->npcmblocks); + + if (x96_synth) { + // Filter 96 kHz oversampled LFE PCM to attenuate high frequency + // (47.6 - 48.0 kHz) components of interpolation image + s->dcadsp->lfe_x96_float(output_samples[DCA_SPEAKER_LFE1], + samples, &s->output_history_lfe_float, + nsamples / 2); + } + + // Update LFE history + for (n = DCA_LFE_HISTORY - 1; n >= 0; n--) + s->lfe_samples[n] = s->lfe_samples[nlfesamples + n]; + } + + // Undo embedded XCH downmix + if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH) + && s->audio_mode >= AMODE_2F2R) { + s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls], + output_samples[DCA_SPEAKER_Cs], + -M_SQRT1_2, nsamples); + s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Rs], + output_samples[DCA_SPEAKER_Cs], + -M_SQRT1_2, nsamples); + } + + // Undo embedded XXCH downmix + if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) + && s->xxch_dmix_embedded) { + float scale_inv = s->xxch_dmix_scale_inv * (1.0f / (1 << 16)); + int *coeff_ptr = s->xxch_dmix_coeff; + int xch_base = ff_dca_channels[s->audio_mode]; + av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX); + + // Undo downmix + for (ch = xch_base; ch < s->nchannels; ch++) { + int src_spkr = map_prm_ch_to_spkr(s, ch); + if (src_spkr < 0) + return AVERROR(EINVAL); + for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) { + if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) { + int coeff = *coeff_ptr++; + if (coeff) { + s->float_dsp->vector_fmac_scalar(output_samples[ spkr], + output_samples[src_spkr], + coeff * (-1.0f / (1 << 15)), + nsamples); + } + } + } + } + + // Undo embedded core downmix pre-scaling + for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) { + if (s->xxch_core_mask & (1U << spkr)) { + s->float_dsp->vector_fmul_scalar(output_samples[spkr], + output_samples[spkr], + scale_inv, nsamples); + } + } + } + + if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) { + // Front sum/difference decoding + if ((s->sumdiff_front && s->audio_mode > AMODE_MONO) + || s->audio_mode == AMODE_STEREO_SUMDIFF) { + s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_L], + output_samples[DCA_SPEAKER_R], + nsamples); + } + + // Surround sum/difference decoding + if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) { + s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls], + output_samples[DCA_SPEAKER_Rs], + nsamples); + } + } + + // Downmix primary channel set to stereo + if (s->request_mask != s->ch_mask) { + ff_dca_downmix_to_stereo_float(s->float_dsp, output_samples, + s->prim_dmix_coeff, + nsamples, s->ch_mask); + } + + return 0; +} + +int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame) +{ + AVCodecContext *avctx = s->avctx; + DCAContext *dca = avctx->priv_data; + DCAExssAsset *asset = &dca->exss.assets[0]; + enum AVMatrixEncoding matrix_encoding; + int ret; + + // Handle downmixing to stereo request + if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO + && s->audio_mode > AMODE_MONO && s->prim_dmix_embedded + && (s->prim_dmix_type == DCA_DMIX_TYPE_LoRo || + s->prim_dmix_type == DCA_DMIX_TYPE_LtRt)) + s->request_mask = DCA_SPEAKER_LAYOUT_STEREO; + else + s->request_mask = s->ch_mask; + if (!ff_dca_set_channel_layout(avctx, s->ch_remap, s->request_mask)) + return AVERROR(EINVAL); + + // Force fixed point mode when falling back from XLL + if ((avctx->flags & AV_CODEC_FLAG_BITEXACT) || ((dca->packet & DCA_PACKET_EXSS) + && (asset->extension_mask & DCA_EXSS_XLL))) + ret = filter_frame_fixed(s, frame); + else + ret = filter_frame_float(s, frame); + if (ret < 0) + return ret; + + // Set profile, bit rate, etc + if (s->ext_audio_mask & DCA_EXSS_MASK) + avctx->profile = FF_PROFILE_DTS_HD_HRA; + else if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH)) + avctx->profile = FF_PROFILE_DTS_ES; + else if (s->ext_audio_mask & DCA_CSS_X96) + avctx->profile = FF_PROFILE_DTS_96_24; + else + avctx->profile = FF_PROFILE_DTS; + + if (s->bit_rate > 3 && !(s->ext_audio_mask & DCA_EXSS_MASK)) + avctx->bit_rate = s->bit_rate; + else + avctx->bit_rate = 0; + + if (s->audio_mode == AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask && + s->prim_dmix_type == DCA_DMIX_TYPE_LtRt)) + matrix_encoding = AV_MATRIX_ENCODING_DOLBY; + else + matrix_encoding = AV_MATRIX_ENCODING_NONE; + if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0) + return ret; + + return 0; +} + +av_cold void ff_dca_core_flush(DCACoreDecoder *s) +{ + if (s->subband_buffer) { + erase_adpcm_history(s); + memset(s->lfe_samples, 0, DCA_LFE_HISTORY * sizeof(int32_t)); + } + + if (s->x96_subband_buffer) + erase_x96_adpcm_history(s); + + erase_dsp_history(s); +} + +av_cold int ff_dca_core_init(DCACoreDecoder *s) +{ + dca_init_vlcs(); + + if (!(s->float_dsp = avpriv_float_dsp_alloc(0))) + return -1; + if (!(s->fixed_dsp = avpriv_alloc_fixed_dsp(0))) + return -1; + + ff_dcadct_init(&s->dcadct); + if (ff_mdct_init(&s->imdct[0], 6, 1, 1.0) < 0) + return -1; + if (ff_mdct_init(&s->imdct[1], 7, 1, 1.0) < 0) + return -1; + ff_synth_filter_init(&s->synth); + + s->x96_rand = 1; + return 0; +} + +av_cold void ff_dca_core_close(DCACoreDecoder *s) +{ + av_freep(&s->float_dsp); + av_freep(&s->fixed_dsp); + + ff_mdct_end(&s->imdct[0]); + ff_mdct_end(&s->imdct[1]); + + av_freep(&s->subband_buffer); + s->subband_size = 0; + + av_freep(&s->x96_subband_buffer); + s->x96_subband_size = 0; + + av_freep(&s->output_buffer); + s->output_size = 0; +} diff --git a/libavcodec/dca_core.h b/libavcodec/dca_core.h new file mode 100644 index 0000000..112b72b --- /dev/null +++ b/libavcodec/dca_core.h @@ -0,0 +1,206 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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_DCA_CORE_H +#define AVCODEC_DCA_CORE_H + +#include "libavutil/common.h" +#include "libavutil/float_dsp.h" +#include "libavutil/fixed_dsp.h" +#include "libavutil/mem.h" + +#include "avcodec.h" +#include "internal.h" +#include "get_bits.h" +#include "dca.h" +#include "dca_exss.h" +#include "dcadsp.h" +#include "dcadct.h" +#include "fft.h" +#include "synth_filter.h" + +#define DCA_CHANNELS 7 +#define DCA_SUBBANDS 32 +#define DCA_SUBBANDS_X96 64 +#define DCA_SUBFRAMES 16 +#define DCA_SUBBAND_SAMPLES 8 +#define DCA_PCMBLOCK_SAMPLES 32 +#define DCA_ADPCM_COEFFS 4 +#define DCA_LFE_HISTORY 8 +#define DCA_CODE_BOOKS 10 +#define DCA_ABITS_MAX 26 + +#define DCA_CORE_CHANNELS_MAX 6 +#define DCA_DMIX_CHANNELS_MAX 4 +#define DCA_XXCH_CHANNELS_MAX 2 +#define DCA_EXSS_CHANNELS_MAX 8 +#define DCA_EXSS_CHSETS_MAX 4 + +#define DCA_FILTER_MODE_X96 0x01 +#define DCA_FILTER_MODE_FIXED 0x02 + +typedef struct DCADSPData { + union { + struct { + DECLARE_ALIGNED(32, float, hist1)[1024]; + DECLARE_ALIGNED(32, float, hist2)[64]; + } flt; + struct { + DECLARE_ALIGNED(32, int32_t, hist1)[1024]; + DECLARE_ALIGNED(32, int32_t, hist2)[64]; + } fix; + } u; + int offset; +} DCADSPData; + +typedef struct DCACoreDecoder { + AVCodecContext *avctx; + GetBitContext gb; + + // Bit stream header + int crc_present; ///< CRC present flag + int npcmblocks; ///< Number of PCM sample blocks + int frame_size; ///< Primary frame byte size + int audio_mode; ///< Audio channel arrangement + int sample_rate; ///< Core audio sampling frequency + int bit_rate; ///< Transmission bit rate + int drc_present; ///< Embedded dynamic range flag + int ts_present; ///< Embedded time stamp flag + int aux_present; ///< Auxiliary data flag + int ext_audio_type; ///< Extension audio descriptor flag + int ext_audio_present; ///< Extended coding flag + int sync_ssf; ///< Audio sync word insertion flag + int lfe_present; ///< Low frequency effects flag + int predictor_history; ///< Predictor history flag switch + int filter_perfect; ///< Multirate interpolator switch + int source_pcm_res; ///< Source PCM resolution + int es_format; ///< Extended surround (ES) mastering flag + int sumdiff_front; ///< Front sum/difference flag + int sumdiff_surround; ///< Surround sum/difference flag + + // Primary audio coding header + int nsubframes; ///< Number of subframes + int nchannels; ///< Number of primary audio channels (incl. extension channels) + int ch_mask; ///< Speaker layout mask (incl. LFE and extension channels) + int8_t nsubbands[DCA_CHANNELS]; ///< Subband activity count + int8_t subband_vq_start[DCA_CHANNELS]; ///< High frequency VQ start subband + int8_t joint_intensity_index[DCA_CHANNELS]; ///< Joint intensity coding index + int8_t transition_mode_sel[DCA_CHANNELS]; ///< Transient mode code book + int8_t scale_factor_sel[DCA_CHANNELS]; ///< Scale factor code book + int8_t bit_allocation_sel[DCA_CHANNELS]; ///< Bit allocation quantizer select + int8_t quant_index_sel[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Quantization index codebook select + int32_t scale_factor_adj[DCA_CHANNELS][DCA_CODE_BOOKS]; ///< Scale factor adjustment + + // Primary audio coding side information + int8_t nsubsubframes[DCA_SUBFRAMES]; ///< Subsubframe count for each subframe + int8_t prediction_mode[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction mode + int16_t prediction_vq_index[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Prediction coefficients VQ address + int8_t bit_allocation[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Bit allocation index + int8_t transition_mode[DCA_SUBFRAMES][DCA_CHANNELS][DCA_SUBBANDS]; ///< Transition mode + int32_t scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2]; ///< Scale factors (2x for transients and X96) + int8_t joint_scale_sel[DCA_CHANNELS]; ///< Joint subband codebook select + int32_t joint_scale_factors[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< Scale factors for joint subband coding + + // Auxiliary data + int prim_dmix_embedded; ///< Auxiliary dynamic downmix flag + int prim_dmix_type; ///< Auxiliary primary channel downmix type + int prim_dmix_coeff[DCA_DMIX_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Dynamic downmix code coefficients + + // Core extensions + int ext_audio_mask; ///< Bit mask of fully decoded core extensions + + // XCH extension data + int xch_pos; ///< Bit position of XCH frame in core substream + + // XXCH extension data + int xxch_crc_present; ///< CRC presence flag for XXCH channel set header + int xxch_mask_nbits; ///< Number of bits for loudspeaker mask + int xxch_core_mask; ///< Core loudspeaker activity mask + int xxch_spkr_mask; ///< Loudspeaker layout mask + int xxch_dmix_embedded; ///< Downmix already performed by encoder + int xxch_dmix_scale_inv; ///< Downmix scale factor + int xxch_dmix_mask[DCA_XXCH_CHANNELS_MAX]; ///< Downmix channel mapping mask + int xxch_dmix_coeff[DCA_XXCH_CHANNELS_MAX * DCA_CORE_CHANNELS_MAX]; ///< Downmix coefficients + int xxch_pos; ///< Bit position of XXCH frame in core substream + + // X96 extension data + int x96_rev_no; ///< X96 revision number + int x96_crc_present; ///< CRC presence flag for X96 channel set header + int x96_nchannels; ///< Number of primary channels in X96 extension + int x96_high_res; ///< X96 high resolution flag + int x96_subband_start; ///< First encoded subband in X96 extension + int x96_rand; ///< Random seed for generating samples for unallocated X96 subbands + int x96_pos; ///< Bit position of X96 frame in core substream + + // Sample buffers + unsigned int x96_subband_size; + int32_t *x96_subband_buffer; ///< X96 subband sample buffer base + int32_t *x96_subband_samples[DCA_CHANNELS][DCA_SUBBANDS_X96]; ///< X96 subband samples + + unsigned int subband_size; + int32_t *subband_buffer; ///< Subband sample buffer base + int32_t *subband_samples[DCA_CHANNELS][DCA_SUBBANDS]; ///< Subband samples + int32_t *lfe_samples; ///< Decimated LFE samples + + // DSP contexts + DCADSPData dcadsp_data[DCA_CHANNELS]; ///< FIR history buffers + DCADSPContext *dcadsp; + DCADCTContext dcadct; + FFTContext imdct[2]; + SynthFilterContext synth; + AVFloatDSPContext *float_dsp; + AVFixedDSPContext *fixed_dsp; + + // PCM output data + unsigned int output_size; + void *output_buffer; ///< PCM output buffer base + int32_t *output_samples[DCA_SPEAKER_COUNT]; ///< PCM output for fixed point mode + int32_t output_history_lfe_fixed; ///< LFE PCM history for X96 filter + float output_history_lfe_float; ///< LFE PCM history for X96 filter + + int ch_remap[DCA_SPEAKER_COUNT]; ///< Channel to speaker map + int request_mask; ///< Requested channel layout (for stereo downmix) + + int npcmsamples; ///< Number of PCM samples per channel + int output_rate; ///< Output sample rate (1x or 2x header rate) + + int filter_mode; ///< Previous filtering mode for detecting changes +} DCACoreDecoder; + +static inline int ff_dca_core_map_spkr(DCACoreDecoder *core, int spkr) +{ + if (core->ch_mask & (1U << spkr)) + return spkr; + if (spkr == DCA_SPEAKER_Lss && (core->ch_mask & DCA_SPEAKER_MASK_Ls)) + return DCA_SPEAKER_Ls; + if (spkr == DCA_SPEAKER_Rss && (core->ch_mask & DCA_SPEAKER_MASK_Rs)) + return DCA_SPEAKER_Rs; + return -1; +} + +int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size); +int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset); +int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth); +int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame); +av_cold void