mirror of
https://github.com/mpv-player/mpv
synced 2024-10-06 14:54:02 +02:00
FLAC decoding support via imported libmpflac.
TODO: fix FLAC-in-ogg decoding. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@11005 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
lumag
parent
31235dd9a4
commit
1e880aa659
12
Makefile
12
Makefile
@ -36,7 +36,7 @@ OBJS_MPLAYER = $(SRCS_MPLAYER:.c=.o)
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VO_LIBS = $(AA_LIB) $(X_LIB) $(SDL_LIB) $(GGI_LIB) $(MP1E_LIB) $(MLIB_LIB) $(SVGA_LIB) $(DIRECTFB_LIB)
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AO_LIBS = $(ARTS_LIB) $(ESD_LIB) $(NAS_LIB) $(SGIAUDIO_LIB)
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CODEC_LIBS = $(AV_LIB) $(FAME_LIB) $(MAD_LIB) $(VORBIS_LIB) $(THEORA_LIB) $(FAAD_LIB) $(LIBLZO_LIB) $(DECORE_LIB) $(XVID_LIB) $(PNG_LIB) $(Z_LIB) $(JPEG_LIB) $(ALSA_LIB) $(XMMS_LIB) $(MATROSKA_LIB)
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COMMON_LIBS = libmpcodecs/libmpcodecs.a mp3lib/libMP3.a liba52/liba52.a libmpeg2/libmpeg2.a $(W32_LIB) $(DS_LIB) libaf/libaf.a libmpdemux/libmpdemux.a input/libinput.a postproc/libswscale.a osdep/libosdep.a $(CSS_LIB) $(CODEC_LIBS) $(FREETYPE_LIB) $(TERMCAP_LIB) $(CDPARANOIA_LIB) $(MPLAYER_NETWORK_LIB) $(WIN32_LIB) $(GIF_LIB) $(MACOSX_FRAMEWORKS) $(SMBSUPPORT_LIB) $(FRIBIDI_LIB)
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COMMON_LIBS = libmpcodecs/libmpcodecs.a mp3lib/libMP3.a liba52/liba52.a libmpeg2/libmpeg2.a $(W32_LIB) $(DS_LIB) libaf/libaf.a libmpdemux/libmpdemux.a input/libinput.a postproc/libswscale.a osdep/libosdep.a $(CSS_LIB) $(CODEC_LIBS) $(FREETYPE_LIB) $(TERMCAP_LIB) $(CDPARANOIA_LIB) $(MPLAYER_NETWORK_LIB) $(WIN32_LIB) $(GIF_LIB) $(MACOSX_FRAMEWORKS) $(SMBSUPPORT_LIB) $(FRIBIDI_LIB) $(FLAC_LIB)
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CFLAGS = $(OPTFLAGS) -Ilibmpdemux -Iloader -Ilibvo $(FREETYPE_INC) $(EXTRA_INC) $(CDPARANOIA_INC) $(SDL_INC) $(X11_INC) $(FRIBIDI_INC) $(DVB_INC) # -Wall
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@ -67,6 +67,9 @@ endif
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ifeq ($(LIBMENU),yes)
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PARTS += libmenu
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endif
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ifneq ($(MPFLAC),none)
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PARTS += libmpflac
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endif
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ALL_PRG = $(PRG)
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ifeq ($(MENCODER),yes)
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@ -105,6 +108,10 @@ COMMON_DEPS += Gui/libgui.a
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GUI_LIBS = Gui/libgui.a
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endif
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ifneq ($(MPFLAC),none)
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COMMON_DEPS += libmpflac/libmpflac.a
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endif
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.SUFFIXES: .cc .c .o
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# .PHONY: $(COMMON_DEPS)
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@ -186,6 +193,9 @@ libmenu/libmenu.a:
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libavcodec/libpostproc/libpostproc.so:
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$(MAKE) -C libavcodec/libpostproc
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libmpflac/libmpflac.a:
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$(MAKE) -C libmpflac
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MPLAYER_DEP = $(OBJS_MPLAYER) $(COMMON_DEPS)
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ifeq ($(LIBMENU),yes)
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65
configure
vendored
65
configure
vendored
@ -198,6 +198,8 @@ Codecs:
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--disable-mad disable libmad (mpeg audio) support [autodetect]
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--enable-xmms build with XMMS inputplugin support [disabled]
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--enable-externalfaad use external faad library if available [disabled]
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--enable-flac build with FLAC support [autodetect]
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--enable-external-flac build with external libFLAC [disable]
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Video output:
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--disable-vidix disable VIDIX stuff [enable on x86 *nix]
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@ -1087,6 +1089,8 @@ _tremor=no
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_faad=auto
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_faad_local=yes
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_xmms=no
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_flac=auto
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_external_flac=no
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_css=auto
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# dvdnav disabled, it does not work
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#_dvdnav=no
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@ -1252,6 +1256,10 @@ for ac_option do
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--enable-externalfaad) _faad_local=no ;;
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--disable-externalfaad) _faad_local=yes ;;
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--enable-xmms) _xmms=yes ;;
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--enable-flac) _flac=yes ;;