ff_dca_core_flush(DCACoreDecoder *s); +av_cold int ff_dca_core_init(DCACoreDecoder *s); +av_cold void ff_dca_core_close(DCACoreDecoder *s); + +#endif diff --git a/libavcodec/dca_exss.c b/libavcodec/dca_exss.c new file mode 100644 index 0000000..4579f23 --- /dev/null +++ b/libavcodec/dca_exss.c @@ -0,0 +1,514 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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 "dcadec.h" +#include "dcadata.h" + +static int count_chs_for_mask(int mask) +{ + return av_popcount(mask) + av_popcount(mask & 0xae66); +} + +static void parse_xll_parameters(DCAExssParser *s, DCAExssAsset *asset) +{ + // Size of XLL data in extension substream + asset->xll_size = get_bits(&s->gb, s->exss_size_nbits) + 1; + + // XLL sync word present flag + if (asset->xll_sync_present = get_bits1(&s->gb)) { + int xll_delay_nbits; + + // Peak bit rate smoothing buffer size + skip_bits(&s->gb, 4); + + // Number of bits for XLL decoding delay + xll_delay_nbits = get_bits(&s->gb, 5) + 1; + + // Initial XLL decoding delay in frames + asset->xll_delay_nframes = get_bits_long(&s->gb, xll_delay_nbits); + + // Number of bytes offset to XLL sync + asset->xll_sync_offset = get_bits(&s->gb, s->exss_size_nbits); + } else { + asset->xll_delay_nframes = 0; + asset->xll_sync_offset = 0; + } +} + +static void parse_lbr_parameters(DCAExssParser *s, DCAExssAsset *asset) +{ + // Size of LBR component in extension substream + asset->lbr_size = get_bits(&s->gb, 14) + 1; + + // LBR sync word present flag + if (get_bits1(&s->gb)) + // LBR sync distance + skip_bits(&s->gb, 2); +} + +static int parse_descriptor(DCAExssParser *s, DCAExssAsset *asset) +{ + int i, j, drc_present, descr_size, descr_pos = get_bits_count(&s->gb); + + // Size of audio asset descriptor in bytes + descr_size = get_bits(&s->gb, 9) + 1; + + // Audio asset identifier + asset->asset_index = get_bits(&s->gb, 3); + + // + // Per stream static metadata + // + + if (s->static_fields_present) { + // Asset type descriptor presence + if (get_bits1(&s->gb)) + // Asset type descriptor + skip_bits(&s->gb, 4); + + // Language descriptor presence + if (get_bits1(&s->gb)) + // Language descriptor + skip_bits(&s->gb, 24); + + // Additional textual information presence + if (get_bits1(&s->gb)) { + // Byte size of additional text info + int text_size = get_bits(&s->gb, 10) + 1; + + // Sanity check available size + if (get_bits_left(&s->gb) < text_size * 8) + return AVERROR_INVALIDDATA; + + // Additional textual information string + skip_bits_long(&s->gb, text_size * 8); + } + + // PCM bit resolution + asset->pcm_bit_res = get_bits(&s->gb, 5) + 1; + + // Maximum sample rate + asset->max_sample_rate = ff_dca_sampling_freqs[get_bits(&s->gb, 4)]; + + // Total number of channels + asset->nchannels_total = get_bits(&s->gb, 8) + 1; + + // One to one map channel to speakers + if (asset->one_to_one_map_ch_to_spkr = get_bits1(&s->gb)) { + int spkr_mask_nbits = 0; + int spkr_remap_nsets; + int nspeakers[8]; + + // Embedded stereo flag + if (asset->nchannels_total > 2) + asset->embedded_stereo = get_bits1(&s->gb); + + // Embedded 6 channels flag + if (asset->nchannels_total > 6) + asset->embedded_6ch = get_bits1(&s->gb); + + // Speaker mask enabled flag + if (asset->spkr_mask_enabled = get_bits1(&s->gb)) { + // Number of bits for speaker activity mask + spkr_mask_nbits = (get_bits(&s->gb, 2) + 1) << 2; + + // Loudspeaker activity mask + asset->spkr_mask = get_bits(&s->gb, spkr_mask_nbits); + } + + // Number of speaker remapping sets + if ((spkr_remap_nsets = get_bits(&s->gb, 3)) && !spkr_mask_nbits) { + av_log(s->avctx, AV_LOG_ERROR, "Speaker mask disabled yet there are remapping sets\n"); + return AVERROR_INVALIDDATA; + } + + // Standard loudspeaker layout mask + for (i = 0; i < spkr_remap_nsets; i++) + nspeakers[i] = count_chs_for_mask(get_bits(&s->gb, spkr_mask_nbits)); + + for (i = 0; i < spkr_remap_nsets; i++) { + // Number of channels to be decoded for speaker remapping + int nch_for_remaps = get_bits(&s->gb, 5) + 1; + + for (j = 0; j < nspeakers[i]; j++) { + // Decoded channels to output speaker mapping mask + int remap_ch_mask = get_bits_long(&s->gb, nch_for_remaps); + + // Loudspeaker remapping codes + skip_bits_long(&s->gb, av_popcount(remap_ch_mask) * 5); + } + } + } else { + asset->embedded_stereo = 0; + asset->embedded_6ch = 0; + asset->spkr_mask_enabled = 0; + asset->spkr_mask = 0; + + // Representation type + asset->representation_type = get_bits(&s->gb, 3); + } + } + + // + // DRC, DNC and mixing metadata + // + + // Dynamic range coefficient presence flag + drc_present = get_bits1(&s->gb); + + // Code for dynamic range coefficient + if (drc_present) + skip_bits(&s->gb, 8); + + // Dialog normalization presence flag + if (get_bits1(&s->gb)) + // Dialog normalization code + skip_bits(&s->gb, 5); + + // DRC for stereo downmix + if (drc_present && asset->embedded_stereo) + skip_bits(&s->gb, 8); + + // Mixing metadata presence flag + if (s->mix_metadata_enabled && get_bits1(&s->gb)) { + int nchannels_dmix; + + // External mixing flag + skip_bits1(&s->gb); + + // Post mixing / replacement gain adjustment + skip_bits(&s->gb, 6); + + // DRC prior to mixing + if (get_bits(&s->gb, 2) == 3) + // Custom code for mixing DRC + skip_bits(&s->gb, 8); + else + // Limit for mixing DRC + skip_bits(&s->gb, 3); + + // Scaling type for channels of main audio + // Scaling parameters of main audio + if (get_bits1(&s->gb)) + for (i = 0; i < s->nmixoutconfigs; i++) + skip_bits_long(&s->gb, 6 * s->nmixoutchs[i]); + else + skip_bits_long(&s->gb, 6 * s->nmixoutconfigs); + + nchannels_dmix = asset->nchannels_total; + if (asset->embedded_6ch) + nchannels_dmix += 6; + if (asset->embedded_stereo) + nchannels_dmix += 2; + + for (i = 0; i < s->nmixoutconfigs; i++) { + if (!s->nmixoutchs[i]) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid speaker layout mask for mixing configuration\n"); + return AVERROR_INVALIDDATA; + } + for (j = 0; j < nchannels_dmix; j++) { + // Mix output mask + int mix_map_mask = get_bits(&s->gb, s->nmixoutchs[i]); + + // Mixing coefficients + skip_bits_long(&s->gb, av_popcount(mix_map_mask) * 6); + } + } + } + + // + // Decoder navigation data + // + + // Coding mode for the asset + asset->coding_mode = get_bits(&s->gb, 2); + + // Coding components used in asset + switch (asset->coding_mode) { + case 0: // Coding mode that may contain multiple coding components + asset->extension_mask = get_bits(&s->gb, 12); + + if (asset->extension_mask & DCA_EXSS_CORE) { + // Size of core component in extension substream + asset->core_size = get_bits(&s->gb, 14) + 1; + // Core sync word present flag + if (get_bits1(&s->gb)) + // Core sync distance + skip_bits(&s->gb, 2); + } + + if (asset->extension_mask & DCA_EXSS_XBR) + // Size of XBR extension in extension substream + asset->xbr_size = get_bits(&s->gb, 14) + 1; + + if (asset->extension_mask & DCA_EXSS_XXCH) + // Size of XXCH extension in extension substream + asset->xxch_size = get_bits(&s->gb, 14) + 1; + + if (asset->extension_mask & DCA_EXSS_X96) + // Size of X96 extension in extension substream + asset->x96_size = get_bits(&s->gb, 12) + 1; + + if (asset->extension_mask & DCA_EXSS_LBR) + parse_lbr_parameters(s, asset); + + if (asset->extension_mask & DCA_EXSS_XLL) + parse_xll_parameters(s, asset); + + if (asset->extension_mask & DCA_EXSS_RSV1) + skip_bits(&s->gb, 16); + + if (asset->extension_mask & DCA_EXSS_RSV2) + skip_bits(&s->gb, 16); + break; + + case 1: // Loss-less coding mode without CBR component + asset->extension_mask = DCA_EXSS_XLL; + parse_xll_parameters(s, asset); + break; + + case 2: // Low bit rate mode + asset->extension_mask = DCA_EXSS_LBR; + parse_lbr_parameters(s, asset); + break; + + case 3: // Auxiliary coding mode + asset->extension_mask = 0; + + // Size of auxiliary coded data + skip_bits(&s->gb, 14); + + // Auxiliary codec identification + skip_bits(&s->gb, 8); + + // Aux sync word present flag + if (get_bits1(&s->gb)) + // Aux sync distance + skip_bits(&s->gb, 3); + break; + } + + if (asset->extension_mask & DCA_EXSS_XLL) + // DTS-HD stream ID + asset->hd_stream_id = get_bits(&s->gb, 3); + + // One to one mixing flag + // Per channel main audio scaling flag + // Main audio scaling codes + // Decode asset in secondary decoder flag + // Revision 2 DRC metadata + // Reserved + // Zero pad + if (ff_dca_seek_bits(&s->gb, descr_pos + descr_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of EXSS asset descriptor\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int set_exss_offsets(DCAExssAsset *asset) +{ + int offs = asset->asset_offset; + int size = asset->asset_size; + + if (asset->extension_mask & DCA_EXSS_CORE) { + asset->core_offset = offs; + if (asset->core_size > size) + return AVERROR_INVALIDDATA; + offs += asset->core_size; + size -= asset->core_size; + } + + if (asset->extension_mask & DCA_EXSS_XBR) { + asset->xbr_offset = offs; + if (asset->xbr_size > size) + return AVERROR_INVALIDDATA; + offs += asset->xbr_size; + size -= asset->xbr_size; + } + + if (asset->extension_mask & DCA_EXSS_XXCH) { + asset->xxch_offset = offs; + if (asset->xxch_size > size) + return AVERROR_INVALIDDATA; + offs += asset->xxch_size; + size -= asset->xxch_size; + } + + if (asset->extension_mask & DCA_EXSS_X96) { + asset->x96_offset = offs; + if (asset->x96_size > size) + return AVERROR_INVALIDDATA; + offs += asset->x96_size; + size -= asset->x96_size; + } + + if (asset->extension_mask & DCA_EXSS_LBR) { + asset->lbr_offset = offs; + if (asset->lbr_size > size) + return AVERROR_INVALIDDATA; + offs += asset->lbr_size; + size -= asset->lbr_size; + } + + if (asset->extension_mask & DCA_EXSS_XLL) { + asset->xll_offset = offs; + if (asset->xll_size > size) + return AVERROR_INVALIDDATA; + offs += asset->xll_size; + size -= asset->xll_size; + } + + return 0; +} + +int ff_dca_exss_parse(DCAExssParser *s, uint8_t *data, int size) +{ + int i, ret, offset, wide_hdr, header_size; + + if ((ret = init_get_bits8(&s->gb, data, size)) < 0) + return ret; + + // Extension substream sync word + skip_bits_long(&s->gb, 32); + + // User defined bits + skip_bits(&s->gb, 8); + + // Extension substream index + s->exss_index = get_bits(&s->gb, 2); + + // Flag indicating short or long header size + wide_hdr = get_bits1(&s->gb); + + // Extension substream header length + header_size = get_bits(&s->gb, 8 + 4 * wide_hdr) + 1; + + // Check CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, 32 + 8, header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid EXSS header checksum\n"); + return AVERROR_INVALIDDATA; + } + + s->exss_size_nbits = 16 + 4 * wide_hdr; + + // Number of bytes of extension substream + s->exss_size = get_bits(&s->gb, s->exss_size_nbits) + 1; + if (s->exss_size > size) { + av_log(s->avctx, AV_LOG_ERROR, "Packet too short for EXSS frame\n"); + return AVERROR_INVALIDDATA; + } + + // Per stream static fields presence flag + if (s->static_fields_present = get_bits1(&s->gb)) { + int active_exss_mask[8]; + + // Reference clock code + skip_bits(&s->gb, 2); + + // Extension substream frame duration + skip_bits(&s->gb, 3); + + // Timecode presence flag + if (get_bits1(&s->gb)) + // Timecode data + skip_bits_long(&s->gb, 36); + + // Number of defined audio presentations + s->npresents = get_bits(&s->gb, 3) + 1; + if (s->npresents > 1) { + avpriv_request_sample(s->avctx, "%d audio presentations", s->npresents); + return AVERROR_PATCHWELCOME; + } + + // Number of audio assets in extension substream + s->nassets = get_bits(&s->gb, 3) + 1; + if (s->nassets > 1) { + avpriv_request_sample(s->avctx, "%d audio assets", s->nassets); + return AVERROR_PATCHWELCOME; + } + + // Active extension substream mask for audio presentation + for (i = 0; i < s->npresents; i++) + active_exss_mask[i] = get_bits(&s->gb, s->exss_index + 1); + + // Active audio asset mask + for (i = 0; i < s->npresents; i++) + skip_bits_long(&s->gb, av_popcount(active_exss_mask[i]) * 8); + + // Mixing metadata enable flag + if (s->mix_metadata_enabled = get_bits1(&s->gb)) { + int spkr_mask_nbits; + + // Mixing metadata adjustment level + skip_bits(&s->gb, 2); + + // Number of bits for mixer output speaker activity mask + spkr_mask_nbits = (get_bits(&s->gb, 2) + 1) << 2; + + // Number of mixing configurations + s->nmixoutconfigs = get_bits(&s->gb, 2) + 1; + + // Speaker layout mask for mixer output channels + for (i = 0; i < s->nmixoutconfigs; i++) + s->nmixoutchs[i] = count_chs_for_mask(get_bits(&s->gb, spkr_mask_nbits)); + } + } else { + s->npresents = 1; + s->nassets = 1; + } + + // Size of encoded asset data in bytes + offset = header_size; + for (i = 0; i < s->nassets; i++) { + s->assets[i].asset_offset = offset; + s->assets[i].asset_size = get_bits(&s->gb, s->exss_size_nbits) + 1; + offset += s->assets[i].asset_size; + if (offset > s->exss_size) { + av_log(s->avctx, AV_LOG_ERROR, "EXSS asset out of bounds\n"); + return AVERROR_INVALIDDATA; + } + } + + // Audio asset descriptor + for (i = 0; i < s->nassets; i++) { + if ((ret = parse_descriptor(s, &s->assets[i])) < 0) + return ret; + if ((ret = set_exss_offsets(&s->assets[i])) < 0) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid extension size in EXSS asset descriptor\n"); + return ret; + } + } + + // Backward compatible core present + // Backward compatible core substream index + // Backward compatible core asset index + // Reserved + // Byte align + // CRC16 of extension substream header + if (ff_dca_seek_bits(&s->gb, header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of EXSS header\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} diff --git a/libavcodec/dca_exss.h b/libavcodec/dca_exss.h new file mode 100644 index 0000000..323063a --- /dev/null +++ b/libavcodec/dca_exss.h @@ -0,0 +1,92 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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_DCA_EXSS_H +#define AVCODEC_DCA_EXSS_H + +#include "libavutil/common.h" + +#include "avcodec.h" +#include "get_bits.h" + +typedef struct DCAExssAsset { + int asset_offset; ///< Offset to asset data from start of substream + int asset_size; ///< Size of encoded asset data + int asset_index; ///< Audio asset identifier + + int pcm_bit_res; ///< PCM bit resolution + int max_sample_rate; ///< Maximum sample rate + int nchannels_total; ///< Total number of channels + int one_to_one_map_ch_to_spkr; ///< One to one channel to speaker mapping flag + int embedded_stereo; ///< Embedded stereo flag + int embedded_6ch; ///< Embedded 6 channels flag + int spkr_mask_enabled; ///< Speaker mask enabled flag + int spkr_mask; ///< Loudspeaker activity mask + int representation_type; ///< Representation type + + int coding_mode; ///< Coding mode for the asset + int extension_mask; ///< Coding components used in asset + + int core_offset; ///< Offset to core component from start of substream + int core_size; ///< Size of core component in extension substream + + int xbr_offset; ///< Offset to XBR extension from start of substream + int xbr_size; ///< Size of XBR extension in extension substream + + int xxch_offset; ///< Offset to XXCH extension from start of substream + int xxch_size; ///< Size of XXCH extension in extension substream + + int x96_offset; ///< Offset to X96 extension from start of substream + int x96_size; ///< Size of X96 extension in extension substream + + int lbr_offset; ///< Offset to LBR component from start of substream + int lbr_size; ///< Size of LBR component in extension substream + + int xll_offset; ///< Offset to XLL data from start of substream + int xll_size; ///< Size of XLL data in extension substream + int xll_sync_present; ///< XLL sync word present flag + int xll_delay_nframes; ///< Initial XLL decoding delay in frames + int xll_sync_offset; ///< Number of bytes offset to XLL sync + + int hd_stream_id; ///< DTS-HD stream ID +} DCAExssAsset; + +typedef struct DCAExssParser { + AVCodecContext *avctx; + GetBitContext gb; + + int exss_index; ///< Extension substream index + int exss_size_nbits; ///< Number of bits for extension substream size + int exss_size; ///< Number of bytes of extension substream + + int static_fields_present; ///< Per stream static fields presence flag + int npresents; ///< Number of defined audio presentations + int nassets; ///< Number of audio assets in extension substream + + int mix_metadata_enabled; ///< Mixing metadata enable flag + int nmixoutconfigs; ///< Number of mixing configurations + int nmixoutchs[4]; ///< Speaker layout mask for mixer output channels + + DCAExssAsset assets[1]; ///< Audio asset descriptors +} DCAExssParser; + +int ff_dca_exss_parse(DCAExssParser *s, uint8_t *data, int size); + +#endif diff --git a/libavcodec/dca_xll.c b/libavcodec/dca_xll.c new file mode 100644 index 0000000..cd1af81 --- /dev/null +++ b/libavcodec/dca_xll.c @@ -0,0 +1,1499 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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 "dcadec.h" +#include "dcadata.h" +#include "dcamath.h" +#include "dca_syncwords.h" +#include "unary.h" + +static int get_linear(GetBitContext *gb, int n) +{ + unsigned int v = get_bits_long(gb, n); + return (v >> 1) ^ -(v & 1); +} + +static int get_rice_un(GetBitContext *gb, int k) +{ + unsigned int v = get_unary(gb, 1, 128); + return (v << k) | get_bits_long(gb, k); +} + +static int get_rice(GetBitContext *gb, int k) +{ + unsigned int v = get_rice_un(gb, k); + return (v >> 1) ^ -(v & 1); +} + +static void get_array(GetBitContext *gb, int32_t *array, int size, int n) +{ + int i; + + for (i = 0; i < size; i++) + array[i] = get_bits(gb, n); +} + +static void get_linear_array(GetBitContext *gb, int32_t *array, int size, int n) +{ + int i; + + if (n == 0) + memset(array, 0, sizeof(*array) * size); + else for (i = 0; i < size; i++) + array[i] = get_linear(gb, n); +} + +static void get_rice_array(GetBitContext *gb, int32_t *array, int size, int k) +{ + int i; + + for (i = 0; i < size; i++) + array[i] = get_rice(gb, k); +} + +static int parse_dmix_coeffs(DCAXllDecoder *s, DCAXllChSet *c) +{ + // Size of downmix coefficient matrix + int m = c->primary_chset ? ff_dca_dmix_primary_nch[c->dmix_type] : c->hier_ofs; + int i, j, *coeff_ptr = c->dmix_coeff; + + for (i = 0; i < m; i++) { + int code, sign, coeff, scale, scale_inv = 0; + unsigned int index; + + // Downmix scale (only for non-primary channel sets) + if (!c->primary_chset) { + code = get_bits(&s->gb, 9); + sign = (code >> 8) - 1; + index = (code & 0xff) - FF_DCA_DMIXTABLE_OFFSET; + if (index >= FF_DCA_INV_DMIXTABLE_SIZE) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL downmix scale index\n"); + return AVERROR_INVALIDDATA; + } + scale = ff_dca_dmixtable[index + FF_DCA_DMIXTABLE_OFFSET]; + scale_inv = ff_dca_inv_dmixtable[index]; + c->dmix_scale[i] = (scale ^ sign) - sign; + c->dmix_scale_inv[i] = (scale_inv ^ sign) - sign; + } + + // Downmix coefficients + for (j = 0; j < c->nchannels; j++) { + code = get_bits(&s->gb, 9); + sign = (code >> 8) - 1; + index = code & 0xff; + if (index >= FF_DCA_DMIXTABLE_SIZE) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL downmix coefficient index\n"); + return AVERROR_INVALIDDATA; + } + coeff = ff_dca_dmixtable[index]; + if (!c->primary_chset) + // Multiply by |InvDmixScale| to get |UndoDmixScale| + coeff = mul16(scale_inv, coeff); + *coeff_ptr++ = (coeff ^ sign) - sign; + } + } + + return 0; +} + +static int chs_parse_header(DCAXllDecoder *s, DCAXllChSet *c, DCAExssAsset *asset) +{ + int i, j, k, ret, band, header_size, header_pos = get_bits_count(&s->gb); + DCAXllChSet *p = &s->chset[0]; + DCAXllBand *b; + + // Size of channel set sub-header + header_size = get_bits(&s->gb, 10) + 1; + + // Check CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, header_pos, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL sub-header checksum\n"); + return AVERROR_INVALIDDATA; + } + + // Number of channels in the channel set + c->nchannels = get_bits(&s->gb, 4) + 1; + if (c->nchannels > DCA_XLL_CHANNELS_MAX) { + avpriv_request_sample(s->avctx, "%d XLL channels", c->nchannels); + return AVERROR_PATCHWELCOME; + } + + // Residual type + c->residual_encode = get_bits(&s->gb, c->nchannels); + + // PCM bit resolution + c->pcm_bit_res = get_bits(&s->gb, 5) + 1; + + // Storage unit width + c->storage_bit_res = get_bits(&s->gb, 5) + 1; + if (c->storage_bit_res != 16 && c->storage_bit_res != 24) { + avpriv_request_sample(s->avctx, "%d-bit XLL storage resolution", c->storage_bit_res); + return AVERROR_PATCHWELCOME; + } + + if (c->pcm_bit_res > c->storage_bit_res) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid PCM bit resolution for XLL channel set (%d > %d)\n", c->pcm_bit_res, c->storage_bit_res); + return AVERROR_INVALIDDATA; + } + + // Original sampling frequency + c->freq = ff_dca_sampling_freqs[get_bits(&s->gb, 4)]; + if (c->freq > 192000) { + avpriv_request_sample(s->avctx, "%d Hz XLL sampling frequency", c->freq); + return AVERROR_PATCHWELCOME; + } + + // Sampling frequency modifier + if (get_bits(&s->gb, 2)) { + avpriv_request_sample(s->avctx, "XLL sampling frequency modifier"); + return AVERROR_PATCHWELCOME; + } + + // Which replacement set this channel set is member of + if (get_bits(&s->gb, 2)) { + avpriv_request_sample(s->avctx, "XLL replacement set"); + return AVERROR_PATCHWELCOME; + } + + if (asset->one_to_one_map_ch_to_spkr) { + // Primary channel set flag + c->primary_chset = get_bits1(&s->gb); + if (c->primary_chset != (c == p)) { + av_log(s->avctx, AV_LOG_ERROR, "The first (and only) XLL channel set must be primary\n"); + return AVERROR_INVALIDDATA; + } + + // Downmix coefficients present in stream + c->dmix_coeffs_present = get_bits1(&s->gb); + + // Downmix already performed by encoder + c->dmix_embedded = c->dmix_coeffs_present && get_bits1(&s->gb); + + // Downmix type + if (c->dmix_coeffs_present && c->primary_chset) { + c->dmix_type = get_bits(&s->gb, 3); + if (c->dmix_type >= DCA_DMIX_TYPE_COUNT) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL primary channel set downmix type\n"); + return AVERROR_INVALIDDATA; + } + } + + // Whether the channel set is part of a hierarchy + c->hier_chset = get_bits1(&s->gb); + if (!c->hier_chset && s->nchsets != 1) { + avpriv_request_sample(s->avctx, "XLL channel set outside of hierarchy"); + return AVERROR_PATCHWELCOME; + } + + // Downmix coefficients + if (c->dmix_coeffs_present && (ret = parse_dmix_coeffs(s, c)) < 0) + return ret; + + // Channel mask enabled + if (!get_bits1(&s->gb)) { + avpriv_request_sample(s->avctx, "Disabled XLL channel mask"); + return AVERROR_PATCHWELCOME; + } + + // Channel mask for set + c->ch_mask = get_bits_long(&s->gb, s->ch_mask_nbits); + if (av_popcount(c->ch_mask) != c->nchannels) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL channel mask\n"); + return AVERROR_INVALIDDATA; + } + + // Build the channel to speaker map + for (i = 0, j = 0; i < s->ch_mask_nbits; i++) + if (c->ch_mask & (1U << i)) + c->ch_remap[j++] = i; + } else { + // Mapping coeffs present flag + if (c->nchannels != 2 || s->nchsets != 1 || get_bits1(&s->gb)) { + avpriv_request_sample(s->avctx, "Custom XLL channel to speaker mapping"); + return AVERROR_PATCHWELCOME; + } + + // Setup for LtRt decoding + c->primary_chset = 1; + c->dmix_coeffs_present = 0; + c->dmix_embedded = 0; + c->hier_chset = 0; + c->ch_mask = DCA_SPEAKER_LAYOUT_STEREO; + c->ch_remap[0] = DCA_SPEAKER_L; + c->ch_remap[1] = DCA_SPEAKER_R; + } + + if (c->freq > 96000) { + // Extra frequency bands flag + if (get_bits1(&s->gb)) { + avpriv_request_sample(s->avctx, "Extra XLL frequency bands"); + return AVERROR_PATCHWELCOME; + } + c->nfreqbands = 2; + } else { + c->nfreqbands = 1; + } + + // Set the sampling frequency to that of the first frequency band. + // Frequency will be doubled again after bands assembly. + c->freq >>= c->nfreqbands - 1; + + // Verify that all channel sets have the same audio characteristics + if (c != p && (c->nfreqbands != p->nfreqbands || c->freq != p->freq + || c->pcm_bit_res != p->pcm_bit_res + || c->storage_bit_res != p->storage_bit_res)) { + avpriv_request_sample(s->avctx, "Different XLL audio characteristics"); + return AVERROR_PATCHWELCOME; + } + + // Determine number of bits to read bit allocation coding parameter + if (c->storage_bit_res > 16) + c->nabits = 5; + else if (c->storage_bit_res > 8) + c->nabits = 4; + else + c->nabits = 3; + + // Account for embedded downmix and decimator saturation + if ((s->nchsets > 1 || c->nfreqbands > 1) && c->nabits < 5) + c->nabits++; + + for (band = 0, b = c->bands; band < c->nfreqbands; band++, b++) { + // Pairwise channel decorrelation + if ((b->decor_enabled = get_bits1(&s->gb)) && c->nchannels > 1) { + int ch_nbits = av_ceil_log2(c->nchannels); + + // Original channel order + for (i = 0; i < c->nchannels; i++) { + b->orig_order[i] = get_bits(&s->gb, ch_nbits); + if (b->orig_order[i] >= c->nchannels) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL original channel order\n"); + return AVERROR_INVALIDDATA; + } + } + + // Pairwise channel coefficients + for (i = 0; i < c->nchannels / 2; i++) + b->decor_coeff[i] = get_bits1(&s->gb) ? get_linear(&s->gb, 7) : 0; + } else { + for (i = 0; i < c->nchannels; i++) + b->orig_order[i] = i; + for (i = 0; i < c->nchannels / 2; i++) + b->decor_coeff[i] = 0; + } + + // Adaptive predictor order + b->highest_pred_order = 0; + for (i = 0; i < c->nchannels; i++) { + b->adapt_pred_order[i] = get_bits(&s->gb, 4); + if (b->adapt_pred_order[i] > b->highest_pred_order) + b->highest_pred_order = b->adapt_pred_order[i]; + } + if (b->highest_pred_order > s->nsegsamples) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL adaptive predicition order\n"); + return AVERROR_INVALIDDATA; + } + + // Fixed predictor order + for (i = 0; i < c->nchannels; i++) + b->fixed_pred_order[i] = b->adapt_pred_order[i] ? 