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--disable-flac) _flac=no ;;
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--enable-external-flac) _external_flac=yes ;;
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--disable-external-flac) _external_flac=no ;;
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--enable-css) _css=yes ;;
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--disable-css) _css=no ;;
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--enable-dvdread) _dvdread=yes ;;
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@ -5168,6 +5176,55 @@ else
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fi
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echores "$_xmms"
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echocheck "FLAC support"
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if test "$_flac" = auto ; then
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if test "$_external_flac" = yes ; then
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cat > $TMPC << EOF
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#include <FLAC/stream_decoder.h>
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#include <stdlib.h>
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int main()
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{
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FLAC__StreamDecoder *fdec = FLAC__stream_decoder_new();
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return fdec != NULL;
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}
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EOF
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_flac=no
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if cc_check -lFLAC ; then
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_flac=external
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fi
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else
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_flac=yes
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fi
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fi
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if test "$_flac" = external ; then
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_def_flac='#define HAVE_FLAC 1'
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#Still use dither.c & replay_gain from libmpflac
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_def_mpflac='#undef USE_MPFLAC_DECODER'
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_mpflac='process'
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_ld_flac='-lFLAC -Llibmpflac -lmpflac'
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_codecmodules="flac(external) $_codecmodules"
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echores "yes (using external libFLAC)"
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else
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if test "$_flac" = yes ; then
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_def_flac='#define HAVE_FLAC 1'
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#use decoder, dither.c & replay_gain from libmpflac
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_def_mpflac='#define USE_MPFLAC_DECODER 1'
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_mpflac='full'
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_ld_flac='-Llibmpflac -lmpflac'
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_codecmodules="flac(internal) $_codecmodules"
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echores "yes (using internal libmpflac)"
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else
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_def_flac='#undef HAVE_FLAC'
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_def_mpflac='#undef USE_MPFLAC_DECODER'
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_mpflac='none'
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_ld_flac=''
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_nocodecmodules="flac $_nocodecmodules"
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echores "no"
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fi
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fi
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echores "$_flac"
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echocheck "inet6"
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if test "$_inet6" = auto ; then
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@ -5615,6 +5672,8 @@ XMMS_PLUGINS = $_xmms
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XMMS_LIB = $_xmms_lib
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MACOSX = $_macosx
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MACOSX_FRAMEWORKS = $_macosx_frameworks
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FLAC_LIB = $_ld_flac
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MPFLAC = $_mpflac
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# --- Some stuff for autoconfigure ----
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$_target_arch
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@ -5833,6 +5892,12 @@ $_def_lirc
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*/
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$_def_lircc
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/*
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* FLAC decoding
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*/
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$_def_flac
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$_def_mpflac
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/* DeCSS support using libcss */
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$_def_css