0 : get_bits(&s->gb, 2); + + // Adaptive predictor quantized reflection coefficients + for (i = 0; i < c->nchannels; i++) { + for (j = 0; j < b->adapt_pred_order[i]; j++) { + k = get_linear(&s->gb, 8); + if (k == -128) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL reflection coefficient index\n"); + return AVERROR_INVALIDDATA; + } + if (k < 0) + b->adapt_refl_coeff[i][j] = -(int)ff_dca_xll_refl_coeff[-k]; + else + b->adapt_refl_coeff[i][j] = (int)ff_dca_xll_refl_coeff[ k]; + } + } + + // Downmix performed by encoder in extension frequency band + b->dmix_embedded = c->dmix_embedded && (band == 0 || get_bits1(&s->gb)); + + // MSB/LSB split flag in extension frequency band + if ((band == 0 && s->scalable_lsbs) || (band != 0 && get_bits1(&s->gb))) { + // Size of LSB section in any segment + b->lsb_section_size = get_bits_long(&s->gb, s->seg_size_nbits); + if (b->lsb_section_size < 0 || b->lsb_section_size > s->frame_size) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid LSB section size\n"); + return AVERROR_INVALIDDATA; + } + + // Account for optional CRC bytes after LSB section + if (b->lsb_section_size && (s->band_crc_present > 2 || + (band == 0 && s->band_crc_present > 1))) + b->lsb_section_size += 2; + + // Number of bits to represent the samples in LSB part + for (i = 0; i < c->nchannels; i++) { + b->nscalablelsbs[i] = get_bits(&s->gb, 4); + if (b->nscalablelsbs[i] && !b->lsb_section_size) { + av_log(s->avctx, AV_LOG_ERROR, "LSB section missing with non-zero LSB width\n"); + return AVERROR_INVALIDDATA; + } + } + } else { + b->lsb_section_size = 0; + for (i = 0; i < c->nchannels; i++) + b->nscalablelsbs[i] = 0; + } + + // Scalable resolution flag in extension frequency band + if ((band == 0 && s->scalable_lsbs) || (band != 0 && get_bits1(&s->gb))) { + // Number of bits discarded by authoring + for (i = 0; i < c->nchannels; i++) + b->bit_width_adjust[i] = get_bits(&s->gb, 4); + } else { + for (i = 0; i < c->nchannels; i++) + b->bit_width_adjust[i] = 0; + } + } + + // Reserved + // Byte align + // CRC16 of channel set sub-header + if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL sub-header\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int chs_alloc_msb_band_data(DCAXllDecoder *s, DCAXllChSet *c) +{ + int ndecisamples = c->nfreqbands > 1 ? DCA_XLL_DECI_HISTORY_MAX : 0; + int nchsamples = s->nframesamples + ndecisamples; + int i, j, nsamples = nchsamples * c->nchannels * c->nfreqbands; + int32_t *ptr; + + // Reallocate MSB sample buffer + av_fast_malloc(&c->sample_buffer[0], &c->sample_size[0], nsamples * sizeof(int32_t)); + if (!c->sample_buffer[0]) + return AVERROR(ENOMEM); + + ptr = c->sample_buffer[0] + ndecisamples; + for (i = 0; i < c->nfreqbands; i++) { + for (j = 0; j < c->nchannels; j++) { + c->bands[i].msb_sample_buffer[j] = ptr; + ptr += nchsamples; + } + } + + return 0; +} + +static int chs_alloc_lsb_band_data(DCAXllDecoder *s, DCAXllChSet *c) +{ + int i, j, nsamples = 0; + int32_t *ptr; + + // Determine number of frequency bands that have MSB/LSB split + for (i = 0; i < c->nfreqbands; i++) + if (c->bands[i].lsb_section_size) + nsamples += s->nframesamples * c->nchannels; + if (!nsamples) + return 0; + + // Reallocate LSB sample buffer + av_fast_malloc(&c->sample_buffer[1], &c->sample_size[1], nsamples * sizeof(int32_t)); + if (!c->sample_buffer[1]) + return AVERROR(ENOMEM); + + ptr = c->sample_buffer[1]; + for (i = 0; i < c->nfreqbands; i++) { + if (c->bands[i].lsb_section_size) { + for (j = 0; j < c->nchannels; j++) { + c->bands[i].lsb_sample_buffer[j] = ptr; + ptr += s->nframesamples; + } + } else { + for (j = 0; j < c->nchannels; j++) + c->bands[i].lsb_sample_buffer[j] = NULL; + } + } + + return 0; +} + +static int chs_parse_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band, int seg, int band_data_end) +{ + DCAXllBand *b = &c->bands[band]; + int i, j, k; + + // Start unpacking MSB portion of the segment + if (!(seg && get_bits1(&s->gb))) { + // Unpack segment type + // 0 - distinct coding parameters for each channel + // 1 - common coding parameters for all channels + c->seg_common = get_bits1(&s->gb); + + // Determine number of coding parameters encoded in segment + k = c->seg_common ? 1 : c->nchannels; + + // Unpack Rice coding parameters + for (i = 0; i < k; i++) { + // Unpack Rice coding flag + // 0 - linear code, 1 - Rice code + c->rice_code_flag[i] = get_bits1(&s->gb); + if (!c->seg_common && c->rice_code_flag[i]) { + // Unpack Hybrid Rice coding flag + // 0 - Rice code, 1 - Hybrid Rice code + if (get_bits1(&s->gb)) + // Unpack binary code length for isolated samples + c->bitalloc_hybrid_linear[i] = get_bits(&s->gb, c->nabits) + 1; + else + // 0 indicates no Hybrid Rice coding + c->bitalloc_hybrid_linear[i] = 0; + } else { + // 0 indicates no Hybrid Rice coding + c->bitalloc_hybrid_linear[i] = 0; + } + } + + // Unpack coding parameters + for (i = 0; i < k; i++) { + if (seg == 0) { + // Unpack coding parameter for part A of segment 0 + c->bitalloc_part_a[i] = get_bits(&s->gb, c->nabits); + + // Adjust for the linear code + if (!c->rice_code_flag[i] && c->bitalloc_part_a[i]) + c->bitalloc_part_a[i]++; + + if (!c->seg_common) + c->nsamples_part_a[i] = b->adapt_pred_order[i]; + else + c->nsamples_part_a[i] = b->highest_pred_order; + } else { + c->bitalloc_part_a[i] = 0; + c->nsamples_part_a[i] = 0; + } + + // Unpack coding parameter for part B of segment + c->bitalloc_part_b[i] = get_bits(&s->gb, c->nabits); + + // Adjust for the linear code + if (!c->rice_code_flag[i] && c->bitalloc_part_b[i]) + c->bitalloc_part_b[i]++; + } + } + + // Unpack entropy codes + for (i = 0; i < c->nchannels; i++) { + int32_t *part_a, *part_b; + int nsamples_part_b; + + // Select index of coding parameters + k = c->seg_common ? 0 : i; + + // Slice the segment into parts A and B + part_a = b->msb_sample_buffer[i] + seg * s->nsegsamples; + part_b = part_a + c->nsamples_part_a[k]; + nsamples_part_b = s->nsegsamples - c->nsamples_part_a[k]; + + if (get_bits_left(&s->gb) < 0) + return AVERROR_INVALIDDATA; + + if (!c->rice_code_flag[k]) { + // Linear codes + // Unpack all residuals of part A of segment 0 + get_linear_array(&s->gb, part_a, c->nsamples_part_a[k], + c->bitalloc_part_a[k]); + + // Unpack all residuals of part B of segment 0 and others + get_linear_array(&s->gb, part_b, nsamples_part_b, + c->bitalloc_part_b[k]); + } else { + // Rice codes + // Unpack all residuals of part A of segment 0 + get_rice_array(&s->gb, part_a, c->nsamples_part_a[k], + c->bitalloc_part_a[k]); + + if (c->bitalloc_hybrid_linear[k]) { + // Hybrid Rice codes + // Unpack the number of isolated samples + int nisosamples = get_bits(&s->gb, s->nsegsamples_log2); + + // Set all locations to 0 + memset(part_b, 0, sizeof(*part_b) * nsamples_part_b); + + // Extract the locations of isolated samples and flag by -1 + for (j = 0; j < nisosamples; j++) { + int loc = get_bits(&s->gb, s->nsegsamples_log2); + if (loc >= nsamples_part_b) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid isolated sample location\n"); + return AVERROR_INVALIDDATA; + } + part_b[loc] = -1; + } + + // Unpack all residuals of part B of segment 0 and others + for (j = 0; j < nsamples_part_b; j++) { + if (part_b[j]) + part_b[j] = get_linear(&s->gb, c->bitalloc_hybrid_linear[k]); + else + part_b[j] = get_rice(&s->gb, c->bitalloc_part_b[k]); + } + } else { + // Rice codes + // Unpack all residuals of part B of segment 0 and others + get_rice_array(&s->gb, part_b, nsamples_part_b, c->bitalloc_part_b[k]); + } + } + } + + // Unpack decimator history for frequency band 1 + if (seg == 0 && band == 1) { + int nbits = get_bits(&s->gb, 5) + 1; + for (i = 0; i < c->nchannels; i++) + for (j = 1; j < DCA_XLL_DECI_HISTORY_MAX; j++) + c->deci_history[i][j] = get_sbits_long(&s->gb, nbits); + } + + // Start unpacking LSB portion of the segment + if (b->lsb_section_size) { + // Skip to the start of LSB portion + if (ff_dca_seek_bits(&s->gb, band_data_end - b->lsb_section_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL band data\n"); + return AVERROR_INVALIDDATA; + } + + // Unpack all LSB parts of residuals of this segment + for (i = 0; i < c->nchannels; i++) { + if (b->nscalablelsbs[i]) { + get_array(&s->gb, + b->lsb_sample_buffer[i] + seg * s->nsegsamples, + s->nsegsamples, b->nscalablelsbs[i]); + } + } + } + + // Skip to the end of band data + if (ff_dca_seek_bits(&s->gb, band_data_end)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL band data\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static void av_cold chs_clear_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band, int seg) +{ + DCAXllBand *b = &c->bands[band]; + int i, offset, nsamples; + + if (seg < 0) { + offset = 0; + nsamples = s->nframesamples; + } else { + offset = seg * s->nsegsamples; + nsamples = s->nsegsamples; + } + + for (i = 0; i < c->nchannels; i++) { + memset(b->msb_sample_buffer[i] + offset, 0, nsamples * sizeof(int32_t)); + if (b->lsb_section_size) + memset(b->lsb_sample_buffer[i] + offset, 0, nsamples * sizeof(int32_t)); + } + + if (seg <= 0 && band) + memset(c->deci_history, 0, sizeof(c->deci_history)); + + if (seg < 0) { + memset(b->nscalablelsbs, 0, sizeof(b->nscalablelsbs)); + memset(b->bit_width_adjust, 0, sizeof(b->bit_width_adjust)); + } +} + +static void chs_filter_band_data(DCAXllDecoder *s, DCAXllChSet *c, int band) +{ + DCAXllBand *b = &c->bands[band]; + int nsamples = s->nframesamples; + int i, j, k; + + // Inverse adaptive or fixed prediction + for (i = 0; i < c->nchannels; i++) { + int32_t *buf = b->msb_sample_buffer[i]; + int order = b->adapt_pred_order[i]; + if (order > 0) { + int coeff[DCA_XLL_ADAPT_PRED_ORDER_MAX]; + // Conversion from reflection coefficients to direct form coefficients + for (j = 0; j < order; j++) { + int rc = b->adapt_refl_coeff[i][j]; + for (k = 0; k < (j + 1) / 2; k++) { + int tmp1 = coeff[ k ]; + int tmp2 = coeff[j - k - 1]; + coeff[ k ] = tmp1 + mul16(rc, tmp2); + coeff[j - k - 1] = tmp2 + mul16(rc, tmp1); + } + coeff[j] = rc; + } + // Inverse adaptive prediction + for (j = 0; j < nsamples - order; j++) { + int64_t err = 0; + for (k = 0; k < order; k++) + err += (int64_t)buf[j + k] * coeff[order - k - 1]; + buf[j + k] -= clip23(norm16(err)); + } + } else { + // Inverse fixed coefficient prediction + for (j = 0; j < b->fixed_pred_order[i]; j++) + for (k = 1; k < nsamples; k++) + buf[k] += buf[k - 1]; + } + } + + // Inverse pairwise channel decorrellation + if (b->decor_enabled) { + int32_t *tmp[DCA_XLL_CHANNELS_MAX]; + + for (i = 0; i < c->nchannels / 2; i++) { + int coeff = b->decor_coeff[i]; + if (coeff) { + s->dcadsp->decor(b->msb_sample_buffer[i * 2 + 1], + b->msb_sample_buffer[i * 2 ], + coeff, nsamples); + } + } + + // Reorder channel pointers to the original order + for (i = 0; i < c->nchannels; i++) + tmp[i] = b->msb_sample_buffer[i]; + + for (i = 0; i < c->nchannels; i++) + b->msb_sample_buffer[b->orig_order[i]] = tmp[i]; + } + + // Map output channel pointers for frequency band 0 + if (c->nfreqbands == 1) + for (i = 0; i < c->nchannels; i++) + s->output_samples[c->ch_remap[i]] = b->msb_sample_buffer[i]; +} + +static int chs_get_lsb_width(DCAXllDecoder *s, DCAXllChSet *c, int band, int ch) +{ + int adj = c->bands[band].bit_width_adjust[ch]; + int shift = c->bands[band].nscalablelsbs[ch]; + + if (s->fixed_lsb_width) + shift = s->fixed_lsb_width; + else if (shift && adj) + shift += adj - 1; + else + shift += adj; + + return shift; +} + +static void chs_assemble_msbs_lsbs(DCAXllDecoder *s, DCAXllChSet *c, int band) +{ + DCAXllBand *b = &c->bands[band]; + int n, ch, nsamples = s->nframesamples; + + for (ch = 0; ch < c->nchannels; ch++) { + int shift = chs_get_lsb_width(s, c, band, ch); + if (shift) { + int32_t *msb = b->msb_sample_buffer[ch]; + if (b->nscalablelsbs[ch]) { + int32_t *lsb = b->lsb_sample_buffer[ch]; + int adj = b->bit_width_adjust[ch]; + for (n = 0; n < nsamples; n++) + msb[n] = msb[n] * (1 << shift) + (lsb[n] << adj); + } else { + for (n = 0; n < nsamples; n++) + msb[n] = msb[n] * (1 << shift); + } + } + } +} + +static int chs_assemble_freq_bands(DCAXllDecoder *s, DCAXllChSet *c) +{ + int ch, nsamples = s->nframesamples; + int32_t *ptr; + + av_assert1(c->nfreqbands > 1); + + // Reallocate frequency band assembly buffer + av_fast_malloc(&c->sample_buffer[2], &c->sample_size[2], + 2 * nsamples * c->nchannels * sizeof(int32_t)); + if (!c->sample_buffer[2]) + return AVERROR(ENOMEM); + + // Assemble frequency bands 0 and 1 + ptr = c->sample_buffer[2]; + for (ch = 0; ch < c->nchannels; ch++) { + int32_t *band0 = c->bands[0].msb_sample_buffer[ch]; + int32_t *band1 = c->bands[1].msb_sample_buffer[ch]; + + // Copy decimator history + memcpy(band0 - DCA_XLL_DECI_HISTORY_MAX, + c->deci_history[ch], sizeof(c->deci_history[0])); + + // Filter + s->dcadsp->assemble_freq_bands(ptr, band0, band1, + ff_dca_xll_band_coeff, + nsamples); + + // Remap output channel pointer to assembly buffer + s->output_samples[c->ch_remap[ch]] = ptr; + ptr += nsamples * 2; + } + + return 0; +} + +static int parse_common_header(DCAXllDecoder *s) +{ + int stream_ver, header_size, frame_size_nbits, nframesegs_log2; + + // XLL extension sync word + if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XLL) { + av_log(s->avctx, AV_LOG_VERBOSE, "Invalid XLL sync word\n"); + return AVERROR(EAGAIN); + } + + // Version number + stream_ver = get_bits(&s->gb, 4) + 1; + if (stream_ver > 1) { + avpriv_request_sample(s->avctx, "XLL stream version %d", stream_ver); + return AVERROR_PATCHWELCOME; + } + + // Lossless frame header length + header_size = get_bits(&s->gb, 8) + 1; + + // Check CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, 32, header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL common header checksum\n"); + return AVERROR_INVALIDDATA; + } + + // Number of bits used to read frame size + frame_size_nbits = get_bits(&s->gb, 5) + 1; + + // Number of bytes in a lossless frame + s->frame_size = get_bits_long(&s->gb, frame_size_nbits); + if (s->frame_size < 0 || s->frame_size >= DCA_XLL_PBR_BUFFER_MAX) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid XLL frame size (%d bytes)\n", s->frame_size); + return AVERROR_INVALIDDATA; + } + s->frame_size++; + + // Number of channels sets per frame + s->nchsets = get_bits(&s->gb, 4) + 1; + if (s->nchsets > DCA_XLL_CHSETS_MAX) { + avpriv_request_sample(s->avctx, "%d XLL channel sets", s->nchsets); + return AVERROR_PATCHWELCOME; + } + + // Number of segments per frame + nframesegs_log2 = get_bits(&s->gb, 4); + s->nframesegs = 1 << nframesegs_log2; + if (s->nframesegs > 1024) { + av_log(s->avctx, AV_LOG_ERROR, "Too many segments per XLL frame\n"); + return AVERROR_INVALIDDATA; + } + + // Samples in segment per one frequency band for the first channel set + // Maximum value is 256 for sampling frequencies <= 48 kHz + // Maximum value is 512 for sampling frequencies > 48 kHz + s->nsegsamples_log2 = get_bits(&s->gb, 4); + if (!s->nsegsamples_log2) { + av_log(s->avctx, AV_LOG_ERROR, "Too few samples per XLL segment\n"); + return AVERROR_INVALIDDATA; + } + s->nsegsamples = 1 << s->nsegsamples_log2; + if (s->nsegsamples > 512) { + av_log(s->avctx, AV_LOG_ERROR, "Too many samples per XLL segment\n"); + return AVERROR_INVALIDDATA; + } + + // Samples in frame per one frequency band for the first channel set + s->nframesamples_log2 = s->nsegsamples_log2 + nframesegs_log2; + s->nframesamples = 1 << s->nframesamples_log2; + if (s->nframesamples > 65536) { + av_log(s->avctx, AV_LOG_ERROR, "Too many samples per XLL frame\n"); + return AVERROR_INVALIDDATA; + } + + // Number of bits used to read segment size + s->seg_size_nbits = get_bits(&s->gb, 5) + 1; + + // Presence of CRC16 within each frequency band + // 0 - No CRC16 within band + // 1 - CRC16 placed at the end of MSB0 + // 2 - CRC16 placed at the end of MSB0 and LSB0 + // 3 - CRC16 placed at the end of MSB0 and LSB0 and other frequency bands + s->band_crc_present = get_bits(&s->gb, 2); + + // MSB/LSB split flag + s->scalable_lsbs = get_bits1(&s->gb); + + // Channel position mask + s->ch_mask_nbits = get_bits(&s->gb, 5) + 1; + + // Fixed LSB width + if (s->scalable_lsbs) + s->fixed_lsb_width = get_bits(&s->gb, 4); + else + s->fixed_lsb_width = 0; + + // Reserved + // Byte align + // Header CRC16 protection + if (ff_dca_seek_bits(&s->gb, header_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL common header\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int is_hier_dmix_chset(DCAXllChSet *c) +{ + return !c->primary_chset && c->dmix_embedded && c->hier_chset; +} + +static DCAXllChSet *find_next_hier_dmix_chset(DCAXllDecoder *s, DCAXllChSet *c) +{ + if (c->hier_chset) + while (++c < &s->chset[s->nchsets]) + if (is_hier_dmix_chset(c)) + return c; + + return NULL; +} + +static void prescale_down_mix(DCAXllChSet *c, DCAXllChSet *o) +{ + int i, j, *coeff_ptr = c->dmix_coeff; + + for (i = 0; i < c->hier_ofs; i++) { + int scale = o->dmix_scale[i]; + int scale_inv = o->dmix_scale_inv[i]; + c->dmix_scale[i] = mul15(c->dmix_scale[i], scale); + c->dmix_scale_inv[i] = mul16(c->dmix_scale_inv[i], scale_inv); + for (j = 0; j < c->nchannels; j++) { + int coeff = mul16(*coeff_ptr, scale_inv); + *coeff_ptr++ = mul15(coeff, o->dmix_scale[c->hier_ofs + j]); + } + } +} + +static int parse_sub_headers(DCAXllDecoder *s, DCAExssAsset *asset) +{ + DCAContext *dca = s->avctx->priv_data; + DCAXllChSet *c; + int i, ret; + + // Parse channel set headers + s->nfreqbands = 0; + s->nchannels = 0; + s->nreschsets = 0; + for (i = 0, c = s->chset; i < s->nchsets; i++, c++) { + c->hier_ofs = s->nchannels; + if ((ret = chs_parse_header(s, c, asset)) < 0) + return ret; + if (c->nfreqbands > s->nfreqbands) + s->nfreqbands = c->nfreqbands; + if (c->hier_chset) + s->nchannels += c->nchannels; + if (c->residual_encode != (1 << c->nchannels) - 1) + s->nreschsets++; + } + + // Pre-scale downmixing coefficients for all non-primary channel sets + for (i = s->nchsets - 1, c = &s->chset[i]; i > 0; i--, c--) { + if (is_hier_dmix_chset(c)) { + DCAXllChSet *o = find_next_hier_dmix_chset(s, c); + if (o) + prescale_down_mix(c, o); + } + } + + // Determine number of active channel sets to decode + switch (dca->request_channel_layout) { + case DCA_SPEAKER_LAYOUT_STEREO: + s->nactivechsets = 1; + break; + case DCA_SPEAKER_LAYOUT_5POINT0: + case DCA_SPEAKER_LAYOUT_5POINT1: + s->nactivechsets = (s->chset[0].nchannels < 5 && s->nchsets > 1) ? 2 : 1; + break; + default: + s->nactivechsets = s->nchsets; + break; + } + + return 0; +} + +static int parse_navi_table(DCAXllDecoder *s) +{ + int chs, seg, band, navi_nb, navi_pos, *navi_ptr; + DCAXllChSet *c; + + // Determine size of NAVI table + navi_nb = s->nfreqbands * s->nframesegs * s->nchsets; + if (navi_nb > 1024) { + av_log(s->avctx, AV_LOG_ERROR, "Too many NAVI entries (%d)\n", navi_nb); + return AVERROR_INVALIDDATA; + } + + // Reallocate NAVI table + av_fast_malloc(&s->navi, &s->navi_size, navi_nb * sizeof(*s->navi)); + if (!s->navi) + return AVERROR(ENOMEM); + + // Parse NAVI + navi_pos = get_bits_count(&s->gb); + navi_ptr = s->navi; + for (band = 0; band < s->nfreqbands; band++) { + for (seg = 0; seg < s->nframesegs; seg++) { + for (chs = 0, c = s->chset; chs < s->nchsets; chs++, c++) { + int size = 0; + if (c->nfreqbands > band) { + size = get_bits_long(&s->gb, s->seg_size_nbits); + if (size < 0 || size >= s->frame_size) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid NAVI segment size (%d bytes)\n", size); + return AVERROR_INVALIDDATA; + } + size++; + } + *navi_ptr++ = size; + } + } + } + + // Byte align + // CRC16 + skip_bits(&s->gb, -get_bits_count(&s->gb) & 7); + skip_bits(&s->gb, 16); + + // Check CRC + if ((s->avctx->err_recognition & (AV_EF_CRCCHECK | AV_EF_CAREFUL)) + && ff_dca_check_crc(&s->gb, navi_pos, get_bits_count(&s->gb))) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid NAVI checksum\n"); + return AVERROR_INVALIDDATA; + } + + return 0; +} + +static int parse_band_data(DCAXllDecoder *s) +{ + int ret, chs, seg, band, navi_pos, *navi_ptr; + DCAXllChSet *c; + + for (chs = 0, c = s->chset; chs < s->nactivechsets; chs++, c++) { + if ((ret = chs_alloc_msb_band_data(s, c)) < 0) + return ret; + if ((ret = chs_alloc_lsb_band_data(s, c)) < 0) + return ret; + } + + navi_pos = get_bits_count(&s->gb); + navi_ptr = s->navi; + for (band = 0; band < s->nfreqbands; band++) { + for (seg = 0; seg < s->nframesegs; seg++) { + for (chs = 0, c = s->chset; chs < s->nchsets; chs++, c++) { + if (c->nfreqbands > band) { + navi_pos += *navi_ptr * 8; + if (navi_pos > s->gb.size_in_bits) { + av_log(s->avctx, AV_LOG_ERROR, "Invalid NAVI position\n"); + return AVERROR_INVALIDDATA; + } + if (chs < s->nactivechsets && + (ret = chs_parse_band_data(s, c, band, seg, navi_pos)) < 0) { + if (s->avctx->err_recognition & AV_EF_EXPLODE) + return ret; + chs_clear_band_data(s, c, band, seg); + } + s->gb.index = navi_pos; + } + navi_ptr++; + } + } + } + + return 0; +} + +static int parse_frame(DCAXllDecoder *s, uint8_t *data, int size, DCAExssAsset *asset) +{ + int ret; + + if ((ret = init_get_bits8(&s->gb, data, size)) < 0) + return ret; + if ((ret = parse_common_header(s)) < 0) + return ret; + if ((ret = parse_sub_headers(s, asset)) < 0) + return ret; + if ((ret = parse_navi_table(s)) < 0) + return ret; + if ((ret = parse_band_data(s)) < 0) + return ret; + if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) { + av_log(s->avctx, AV_LOG_ERROR, "Read past end of XLL frame\n"); + return AVERROR_INVALIDDATA; + } + return ret; +} + +static void clear_pbr(DCAXllDecoder *s) +{ + s->pbr_length = 0; + s->pbr_delay = 0; +} + +static int copy_to_pbr(DCAXllDecoder *s, uint8_t *data, int size, int delay) +{ + if (size > DCA_XLL_PBR_BUFFER_MAX) + return AVERROR(ENOSPC); + + if (!s->pbr_buffer && !(s->pbr_buffer = av_malloc(DCA_XLL_PBR_BUFFER_MAX + DCA_BUFFER_PADDING_SIZE))) + return AVERROR(ENOMEM); + + memcpy(s->pbr_buffer, data, size); + s->pbr_length = size; + s->pbr_delay = delay; + return 0; +} + +static int parse_frame_no_pbr(DCAXllDecoder *s, uint8_t *data, int size, DCAExssAsset *asset) +{ + int ret = parse_frame(s, data, size, asset); + + // If XLL packet data didn't start with a sync word, we must have jumped + // right into the middle of PBR smoothing period + if (ret == AVERROR(EAGAIN) && asset->xll_sync_present && asset->xll_sync_offset < size) { + // Skip to the next sync word in this packet + data += asset->xll_sync_offset; + size -= asset->xll_sync_offset; + + // If decoding delay is set, put the frame into PBR buffer and return + // failure code. Higher level decoder is expected to switch to lossy + // core decoding or mute its output until decoding delay expires. + if (asset->xll_delay_nframes > 0) { + if ((ret = copy_to_pbr(s, data, size, asset->xll_delay_nframes)) < 0) + return ret; + return AVERROR(EAGAIN); + } + + // No decoding delay, just parse the frame in place + ret = parse_frame(s, data, size, asset); + } + + if (ret < 0) + return ret; + + if (s->frame_size > size) + return AVERROR(EINVAL); + + // If the XLL decoder didn't consume full packet, start PBR smoothing period + if (s->frame_size < size) + if ((ret = copy_to_pbr(s, data + s->frame_size, size - s->frame_size, 0)) < 0) + return ret; + + return 0; +} + +static int parse_frame_pbr(DCAXllDecoder *s, uint8_t *data, int size, DCAExssAsset *asset) +{ + int ret; + + if (size > DCA_XLL_PBR_BUFFER_MAX - s->pbr_length) { + ret = AVERROR(ENOSPC); + goto fail; + } + + memcpy(s->pbr_buffer + s->pbr_length, data, size); + s->pbr_length += size; + + // Respect decoding delay after synchronization error + if (s->pbr_delay > 0 && --s->pbr_delay) + return AVERROR(EAGAIN); + + if ((ret = parse_frame(s, s->pbr_buffer, s->pbr_length, asset)) < 0) + goto fail; + + if (s->frame_size > s->pbr_length) { + ret = AVERROR(EINVAL); + goto fail; + } + + if (s->frame_size == s->pbr_length) { + // End of PBR smoothing period + clear_pbr(s); + } else { + s->pbr_length -= s->frame_size; + memmove(s->pbr_buffer, s->pbr_buffer + s->frame_size, s->pbr_length); + } + + return 0; + +fail: + // For now, throw out all PBR state on failure. + // Perhaps we can be smarter and try to resync somehow. + clear_pbr(s); + return ret; +} + +int ff_dca_xll_parse(DCAXllDecoder *s, uint8_t *data, DCAExssAsset *asset) +{ + int ret; + + if (s->hd_stream_id != asset->hd_stream_id) { + clear_pbr(s); + s->hd_stream_id = asset->hd_stream_id; + } + + if (s->pbr_length) + ret = parse_frame_pbr(s, data + asset->xll_offset, asset->xll_size, asset); + else + ret = parse_frame_no_pbr(s, data + asset->xll_offset, asset->xll_size, asset); + + return ret; +} + +static void undo_down_mix(DCAXllDecoder *s, DCAXllChSet *o, int band) +{ + int i, j, k, nchannels = 0, *coeff_ptr = o->dmix_coeff; + DCAXllChSet *c; + + for (i = 0, c = s->chset; i < s->nactivechsets; i++, c++) { + if (!c->hier_chset) + continue; + + av_assert1(band < c->nfreqbands); + for (j = 0; j < c->nchannels; j++) { + for (k = 0; k < o->nchannels; k++) { + int coeff = *coeff_ptr++; + if (coeff) { + s->dcadsp->dmix_sub(c->bands[band].msb_sample_buffer[j], + o->bands[band].msb_sample_buffer[k], + coeff, s->nframesamples); + if (band) + s->dcadsp->dmix_sub(c->deci_history[j], + o->deci_history[k], + coeff, DCA_XLL_DECI_HISTORY_MAX); + } + } + } + + nchannels += c->nchannels; + if (nchannels >= o->hier_ofs) + break; + } +} + +static void scale_down_mix(DCAXllDecoder *s, DCAXllChSet *o, int band) +{ + int i, j, nchannels = 0; + DCAXllChSet *c; + + for (i = 0, c = s->chset; i < s->nactivechsets; i++, c++) { + if (!c->hier_chset) + continue; + + av_assert1(band < c->nfreqbands); + for (j = 0; j < c->nchannels; j++) { + int scale = o->dmix_scale[nchannels++]; + if (scale != (1 << 15)) { + s->dcadsp->dmix_scale(c->bands[band].msb_sample_buffer[j], + scale, s->nframesamples); + if (band) + s->dcadsp->dmix_scale(c->deci_history[j], + scale, DCA_XLL_DECI_HISTORY_MAX); + } + } + + if (nchannels >= o->hier_ofs) + break; + } +} + +// Clear all band data and replace non-residual encoded channels with lossy +// counterparts +static void av_cold force_lossy_output(DCAXllDecoder *s, DCAXllChSet *c) +{ + DCAContext *dca = s->avctx->priv_data; + int band, ch; + + for (band = 0; band < c->nfreqbands; band++) + chs_clear_band_data(s, c, band, -1); + + for (ch = 0; ch < c->nchannels; ch++) { + if (!(c->residual_encode & (1 << ch))) + continue; + if (ff_dca_core_map_spkr(&dca->core, c->ch_remap[ch]) < 0) + continue; + c->residual_encode &= ~(1 << ch); + } +} + +static int combine_residual_frame(DCAXllDecoder *s, DCAXllChSet *c) +{ + DCAContext *dca = s->avctx->priv_data; + int ch, nsamples = s->nframesamples; + DCAXllChSet *o; + + // Verify that core is compatible + if (!