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@ -24,4 +24,7 @@ Gui/libgui.a: $(wildcard Gui/*.[ch])
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linux/libosdep.a: $(wildcard linux/*.[ch])
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postproc/libpostproc.a: $(wildcard postproc/*.[ch])
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input/libinput.a: $(wildcard input/*.[ch])
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ifneq ($(MPFLAC),none)
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libmpflac/libmpflac.a: $(wildcard libmpflac/*.[ch])
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endif
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@ -2067,3 +2067,13 @@ audiocodec lhacm
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format 0x1104 ; SBC
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driver acm
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dll "lhacm.acm"
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audiocodec flac
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info "Free Lossless Audio Codec"
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status untested
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flags seekable
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comment "using libmpflac or libFLAC. Internal format No"
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format 0x43614C66 ; fLaC with mmioFOURCC
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driver flac
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dll "libmpflac"
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@ -6,7 +6,7 @@ LIBNAME2 = libmpencoders.a
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AUDIO_SRCS_LIB=ad_liba52.c ad_hwac3.c ad_mp3lib.c
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AUDIO_SRCS_NAT=ad_alaw.c ad_dk3adpcm.c ad_pcm.c ad_dvdpcm.c ad_imaadpcm.c ad_msadpcm.c ad_msgsm.c ad_roqaudio.c ad_ra1428.c
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AUDIO_SRCS_OPT=ad_acm.c ad_dshow.c ad_dmo.c ad_qtaudio.c ad_ffmpeg.c ad_faad.c ad_libvorbis.c ad_libmad.c ad_realaud.c ad_libdv.c
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AUDIO_SRCS_OPT=ad_acm.c ad_dshow.c ad_dmo.c ad_qtaudio.c ad_ffmpeg.c ad_faad.c ad_libvorbis.c ad_libmad.c ad_realaud.c ad_libdv.c ad_flac.c
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AUDIO_SRCS=dec_audio.c ad.c $(AUDIO_SRCS_LIB) $(AUDIO_SRCS_NAT) $(AUDIO_SRCS_OPT)
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VIDEO_SRCS_LIB=vd_libmpeg2.c vd_nuv.c vd_lzo.c
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@ -38,6 +38,9 @@ SRCS2=$(ENCODER_SRCS)
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OBJS2=$(SRCS2:.c=.o)
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CFLAGS = $(OPTFLAGS) -I. -Inative -I.. -I../libmpdemux -I../loader $(EXTRA_INC) -D_GNU_SOURCE
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ifneq ($(MPFLAC),none)
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CFLAGS += -I../libmpflac
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endif
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.SUFFIXES: .c .o
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@ -39,6 +39,7 @@ extern ad_functions_t mpcodecs_ad_realaud;
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extern ad_functions_t mpcodecs_ad_libdv;
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extern ad_functions_t mpcodecs_ad_qtaudio;
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extern ad_functions_t mpcodecs_ad_ra1428;
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extern ad_functions_t mpcodecs_ad_flac;
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ad_functions_t* mpcodecs_ad_drivers[] =
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{
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@ -87,5 +88,8 @@ ad_functions_t* mpcodecs_ad_drivers[] =
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&mpcodecs_ad_libdv,
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#endif
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&mpcodecs_ad_ra1428,
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#ifdef HAVE_FLAC
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&mpcodecs_ad_flac,
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#endif
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NULL
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};
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538
libmpcodecs/ad_flac.c
Normal file
538
libmpcodecs/ad_flac.c
Normal file
@ -0,0 +1,538 @@
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/*
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* This is FLAC decoder for MPlayer using stream_decoder from libFLAC
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* (directly or from libmpflac).
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* This file is part of MPlayer, see http://mplayerhq.hu/ for info.
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* Copyright (C) 2003 Dmitry Baryshkov <mitya at school.ioffe.ru>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*
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* parse_double_, grabbag__replaygain_load_from_vorbiscomment, grabbag__replaygain_compute_scale_factor
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* functions are imported from FLAC project (from grabbag lib sources (replaygain.c)) and are
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* Copyright (C) 2002,2003 Josh Coalson under the terms of GPL.
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*/
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/*
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* TODO:
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* in demux_audio use data from seektable block for seeking.