(dca->packet & DCA_PACKET_CORE)) { + av_log(s->avctx, AV_LOG_ERROR, "Residual encoded channels are present without core\n"); + return AVERROR(EINVAL); + } + + if (c->freq != dca->core.output_rate) { + av_log(s->avctx, AV_LOG_WARNING, "Sample rate mismatch between core (%d Hz) and XLL (%d Hz)\n", dca->core.output_rate, c->freq); + return AVERROR_INVALIDDATA; + } + + if (nsamples != dca->core.npcmsamples) { + av_log(s->avctx, AV_LOG_WARNING, "Number of samples per frame mismatch between core (%d) and XLL (%d)\n", dca->core.npcmsamples, nsamples); + return AVERROR_INVALIDDATA; + } + + // See if this channel set is downmixed and find the next channel set in + // hierarchy. If downmixed, undo core pre-scaling before combining with + // residual (residual is not scaled). + o = find_next_hier_dmix_chset(s, c); + + // Reduce core bit width and combine with residual + for (ch = 0; ch < c->nchannels; ch++) { + int n, spkr, shift, round; + int32_t *src, *dst; + + if (c->residual_encode & (1 << ch)) + continue; + + // Map this channel to core speaker + spkr = ff_dca_core_map_spkr(&dca->core, c->ch_remap[ch]); + if (spkr < 0) { + av_log(s->avctx, AV_LOG_WARNING, "Residual encoded channel (%d) references unavailable core channel\n", c->ch_remap[ch]); + return AVERROR_INVALIDDATA; + } + + // Account for LSB width + shift = 24 - c->pcm_bit_res + chs_get_lsb_width(s, c, 0, ch); + if (shift > 24) { + av_log(s->avctx, AV_LOG_WARNING, "Invalid core shift (%d bits)\n", shift); + return AVERROR_INVALIDDATA; + } + + round = shift > 0 ? 1 << (shift - 1) : 0; + + src = dca->core.output_samples[spkr]; + dst = c->bands[0].msb_sample_buffer[ch]; + if (o) { + // Undo embedded core downmix pre-scaling + int scale_inv = o->dmix_scale_inv[c->hier_ofs + ch]; + for (n = 0; n < nsamples; n++) + dst[n] += clip23((mul16(src[n], scale_inv) + round) >> shift); + } else { + // No downmix scaling + for (n = 0; n < nsamples; n++) + dst[n] += (src[n] + round) >> shift; + } + } + + return 0; +} + +int ff_dca_xll_filter_frame(DCAXllDecoder *s, AVFrame *frame) +{ + AVCodecContext *avctx = s->avctx; + DCAContext *dca = avctx->priv_data; + DCAExssAsset *asset = &dca->exss.assets[0]; + DCAXllChSet *p = &s->chset[0], *c; + enum AVMatrixEncoding matrix_encoding = AV_MATRIX_ENCODING_NONE; + int i, j, k, ret, shift, nsamples, request_mask; + int ch_remap[DCA_SPEAKER_COUNT]; + + // Force lossy downmixed output during recovery + if (dca->packet & DCA_PACKET_RECOVERY) { + for (i = 0, c = s->chset; i < s->nchsets; i++, c++) { + if (i < s->nactivechsets) + force_lossy_output(s, c); + + if (!c->primary_chset) + c->dmix_embedded = 0; + } + + s->scalable_lsbs = 0; + s->fixed_lsb_width = 0; + } + + // Filter frequency bands for active channel sets + s->output_mask = 0; + for (i = 0, c = s->chset; i < s->nactivechsets; i++, c++) { + chs_filter_band_data(s, c, 0); + + if (c->residual_encode != (1 << c->nchannels) - 1 + && (ret = combine_residual_frame(s, c)) < 0) + return ret; + + if (s->scalable_lsbs) + chs_assemble_msbs_lsbs(s, c, 0); + + if (c->nfreqbands > 1) { + chs_filter_band_data(s, c, 1); + chs_assemble_msbs_lsbs(s, c, 1); + } + + s->output_mask |= c->ch_mask; + } + + // Undo hierarchial downmix and/or apply scaling + for (i = 1, c = &s->chset[1]; i < s->nchsets; i++, c++) { + if (!is_hier_dmix_chset(c)) + continue; + + if (i >= s->nactivechsets) { + for (j = 0; j < c->nfreqbands; j++) + if (c->bands[j].dmix_embedded) + scale_down_mix(s, c, j); + break; + } + + for (j = 0; j < c->nfreqbands; j++) + if (c->bands[j].dmix_embedded) + undo_down_mix(s, c, j); + } + + // Assemble frequency bands for active channel sets + if (s->nfreqbands > 1) { + for (i = 0; i < s->nactivechsets; i++) + if ((ret = chs_assemble_freq_bands(s, &s->chset[i])) < 0) + return ret; + } + + // Normalize to regular 5.1 layout if downmixing + if (dca->request_channel_layout) { + if (s->output_mask & DCA_SPEAKER_MASK_Lss) { + s->output_samples[DCA_SPEAKER_Ls] = s->output_samples[DCA_SPEAKER_Lss]; + s->output_mask = (s->output_mask & ~DCA_SPEAKER_MASK_Lss) | DCA_SPEAKER_MASK_Ls; + } + if (s->output_mask & DCA_SPEAKER_MASK_Rss) { + s->output_samples[DCA_SPEAKER_Rs] = s->output_samples[DCA_SPEAKER_Rss]; + s->output_mask = (s->output_mask & ~DCA_SPEAKER_MASK_Rss) | DCA_SPEAKER_MASK_Rs; + } + } + + // Handle downmixing to stereo request + if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO + && DCA_HAS_STEREO(s->output_mask) && p->dmix_embedded + && (p->dmix_type == DCA_DMIX_TYPE_LoRo || + p->dmix_type == DCA_DMIX_TYPE_LtRt)) + request_mask = DCA_SPEAKER_LAYOUT_STEREO; + else + request_mask = s->output_mask; + if (!ff_dca_set_channel_layout(avctx, ch_remap, request_mask)) + return AVERROR(EINVAL); + + avctx->sample_rate = p->freq << (s->nfreqbands - 1); + + switch (p->storage_bit_res) { + case 16: + avctx->sample_fmt = AV_SAMPLE_FMT_S16P; + break; + case 24: + avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + break; + default: + return AVERROR(EINVAL); + } + + avctx->bits_per_raw_sample = p->storage_bit_res; + avctx->profile = FF_PROFILE_DTS_HD_MA; + avctx->bit_rate = 0; + + frame->nb_samples = nsamples = s->nframesamples << (s->nfreqbands - 1); + if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) + return ret; + + // Downmix primary channel set to stereo + if (request_mask != s->output_mask) { + ff_dca_downmix_to_stereo_fixed(s->dcadsp, s->output_samples, + p->dmix_coeff, nsamples, + s->output_mask); + } + + shift = p->storage_bit_res - p->pcm_bit_res; + for (i = 0; i < avctx->channels; i++) { + int32_t *samples = s->output_samples[ch_remap[i]]; + if (frame->format == AV_SAMPLE_FMT_S16P) { + int16_t *plane = (int16_t *)frame->extended_data[i]; + for (k = 0; k < nsamples; k++) + plane[k] = av_clip_int16(samples[k] * (1 << shift)); + } else { + int32_t *plane = (int32_t *)frame->extended_data[i]; + for (k = 0; k < nsamples; k++) + plane[k] = clip23(samples[k] * (1 << shift)) * (1 << 8); + } + } + + if (!asset->one_to_one_map_ch_to_spkr) { + if (asset->representation_type == DCA_REPR_TYPE_LtRt) + matrix_encoding = AV_MATRIX_ENCODING_DOLBY; + else if (asset->representation_type == DCA_REPR_TYPE_LhRh) + matrix_encoding = AV_MATRIX_ENCODING_DOLBYHEADPHONE; + } else if (request_mask != s->output_mask && p->dmix_type == DCA_DMIX_TYPE_LtRt) { + matrix_encoding = AV_MATRIX_ENCODING_DOLBY; + } + if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0) + return ret; + + return 0; +} + +av_cold void ff_dca_xll_flush(DCAXllDecoder *s) +{ + clear_pbr(s); +} + +av_cold void ff_dca_xll_close(DCAXllDecoder *s) +{ + DCAXllChSet *c; + int i, j; + + for (i = 0, c = s->chset; i < DCA_XLL_CHSETS_MAX; i++, c++) { + for (j = 0; j < DCA_XLL_SAMPLE_BUFFERS_MAX; j++) { + av_freep(&c->sample_buffer[j]); + c->sample_size[j] = 0; + } + } + + av_freep(&s->navi); + s->navi_size = 0; + + av_freep(&s->pbr_buffer); + clear_pbr(s); +} diff --git a/libavcodec/dca_xll.h b/libavcodec/dca_xll.h new file mode 100644 index 0000000..bc0aa65 --- /dev/null +++ b/libavcodec/dca_xll.h @@ -0,0 +1,149 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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_DCA_XLL_H +#define AVCODEC_DCA_XLL_H + +#include "libavutil/common.h" +#include "libavutil/mem.h" + +#include "avcodec.h" +#include "internal.h" +#include "get_bits.h" +#include "dca.h" +#include "dcadsp.h" +#include "dca_exss.h" + +#define DCA_XLL_CHSETS_MAX 3 +#define DCA_XLL_CHANNELS_MAX 8 +#define DCA_XLL_BANDS_MAX 2 +#define DCA_XLL_ADAPT_PRED_ORDER_MAX 16 +#define DCA_XLL_DECI_HISTORY_MAX 8 +#define DCA_XLL_DMIX_SCALES_MAX ((DCA_XLL_CHSETS_MAX - 1) * DCA_XLL_CHANNELS_MAX) +#define DCA_XLL_DMIX_COEFFS_MAX (DCA_XLL_DMIX_SCALES_MAX * DCA_XLL_CHANNELS_MAX) +#define DCA_XLL_PBR_BUFFER_MAX (240 << 10) +#define DCA_XLL_SAMPLE_BUFFERS_MAX 3 + +typedef struct DCAXllBand { + int decor_enabled; ///< Pairwise channel decorrelation flag + int orig_order[DCA_XLL_CHANNELS_MAX]; ///< Original channel order + int decor_coeff[DCA_XLL_CHANNELS_MAX / 2]; ///< Pairwise channel coefficients + + int adapt_pred_order[DCA_XLL_CHANNELS_MAX]; ///< Adaptive predictor order + int highest_pred_order; ///< Highest adaptive predictor order + int fixed_pred_order[DCA_XLL_CHANNELS_MAX]; ///< Fixed predictor order + int adapt_refl_coeff[DCA_XLL_CHANNELS_MAX][DCA_XLL_ADAPT_PRED_ORDER_MAX]; ///< Adaptive predictor reflection coefficients + + int dmix_embedded; ///< Downmix performed by encoder in frequency band + + int lsb_section_size; ///< Size of LSB section in any segment + int nscalablelsbs[DCA_XLL_CHANNELS_MAX]; ///< Number of bits to represent the samples in LSB part + int bit_width_adjust[DCA_XLL_CHANNELS_MAX]; ///< Number of bits discarded by authoring + + int32_t *msb_sample_buffer[DCA_XLL_CHANNELS_MAX]; ///< MSB sample buffer pointers + int32_t *lsb_sample_buffer[DCA_XLL_CHANNELS_MAX]; ///< LSB sample buffer pointers or NULL +} DCAXllBand; + +typedef struct DCAXllChSet { + // Channel set header + int nchannels; ///< Number of channels in the channel set (N) + int residual_encode; ///< Residual encoding mask (0 - residual, 1 - full channel) + int pcm_bit_res; ///< PCM bit resolution (variable) + int storage_bit_res; ///< Storage bit resolution (16 or 24) + int freq; ///< Original sampling frequency (max. 96000 Hz) + + int primary_chset; ///< Primary channel set flag + int dmix_coeffs_present; ///< Downmix coefficients present in stream + int dmix_embedded; ///< Downmix already performed by encoder + int dmix_type; ///< Primary channel set downmix type + int hier_chset; ///< Whether the channel set is part of a hierarchy + int hier_ofs; ///< Number of preceding channels in a hierarchy (M) + int dmix_coeff[DCA_XLL_DMIX_COEFFS_MAX]; ///< Downmixing coefficients + int dmix_scale[DCA_XLL_DMIX_SCALES_MAX]; ///< Downmixing scales + int dmix_scale_inv[DCA_XLL_DMIX_SCALES_MAX]; ///< Inverse downmixing scales + int ch_mask; ///< Channel mask for set + int ch_remap[DCA_XLL_CHANNELS_MAX]; ///< Channel to speaker map + + int nfreqbands; ///< Number of frequency bands (1 or 2) + int nabits; ///< Number of bits to read bit allocation coding parameter + + DCAXllBand bands[DCA_XLL_BANDS_MAX]; ///< Frequency bands + + // Frequency band coding parameters + int seg_common; ///< Segment type + int rice_code_flag[DCA_XLL_CHANNELS_MAX]; ///< Rice coding flag + int bitalloc_hybrid_linear[DCA_XLL_CHANNELS_MAX]; ///< Binary code length for isolated samples + int bitalloc_part_a[DCA_XLL_CHANNELS_MAX]; ///< Coding parameter for part A of segment + int bitalloc_part_b[DCA_XLL_CHANNELS_MAX]; ///< Coding parameter for part B of segment + int nsamples_part_a[DCA_XLL_CHANNELS_MAX]; ///< Number of samples in part A of segment + + // Decimator history + DECLARE_ALIGNED(32, int32_t, deci_history)[DCA_XLL_CHANNELS_MAX][DCA_XLL_DECI_HISTORY_MAX]; ///< Decimator history for frequency band 1 + + // Sample buffers + unsigned int sample_size[DCA_XLL_SAMPLE_BUFFERS_MAX]; + int32_t *sample_buffer[DCA_XLL_SAMPLE_BUFFERS_MAX]; +} DCAXllChSet; + +typedef struct DCAXllDecoder { + AVCodecContext *avctx; + GetBitContext gb; + + int frame_size; ///< Number of bytes in a lossless frame + int nchsets; ///< Number of channels sets per frame + int nframesegs; ///< Number of segments per frame + int nsegsamples_log2; ///< log2(nsegsamples) + int nsegsamples; ///< Samples in segment per one frequency band + int nframesamples_log2; ///< log2(nframesamples) + int nframesamples; ///< Samples in frame per one frequency band + int seg_size_nbits; ///< Number of bits used to read segment size + int band_crc_present; ///< Presence of CRC16 within each frequency band + int scalable_lsbs; ///< MSB/LSB split flag + int ch_mask_nbits; ///< Number of bits used to read channel mask + int fixed_lsb_width; ///< Fixed LSB width + + DCAXllChSet chset[DCA_XLL_CHSETS_MAX]; ///< Channel sets + + int *navi; ///< NAVI table + unsigned int navi_size; + + int nfreqbands; ///< Highest number of frequency bands + int nchannels; ///< Total number of channels in a hierarchy + int nreschsets; ///< Number of channel sets that have residual encoded channels + int nactivechsets; ///< Number of active channel sets to decode + + int hd_stream_id; ///< Previous DTS-HD stream ID for detecting changes + + uint8_t *pbr_buffer; ///< Peak bit rate (PBR) smoothing buffer + int pbr_length; ///< Length in bytes of data currently buffered + int pbr_delay; ///< Delay in frames before decoding buffered data + + DCADSPContext *dcadsp; + + int output_mask; + int32_t *output_samples[DCA_SPEAKER_COUNT]; +} DCAXllDecoder; + +int ff_dca_xll_parse(DCAXllDecoder *s, uint8_t *data, DCAExssAsset *asset); +int ff_dca_xll_filter_frame(DCAXllDecoder *s, AVFrame *frame); +av_cold void ff_dca_xll_flush(DCAXllDecoder *s); +av_cold void ff_dca_xll_close(DCAXllDecoder *s); + +#endif diff --git a/libavcodec/dcadec.c b/libavcodec/dcadec.c new file mode 100644 index 0000000..f3c3972 --- /dev/null +++ b/libavcodec/dcadec.c @@ -0,0 +1,417 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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/opt.h" +#include "libavutil/channel_layout.h" + +#include "dcadec.h" +#include "dcamath.h" +#include "dca_syncwords.h" +#include "profiles.h" + +#define MIN_PACKET_SIZE 16 +#define MAX_PACKET_SIZE 0x104000 + +int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask) +{ + static const uint8_t dca2wav_norm[28] = { + 2, 0, 1, 9, 10, 3, 8, 4, 5, 9, 10, 6, 7, 12, + 13, 14, 3, 6, 7, 11, 12, 14, 16, 15, 17, 8, 4, 5, + }; + + static const uint8_t dca2wav_wide[28] = { + 2, 0, 1, 4, 5, 3, 8, 4, 5, 9, 10, 6, 7, 12, + 13, 14, 3, 9, 10, 11, 12, 14, 16, 15, 17, 8, 4, 5, + }; + + int dca_ch, wav_ch, nchannels = 0; + + if (avctx->request_channel_layout & AV_CH_LAYOUT_NATIVE) { + for (dca_ch = 0; dca_ch < DCA_SPEAKER_COUNT; dca_ch++) + if (dca_mask & (1U << dca_ch)) + ch_remap[nchannels++] = dca_ch; + avctx->channel_layout = dca_mask; + } else { + int wav_mask = 0; + int wav_map[18]; + const uint8_t *dca2wav; + if (dca_mask == DCA_SPEAKER_LAYOUT_7POINT0_WIDE || + dca_mask == DCA_SPEAKER_LAYOUT_7POINT1_WIDE) + dca2wav = dca2wav_wide; + else + dca2wav = dca2wav_norm; + for (dca_ch = 0; dca_ch < 28; dca_ch++) { + if (dca_mask & (1 << dca_ch)) { + wav_ch = dca2wav[dca_ch]; + if (!(wav_mask & (1 << wav_ch))) { + wav_map[wav_ch] = dca_ch; + wav_mask |= 1 << wav_ch; + } + } + } + for (wav_ch = 0; wav_ch < 18; wav_ch++) + if (wav_mask & (1 << wav_ch)) + ch_remap[nchannels++] = wav_map[wav_ch]; + avctx->channel_layout = wav_mask; + } + + avctx->channels = nchannels; + return nchannels; +} + +static uint16_t crc16(const uint8_t *data, int size) +{ + static const uint16_t crctab[16] = { + 0x0000, 0x1021, 0x2042, 0x3063, 0x4084, 0x50a5, 0x60c6, 0x70e7, + 0x8108, 0x9129, 0xa14a, 0xb16b, 0xc18c, 0xd1ad, 0xe1ce, 0xf1ef, + }; + + uint16_t res = 0xffff; + int i; + + for (i = 0; i < size; i++) { + res = (res << 4) ^ crctab[(data[i] >> 4) ^ (res >> 12)]; + res = (res << 4) ^ crctab[(data[i] & 15) ^ (res >> 12)]; + } + + return res; +} + +int ff_dca_check_crc(GetBitContext *s, int p1, int p2) +{ + if (((p1 | p2) & 7) || p1 < 0 || p2 > s->size_in_bits || p2 - p1 < 16) + return -1; + if (crc16(s->buffer + p1 / 8, (p2 - p1) / 8)) + return -1; + return 0; +} + +void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples, + int *coeff_l, int nsamples, int ch_mask) +{ + int pos, spkr, max_spkr = av_log2(ch_mask); + int *coeff_r = coeff_l + av_popcount(ch_mask); + + av_assert0(DCA_HAS_STEREO(ch_mask)); + + // Scale left and right channels + pos = (ch_mask & DCA_SPEAKER_MASK_C); + dcadsp->dmix_scale(samples[DCA_SPEAKER_L], coeff_l[pos ], nsamples); + dcadsp->dmix_scale(samples[DCA_SPEAKER_R], coeff_r[pos + 1], nsamples); + + // Downmix remaining channels + for (spkr = 0; spkr <= max_spkr; spkr++) { + if (!(ch_mask & (1U << spkr))) + continue; + + if (*coeff_l && spkr != DCA_SPEAKER_L) + dcadsp->dmix_add(samples[DCA_SPEAKER_L], samples[spkr], + *coeff_l, nsamples); + + if (*coeff_r && spkr != DCA_SPEAKER_R) + dcadsp->dmix_add(samples[DCA_SPEAKER_R], samples[spkr], + *coeff_r, nsamples); + + coeff_l++; + coeff_r++; + } +} + +void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples, + int *coeff_l, int nsamples, int ch_mask) +{ + int pos, spkr, max_spkr = av_log2(ch_mask); + int *coeff_r = coeff_l + av_popcount(ch_mask); + const float scale = 1.0f / (1 << 15); + + av_assert0(DCA_HAS_STEREO(ch_mask)); + + // Scale left and right channels + pos = (ch_mask & DCA_SPEAKER_MASK_C); + fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_L], samples[DCA_SPEAKER_L], + coeff_l[pos ] * scale, nsamples); + fdsp->vector_fmul_scalar(samples[DCA_SPEAKER_R], samples[DCA_SPEAKER_R], + coeff_r[pos + 1] * scale, nsamples); + + // Downmix remaining channels + for (spkr = 0; spkr <= max_spkr; spkr++) { + if (!(ch_mask & (1U << spkr))) + continue; + + if (*coeff_l && spkr != DCA_SPEAKER_L) + fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_L], samples[spkr], + *coeff_l * scale, nsamples); + + if (*coeff_r && spkr != DCA_SPEAKER_R) + fdsp->vector_fmac_scalar(samples[DCA_SPEAKER_R], samples[spkr], + *coeff_r * scale, nsamples); + + coeff_l++; + coeff_r++; + } +} + +static int convert_bitstream(const uint8_t *src, int src_size, uint8_t *dst, int max_size) +{ + switch (AV_RB32(src)) { + case DCA_SYNCWORD_CORE_BE: + case DCA_SYNCWORD_SUBSTREAM: + memcpy(dst, src, src_size); + return src_size; + case DCA_SYNCWORD_CORE_LE: + case DCA_SYNCWORD_CORE_14B_BE: + case DCA_SYNCWORD_CORE_14B_LE: + return avpriv_dca_convert_bitstream(src, src_size, dst, max_size); + default: + return AVERROR_INVALIDDATA; + } +} + +static int dcadec_decode_frame(AVCodecContext *avctx, void *data, + int *got_frame_ptr, AVPacket *avpkt) +{ + DCAContext *s = avctx->priv_data; + AVFrame *frame = data; + uint8_t *input = avpkt->data; + int input_size = avpkt->size; + int i, ret, prev_packet = s->packet; + + if (input_size < MIN_PACKET_SIZE || input_size > MAX_PACKET_SIZE) { + av_log(avctx, AV_LOG_ERROR, "Invalid packet size\n"); + return AVERROR_INVALIDDATA; + } + + av_fast_malloc(&s->buffer, &s->buffer_size, + FFALIGN(input_size, 4096) + DCA_BUFFER_PADDING_SIZE); + if (!s->buffer) + return AVERROR(ENOMEM); + + for (i = 0, ret = AVERROR_INVALIDDATA; i < input_size - MIN_PACKET_SIZE + 1 && ret < 0; i++) + ret = convert_bitstream(input + i, input_size - i, s->buffer, s->buffer_size); + + if (ret < 0) + return ret; + + input = s->buffer; + input_size = ret; + + s->packet = 0; + + // Parse backward compatible core sub-stream + if (AV_RB32(input) == DCA_SYNCWORD_CORE_BE) { + int frame_size; + + if ((ret = ff_dca_core_parse(&s->core, input, input_size)) < 0) { + s->core_residual_valid = 0; + return ret; + } + + s->packet |= DCA_PACKET_CORE; + + // EXXS data must be aligned on 4-byte boundary + frame_size = FFALIGN(s->core.frame_size, 4); + if (input_size - 4 > frame_size) { + input += frame_size; + input_size -= frame_size; + } + } + + if (!s->core_only) { + DCAExssAsset *asset = NULL; + + // Parse extension sub-stream (EXSS) + if (AV_RB32(input) == DCA_SYNCWORD_SUBSTREAM) { + if ((ret = ff_dca_exss_parse(&s->exss, input, input_size)) < 0) { + if (avctx->err_recognition & AV_EF_EXPLODE) + return ret; + } else { + s->packet |= DCA_PACKET_EXSS; + asset = &s->exss.assets[0]; + } + } + + // Parse XLL component in EXSS + if (asset && (asset->extension_mask & DCA_EXSS_XLL)) { + if ((ret = ff_dca_xll_parse(&s->xll, input, asset)) < 0) { + // Conceal XLL synchronization error + if (ret == AVERROR(EAGAIN) + && (prev_packet & DCA_PACKET_XLL) + && (s->packet & DCA_PACKET_CORE)) + s->packet |= DCA_PACKET_XLL | DCA_PACKET_RECOVERY; + else if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + } else { + s->packet |= DCA_PACKET_XLL; + } + } + + // Parse core extensions in EXSS or backward compatible core sub-stream + if ((s->packet & DCA_PACKET_CORE) + && (ret = ff_dca_core_parse_exss(&s->core, input, asset)) < 0) + return ret; + } + + // Filter the frame + if (s->packet & DCA_PACKET_XLL) { + if (s->packet & DCA_PACKET_CORE) { + int x96_synth = -1; + + // Enable X96 synthesis if needed + if (s->xll.chset[0].freq == 96000 && s->core.sample_rate == 48000) + x96_synth = 1; + + if ((ret = ff_dca_core_filter_fixed(&s->core, x96_synth)) < 0) { + s->core_residual_valid = 0; + return ret; + } + + // Force lossy downmixed output on the first core frame filtered. + // This prevents audible clicks when seeking and is consistent with + // what reference decoder does when there are multiple channel sets. + if (!s->core_residual_valid) { + if (s->xll.nreschsets > 0 && s->xll.nchsets > 1) + s->packet |= DCA_PACKET_RECOVERY; + s->core_residual_valid = 1; + } + } + + if ((ret = ff_dca_xll_filter_frame(&s->xll, frame)) < 0) { + // Fall back to core unless hard error + if (!(s->packet & DCA_PACKET_CORE)) + return ret; + if (ret != AVERROR_INVALIDDATA || (avctx->err_recognition & AV_EF_EXPLODE)) + return ret; + if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) { + s->core_residual_valid = 0; + return ret; + } + } + } else if (s->packet & DCA_PACKET_CORE) { + if ((ret = ff_dca_core_filter_frame(&s->core, frame)) < 0) { + s->core_residual_valid = 0; + return ret; + } + s->core_residual_valid = !!(s->core.filter_mode & DCA_FILTER_MODE_FIXED); + } else { + return AVERROR_INVALIDDATA; + } + + *got_frame_ptr = 1; + + return avpkt->size; +} + +static av_cold void dcadec_flush(AVCodecContext *avctx) +{ + DCAContext *s = avctx->priv_data; + + ff_dca_core_flush(&s->core); + ff_dca_xll_flush(&s->xll); + + s->core_residual_valid = 0; +} + +static av_cold int dcadec_close(AVCodecContext *avctx) +{ + DCAContext *s = avctx->priv_data; + + ff_dca_core_close(&s->core); + ff_dca_xll_close(&s->xll); + + av_freep(&s->buffer); + s->buffer_size = 0; + + return 0; +} + +static av_cold int dcadec_init(AVCodecContext *avctx) +{ + DCAContext *s = avctx->priv_data; + + s->avctx = avctx; + s->core.avctx = avctx; + s->exss.avctx = avctx; + s->xll.avctx = avctx; + + if (ff_dca_core_init(&s->core) < 0) + return AVERROR(ENOMEM); + + ff_dcadsp_init(&s->dcadsp); + s->core.dcadsp = &s->dcadsp; + s->xll.dcadsp = &s->dcadsp; + + switch (avctx->request_channel_layout & ~AV_CH_LAYOUT_NATIVE) { + case 0: + s->request_channel_layout = 0; + break; + case AV_CH_LAYOUT_STEREO: + case AV_CH_LAYOUT_STEREO_DOWNMIX: + s->request_channel_layout = DCA_SPEAKER_LAYOUT_STEREO; + break; + case AV_CH_LAYOUT_5POINT0: + s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT0; + break; + case AV_CH_LAYOUT_5POINT1: + s->request_channel_layout = DCA_SPEAKER_LAYOUT_5POINT1; + break; + default: + av_log(avctx, AV_LOG_WARNING, "Invalid request_channel_layout\n"); + break; + } + + avctx->sample_fmt = AV_SAMPLE_FMT_S32P; + avctx->bits_per_raw_sample = 24; + + return 0; +} + +#define OFFSET(x) offsetof(DCAContext, x) +#define PARAM AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM + +static const AVOption dcadec_options[] = { + { "core_only", "Decode core only without extensions", OFFSET(core_only), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, PARAM }, + { NULL } +}; + +static const AVClass dcadec_class = { + .class_name = "DCA decoder", + .item_name = av_default_item_name, + .option = dcadec_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 = dcadec_init, + .decode = dcadec_decode_frame, + .close = dcadec_close, + .flush = dcadec_flush, + .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF, + .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P, + AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, + .priv_class = &dcadec_class, + .profiles = NULL_IF_CONFIG_SMALL(ff_dca_profiles), + .caps_internal = FF_CODEC_CAP_INIT_CLEANUP, +}; diff --git a/libavcodec/dcadec.h b/libavcodec/dcadec.h new file mode 100644 index 0000000..6726121 --- /dev/null +++ b/libavcodec/dcadec.h @@ -0,0 +1,80 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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_DCADEC_H +#define AVCODEC_DCADEC_H + +#include "libavutil/common.h" +#include "libavutil/float_dsp.h" + +#include "avcodec.h" +#include "get_bits.h" +#include "dca.h" +#include "dcadsp.h" +#include "dca_core.h" +#include "dca_exss.h" +#include "dca_xll.h" + +#define DCA_BUFFER_PADDING_SIZE 1024 + +#define DCA_PACKET_CORE 0x01 +#define DCA_PACKET_EXSS 0x02 +#define DCA_PACKET_XLL 0x04 +#define DCA_PACKET_RECOVERY 0x08 + +typedef struct DCAContext { + const AVClass *class; ///< class for AVOptions + AVCodecContext *avctx; + + DCACoreDecoder core; ///< Core decoder context + DCAExssParser exss; ///< EXSS parser context + DCAXllDecoder xll; ///< XLL decoder context + + DCADSPContext dcadsp; + + uint8_t *buffer; ///< Packet buffer + unsigned int buffer_size; + + int packet; ///< Packet flags + + int core_residual_valid; ///< Core valid for residual decoding + + int request_channel_layout; ///< Converted from avctx.request_channel_layout + int core_only; ///< Core only decoding flag +} DCAContext; + +int ff_dca_set_channel_layout(AVCodecContext *avctx, int *ch_remap, int dca_mask); + +int ff_dca_check_crc(GetBitContext *s, int p1, int p2); + +void ff_dca_downmix_to_stereo_fixed(DCADSPContext *dcadsp, int32_t **samples, + int *coeff_l, int nsamples, int ch_mask); +void ff_dca_downmix_to_stereo_float(AVFloatDSPContext *fdsp, float **samples, + int *coeff_l, int nsamples, int ch_mask); + +static inline int ff_dca_seek_bits(GetBitContext *s, int p) +{ + if (p < s->index || p > s->size_in_bits) + return -1; + s->index = p; + return 0; +} + +#endif diff --git a/libavcodec/dcadsp.c b/libavcodec/dcadsp.c new file mode 100644 index 0000000..cee3d60 --- /dev/null +++ b/libavcodec/dcadsp.c @@ -0,0 +1,413 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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/mem.h" + +#include "dcadsp.h" +#include "dcamath.h" + +static void decode_hf_c(int32_t **dst, + const int32_t *vq_index, + const int8_t hf_vq[1024][32], + int32_t scale_factors[32][2], + intptr_t sb_start, intptr_t sb_end, + intptr_t ofs, intptr_t len) +{ + int i, j; + + for (i = sb_start; i < sb_end; i++) { + const int8_t *coeff = hf_vq[vq_index[i]]; + int32_t scale = scale_factors[i][0]; + for (j = 0; j < len; j++) + dst[i][j + ofs] = clip23(coeff[j] * scale + (1 << 3) >> 4); + } +} + +static void decode_joint_c(int32_t **dst, int32_t **src, + const int32_t *scale_factors, + intptr_t sb_start, intptr_t sb_end, + intptr_t ofs, intptr_t len) +{ + int i, j; + + for (i = sb_start; i < sb_end; i++) { + int32_t scale = scale_factors[i]; + for (j = 0; j < len; j++) + dst[i][j + ofs] = clip23(mul17(src[i][j + ofs], scale)); + } +} + +static void lfe_fir_float_c(float *pcm_samples, int32_t *lfe_samples, + const float *filter_coeff, intptr_t npcmblocks, + int dec_select) +{ + // Select decimation factor + int factor = 64 << dec_select; + int ncoeffs = 8 >> dec_select; + int nlfesamples = npcmblocks >> (dec_select + 1); + int i, j, k; + + for (i = 0; i < nlfesamples; i++) { + // One decimated sample generates 64 or 128 interpolated ones + for (j = 0; j < factor / 2; j++) { + float a = 0; + float b = 0; + + for (k = 0; k < ncoeffs; k++) { + a += filter_coeff[ j * ncoeffs + k] * lfe_samples[-k]; + b += filter_coeff[255 - j * ncoeffs - k] * lfe_samples[-k]; + } + + pcm_samples[ j] = a; + pcm_samples[factor / 2 + j] = b; + } + + lfe_samples++; + pcm_samples += factor; + } +} + +static void lfe_fir1_float_c(float *pcm_samples, int32_t *lfe_samples, + const float *filter_coeff, intptr_t npcmblocks) +{ + lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 0); +} + +static void lfe_fir2_float_c(float *pcm_samples, int32_t *lfe_samples, + const float *filter_coeff, intptr_t npcmblocks) +{ + lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 1); +} + +static void lfe_x96_float_c(float *dst, const float *src, + float *hist, intptr_t len) +{ + float prev = *hist; + int i; + + for (i = 0; i < len; i++) { + float a = 0.25f * src[i] + 0.75f * prev; + float b = 0.75f * src[i] + 0.25f * prev; + prev = src[i]; + *dst++ = a; + *dst++ = b; + } + + *hist = prev; +} + +static void sub_qmf32_float_c(SynthFilterContext *synth, + FFTContext *imdct, + float *pcm_samples, + int32_t **subband_samples_lo, + int32_t **subband_samples_hi, + float *hist1, int *offset, float *hist2, + const float *filter_coeff, intptr_t npcmblocks, + float scale) +{ + LOCAL_ALIGNED(32, float, input, [32]); + int i, j; + + for (j = 0; j < npcmblocks; j++) { + // Load in one sample from each subband + for (i = 0; i < 32; i++) { + if ((i - 1) & 2) + input[i] = -subband_samples_lo[i][j]; + else + input[i] = subband_samples_lo[i][j]; + } + + // One subband sample generates 32 interpolated ones + synth->synth_filter_float(imdct, hist1, offset, + hist2, filter_coeff, + pcm_samples, input, scale); + pcm_samples += 32; + } +} + +static void sub_qmf64_float_c(SynthFilterContext *synth, + FFTContext *imdct, + float *pcm_samples, + int32_t **subband_samples_lo, + int32_t **subband_samples_hi, + float *hist1, int *offset, float *hist2, + const float *filter_coeff, intptr_t npcmblocks, + float scale) +{ + LOCAL_ALIGNED(32, float, input, [64]); + int i, j; + + if (!subband_samples_hi) + memset(&input[32], 0, sizeof(input[0]) * 32); + + for (j = 0; j < npcmblocks; j++) { + // Load in one sample from each subband + if (subband_samples_hi) { + // Full 64 subbands, first 32 are residual coded + for (i = 0; i < 32; i++) { + if ((i - 1) & 2) + input[i] = -subband_samples_lo[i][j] - subband_samples_hi[i][j]; + else + input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j]; + } + for (i = 32; i < 64; i++) { + if ((i - 1) & 2) + input[i] = -subband_samples_hi[i][j]; + else + input[i] = subband_samples_hi[i][j]; + } + } else { + // Only first 32 subbands + for (i = 0; i < 32; i++) { + if ((i - 1) & 2) + input[i] = -subband_samples_lo[i][j]; + else + input[i] = subband_samples_lo[i][j]; + } + } + + // One subband sample generates 64 interpolated ones + synth->synth_filter_float_64(imdct, hist1, offset, + hist2, filter_coeff, + pcm_samples, input, scale); + pcm_samples += 64; + } +} + +static void lfe_fir_fixed_c(int32_t *pcm_samples, int32_t *lfe_samples, + const int32_t *filter_coeff, intptr_t npcmblocks) +{ + // Select decimation factor + int nlfesamples = npcmblocks >> 1; + int i, j, k; + + for (i = 0; i < nlfesamples; i++) { + // One decimated sample generates 64 interpolated ones + for (j = 0; j < 32; j++) { + int64_t a = 0; + int64_t b = 0; + + for (k = 0; k < 8; k++) { + a += (int64_t)filter_coeff[ j * 8 + k] * lfe_samples[-k]; + b += (int64_t)filter_coeff[255 - j * 8 - k] * lfe_samples[-k]; + } + + pcm_samples[ j] = clip23(norm23(a)); + pcm_samples[32 + j] = clip23(norm23(b)); + } + + lfe_samples++; + pcm_samples += 64; + } +} + +static void lfe_x96_fixed_c(int32_t *dst, const int32_t *src, + int32_t *hist, intptr_t len) +{ + int32_t prev = *hist; + int i; + + for (i = 0; i < len; i++) { + int64_t a = INT64_C(2097471) * src[i] + INT64_C(6291137) * prev; + int64_t b = INT64_C(6291137) * src[i] + INT64_C(2097471) * prev; + prev = src[i]; + *dst++ = clip23(norm23(a)); + *dst++ = clip23(norm23(b)); + } + + *hist = prev; +} + +static void sub_qmf32_fixed_c(SynthFilterContext *synth, + DCADCTContext *imdct, + int32_t *pcm_samples, + int32_t **subband_samples_lo, + int32_t **subband_samples_hi, + int32_t *hist1, int *offset, int32_t *hist2, + const int32_t *filter_coeff, intptr_t npcmblocks) +{ + LOCAL_ALIGNED(32, int32_t, input, [32]); + int i, j; + + for (j = 0; j < npcmblocks; j++) { + // Load in one sample from each subband + for (i = 0; i < 32; i++) + input[i] = subband_samples_lo[i][j]; + + // One subband sample generates 32 interpolated ones + synth->synth_filter_fixed(imdct, hist1, offset, + hist2, filter_coeff, + pcm_samples, input); + pcm_samples += 32; + } +} + +static void sub_qmf64_fixed_c(SynthFilterContext *synth, + DCADCTContext *imdct, + int32_t *pcm_samples, + int32_t **subband_samples_lo, + int32_t **subband_samples_hi, + int32_t *hist1, int *offset, int32_t *hist2, + const int32_t *filter_coeff, intptr_t npcmblocks) +{ + LOCAL_ALIGNED(32, int32_t, input, [64]); + int i, j; + + if (!subband_samples_hi) + memset(&input[32], 0, sizeof(input[0]) * 32); + + for (j = 0; j < npcmblocks; j++) { + // Load in one sample from each subband + if (subband_samples_hi) { + // Full 64 subbands, first 32 are residual coded + for (i = 0; i < 32; i++) + input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j]; + for (i = 32; i < 64; i++) + input[i] = subband_samples_hi[i][j]; + } else { + // Only first 32 subbands + for (i = 0; i < 32; i++) + input[i] = subband_samples_lo[i][j]; + } + + // One subband sample generates 64 interpolated ones + synth->synth_filter_fixed_64(imdct, hist1, offset, + hist2, filter_coeff, + pcm_samples, input); + pcm_samples += 64; + } +} + +static void decor_c(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] += src[i] * coeff + (1 << 2) >> 3; +} + +static void dmix_sub_xch_c(int32_t *dst1, int32_t *dst2, + const int32_t *src, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) { + int32_t cs = mul23(src[i], 5931520 /* M_SQRT1_2 * (1 << 23) */); + dst1[i] -= cs; + dst2[i] -= cs; + } +} + +static void dmix_sub_c(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] -= mul15(src[i], coeff); +} + +static void dmix_add_c(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] += mul15(src[i], coeff); +} + +static void dmix_scale_c(int32_t *dst, intptr_t scale, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] = mul15(dst[i], scale); +} + +static void dmix_scale_inv_c(int32_t *dst, intptr_t scale_inv, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] = mul16(dst[i], scale_inv); +} + +static void filter0(int32_t *dst, const int32_t *src, int32_t coeff, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] -= mul22(src[i], coeff); +} + +static void filter1(int32_t *dst, const int32_t *src, int32_t coeff, intptr_t len) +{ + int i; + + for (i = 0; i < len; i++) + dst[i] -= mul23(src[i], coeff); +} + +static void assemble_freq_bands_c(int32_t *dst, int32_t *src0, int32_t *src1, + const int32_t *coeff, intptr_t len) +{ + int i; + + filter0(src0, src1, coeff[0], len); + filter0(src1, src0, coeff[1], len); + filter0(src0, src1, coeff[2], len); + filter0(src1, src0, coeff[3], len); + + for (i = 0; i < 8; i++, src0--) { + filter1(src0, src1, coeff[i + 4], len); + filter1(src1, src0, coeff[i + 12], len); + filter1(src0, src1, coeff[i + 4], len); + } + + for (i = 0; i < len; i++) { + *dst++ = *src1++; + *dst++ = *++src0; + } +} + +av_cold void ff_dcadsp_init(DCADSPContext *s) +{ + s->decode_hf = decode_hf_c; + s->decode_joint = decode_joint_c; + + s->lfe_fir_float[0] = lfe_fir1_float_c; + s->lfe_fir_float[1] = lfe_fir2_float_c; + s->lfe_x96_float = lfe_x96_float_c; + s->sub_qmf_float[0] = sub_qmf32_float_c; + s->sub_qmf_float[1] = sub_qmf64_float_c; + + s->lfe_fir_fixed = lfe_fir_fixed_c; + s->lfe_x96_fixed = lfe_x96_fixed_c; + s->sub_qmf_fixed[0] = sub_qmf32_fixed_c; + s->sub_qmf_fixed[1] = sub_qmf64_fixed_c; + + s->decor = decor_c; + + s->dmix_sub_xch = dmix_sub_xch_c; + s->dmix_sub = dmix_sub_c; + s->dmix_add = dmix_add_c; + s->dmix_scale = dmix_scale_c; + s->dmix_scale_inv = dmix_scale_inv_c; + + s->assemble_freq_bands = assemble_freq_bands_c; +} diff --git a/libavcodec/dcadsp.h b/libavcodec/dcadsp.h new file mode 100644 index 0000000..d8acf37 --- /dev/null +++ b/libavcodec/dcadsp.h @@ -0,0 +1,91 @@ +/* + * Copyright (C) 2016 foo86 + * + * 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 "libavutil/common.h" + +#include "fft.h" +#include "dcadct.h" +#include "synth_filter.h" + +typedef struct DCADSPContext { + void (*decode_hf)(int32_t **dst, + const int32_t *vq_index, + const int8_t hf_vq[1024][32], + int32_t scale_factors[32][2], + intptr_t sb_start, intptr_t sb_end, + intptr_t ofs, intptr_t len); + + void (*decode_joint)(int32_t **dst, int32_t **src, + const int32_t *scale_factors, + intptr_t sb_start, intptr_t sb_end, + intptr_t ofs, intptr_t len); + + void (*lfe_fir_float[2])(float *pcm_samples, int32_t *lfe_samples, + const float *filter_coeff, intptr_t npcmblocks); + + void (*lfe_x96_float)(float *dst, const float *src, + float *hist, intptr_t len); + + void (*sub_qmf_float[2])(SynthFilterContext *synth, + FFTContext *imdct, + float *pcm_samples, + int32_t **subband_samples_lo, + int32_t **subband_samples_hi, + float *hist1, int *offset, float *hist2, + const float *filter_coeff, intptr_t npcmblocks, + float scale); + + void (*lfe_fir_fixed)(int32_t *pcm_samples, int32_t *lfe_samples, + const int32_t *filter_coeff, intptr_t npcmblocks); + + void (*lfe_x96_fixed)(int32_t *dst, const int32_t *src, + int32_t *hist, intptr_t len); + + void (*sub_qmf_fixed[2])(SynthFilterContext *synth, + DCADCTContext *imdct, + int32_t *pcm_samples, + int32_t **subband_samples_lo, + int32_t **subband_samples_hi, + int32_t *hist1, int *offset, int32_t *hist2, + const int32_t *filter_coeff, intptr_t npcmblocks); + + void (*decor)(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len); + + void (*dmix_sub_xch)(int32_t *dst1, int32_t *dst2, + const int32_t *src, intptr_t len); + + void (*dmix_sub)(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len); + + void (*dmix_add)(int32_t *dst, const int32_t *src, intptr_t coeff, intptr_t len); + + void (*dmix_scale)(int32_t *dst, intptr_t scale, intptr_t len); + + void (*dmix_scale_inv)(int32_t *dst, intptr_t scale_inv, intptr_t len); + + void (*assemble_freq_bands)(int32_t *dst, int32_t *src0, int32_t *src1, + const int32_t *coeff, intptr_t len); +} DCADSPContext; + +av_cold void ff_dcadsp_init(DCADSPContext *s); + +#endif diff --git a/libavcodec/version.h b/libavcodec/version.h index 5740137..02063c8 100644 --- a/libavcodec/version.h +++ b/libavcodec/version.h @@ -30,7 +30,7 @@ #define LIBAVCODEC_VERSION_MAJOR 57 #define LIBAVCODEC_VERSION_MINOR 24 -#define LIBAVCODEC_VERSION_MICRO 100 +#define LIBAVCODEC_VERSION_MICRO 101 #define LIBAVCODEC_VERSION_INT AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \ LIBAVCODEC_VERSION_MINOR, \ diff --git a/libavcodec/x86/Makefile b/libavcodec/x86/Makefile index eec98cb..ce06b90 100644 --- a/libavcodec/x86/Makefile +++ b/libavcodec/x86/Makefile @@ -44,7 +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/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 @@ -132,7 +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-$(CONFIG_DCA_DECODER) += x86/synth_filter.o +YASM-OBJS-$(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/tests/checkasm/Makefile b/tests/checkasm/Makefile index 14a11d6..07fe5bc 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-$(CONFIG_DCA_DECODER) += synth_filter.o +AVCODECOBJS-$(CONFIG_DCA_DECODER) += synth_filter.o AVCODECOBJS-$(CONFIG_FLACDSP) += flacdsp.o AVCODECOBJS-$(CONFIG_FMTCONVERT) += fmtconvert.o AVCODECOBJS-$(CONFIG_H264PRED) += h264pred.o diff --git a/tests/checkasm/checkasm.c b/tests/checkasm/checkasm.c index f7d1331..49fd2af 100644 --- a/tests/checkasm/checkasm.c +++ b/tests/checkasm/checkasm.c @@ -71,9 +71,9 @@ static const struct { #if CONFIG_BSWAPDSP { "bswapdsp", checkasm_check_bswapdsp }, #endif -/* #if CONFIG_DCA_DECODER + #if CONFIG_DCA_DECODER { "synth_filter", checkasm_check_synth_filter }, - #endif*/ + #endif #if CONFIG_FLACDSP { "flacdsp", checkasm_check_flacdsp }, #endif diff --git a/tests/fate/acodec.mak b/tests/fate/acodec.mak index 62b1bc1..e0f2320 100644 --- a/tests/fate/acodec.mak +++ b/tests/fate/acodec.mak @@ -99,14 +99,14 @@ FATE_ACODEC-$(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 686b7df..93c19a0 100644 --- a/tests/fate/audio.mak +++ b/tests/fate/audio.mak @@ -21,7 +21,7 @@ fate-dca-core: CMD = pcm -i $(TARGET_SAMPLES)/dts/dts.ts fate-dca-core: CMP = oneoff fate-dca-core: REF = $(SAMPLES)/dts/dts.pcm -#FATE_SAMPLES_AUDIO-$(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 @@ -31,7 +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

2 1