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* support FLAC-in-Ogg.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <math.h>
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#include "config.h"
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#ifdef HAVE_FLAC
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#include "ad_internal.h"
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#include "mp_msg.h"
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static ad_info_t info = {
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"FLAC audio decoder", // name of the driver
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"flac", // driver name. should be the same as filename without ad_
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"Dmitry Baryshkov", // writer/maintainer of _this_ file
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"http://flac.sf.net/", // writer/maintainer/site of the _codec_
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"" // comments
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};
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LIBAD_EXTERN(flac)
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#ifdef USE_MPFLAC_DECODER
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#include "FLAC_stream_decoder.h"
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#include "FLAC_assert.h"
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#include "FLAC_metadata.h"
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#else
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#include "FLAC/stream_decoder.h"
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#include "FLAC/assert.h"
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#include "FLAC/metadata.h"
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#endif
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/* dithering & replaygain always from libmpflac */
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#include "dither.h"
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#include "replaygain_synthesis.h"
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/* Some global constants. Thay have to be configurable, so leaved them as globals. */
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static const FLAC__bool album_mode = true;
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static const int preamp = 0;
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static const FLAC__bool hard_limit = false;
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static const int noise_shaping = 1;
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static const FLAC__bool dither = true;
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typedef struct flac_struct_st
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{
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FLAC__StreamDecoder *flac_dec; /*decoder handle*/
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sh_audio_t *sh; /* link back to corresponding sh */
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/* set this fields before calling FLAC__stream_decoder_process_single */
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unsigned char *buf;
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int minlen;
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int maxlen;
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/* Here goes number written at write_callback */
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int written;
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/* replaygain and dithering via plugin_common */
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FLAC__bool has_replaygain;
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double replay_scale;
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DitherContext dither_context;
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int bits_per_sample;
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} flac_struct_t;
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FLAC__StreamDecoderReadStatus flac_read_callback (const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], unsigned *bytes, void *client_data)
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{
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int b = demux_read_data(((flac_struct_t*)client_data)->sh->ds, buffer, *bytes);
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mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "\nread %d bytes\n", b);
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*bytes = b;
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if (b <= 0)
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return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM;
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return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
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}
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/*FIXME: we need to support format conversion:(flac specs allow bits/sample to be from 4 to 32. Not only 8 and 16 !!!)*/
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FLAC__StreamDecoderWriteStatus flac_write_callback(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data)
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{
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FLAC__byte *buf = ((flac_struct_t*)(client_data))->buf;
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int channel, sample;
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int bps = ((flac_struct_t*)(client_data))->sh->samplesize;
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mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "\nWrite callback (%d bytes)!!!!\n", bps*frame->header.blocksize*frame->header.channels);
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if (buf == NULL)
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{
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/* This is used in control for skipping 1 audio frame */
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return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
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}
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#if 0
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for (sample = 0; sample < frame->header.blocksize; sample ++)
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for (channel = 0; channel < frame->header.channels; channel ++)
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switch (bps)
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{
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case 3:
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buf[bps*(sample*frame->header.channels+channel)+2] = (FLAC__byte)(buffer[channel][sample]>>16);
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case 2:
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buf[bps*(sample*frame->header.channels+channel)+1] = (FLAC__byte)(buffer[channel][sample]>>8);
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buf[bps*(sample*frame->header.channels+channel)+0] = (FLAC__byte)(buffer[channel][sample]);
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break;
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case 1:
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buf[bps*(sample*frame->header.channels+channel)] = buffer[channel][sample]^0x80;
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break;
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}
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#else
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FLAC__plugin_common__apply_gain(
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buf,
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buffer,
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frame->header.blocksize,
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frame->header.channels,
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((flac_struct_t*)(client_data))->bits_per_sample,
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((flac_struct_t*)(client_data))->sh->samplesize * 8,
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((flac_struct_t*)(client_data))->replay_scale,
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hard_limit,
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dither,
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&(((flac_struct_t*)(client_data))->dither_context)
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);
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#endif
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((flac_struct_t*)(client_data))->written += bps*frame->header.blocksize*frame->header.channels;
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return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
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}
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#ifdef local_min
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#undef local_min
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#endif
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#define local_min(a,b) ((a)<(b)?(a):(b))
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static FLAC__bool parse_double_(const FLAC__StreamMetadata_VorbisComment_Entry *entry, double *val)
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{
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char s[32], *end;
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const char *p, *q;
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double v;
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FLAC__ASSERT(0 != entry);
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FLAC__ASSERT(0 != val);
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p = (const char *)entry->entry;
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q = strchr(p, '=');
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if(0 == q)
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return false;
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q++;
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memset(s, 0, sizeof(s)-1);
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strncpy(s, q, local_min(sizeof(s)-1, entry->length - (q-p)));
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v = strtod(s, &end);
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if(end == s)
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return false;
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*val = v;
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return true;
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}
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FLAC__bool grabbag__replaygain_load_from_vorbiscomment(const FLAC__StreamMetadata *block, FLAC__bool album_mode, double *gain, double *peak)
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{
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int gain_offset, peak_offset;
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static const FLAC__byte *tag_title_gain_ = "REPLAYGAIN_TRACK_GAIN";
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static const FLAC__byte *tag_title_peak_ = "REPLAYGAIN_TRACK_PEAK";
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static const FLAC__byte *tag_album_gain_ = "REPLAYGAIN_ALBUM_GAIN";
|
||||
static const FLAC__byte *tag_album_peak_ = "REPLAYGAIN_ALBUM_PEAK";
|
||||
|
||||
FLAC__ASSERT(0 != block);
|
||||
FLAC__ASSERT(block->type == FLAC__METADATA_TYPE_VORBIS_COMMENT);
|
||||
|
||||
if(0 > (gain_offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, /*offset=*/0, (const char *)(album_mode? tag_album_gain_ : tag_title_gain_))))
|
||||
return false;
|
||||
if(0 > (peak_offset = FLAC__metadata_object_vorbiscomment_find_entry_from(block, /*offset=*/0, (const char *)(album_mode? tag_album_peak_ : tag_title_peak_))))
|
||||
return false;
|
||||
|
||||
if(!parse_double_(block->data.vorbis_comment.comments + gain_offset, gain))
|
||||
return false;
|
||||
if(!parse_double_(block->data.vorbis_comment.comments + peak_offset, peak))
|
||||
return false;
|
||||
|
||||
return true;
|
||||
}
|
||||
|
||||
double grabbag__replaygain_compute_scale_factor(double peak, double gain, double preamp, FLAC__bool prevent_clipping)
|
||||
{
|
||||
double scale;
|
||||
FLAC__ASSERT(peak >= 0.0);
|
||||
gain += preamp;
|
||||
scale = (float) pow(10.0, gain * 0.05);
|
||||
if(prevent_clipping && peak > 0.0) {
|
||||
const double max_scale = (float)(1.0 / peak);
|
||||
if(scale > max_scale)
|
||||
scale = max_scale;
|
||||
}
|
||||
return scale;
|
||||
}
|
||||
|
||||
void flac_metadata_callback (const FLAC__StreamDecoder *decoder, const FLAC__StreamMetadata *metadata, void *client_data)
|
||||
{
|
||||
int i, j;
|
||||
sh_audio_t *sh = ((flac_struct_t*)client_data)->sh;
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "Metadata received\n");
|
||||
switch (metadata->type)
|
||||
{
|
||||
case FLAC__METADATA_TYPE_STREAMINFO:
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "STREAMINFO block (%u bytes):\n", metadata->length);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "min_blocksize: %u samples\n", metadata->data.stream_info.min_blocksize);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "max_blocksize: %u samples\n", metadata->data.stream_info.max_blocksize);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "min_framesize: %u bytes\n", metadata->data.stream_info.min_framesize);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "max_framesize: %u bytes\n", metadata->data.stream_info.max_framesize);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "sample_rate: %u Hz\n", metadata->data.stream_info.sample_rate);
|
||||
sh->samplerate = metadata->data.stream_info.sample_rate;
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "channels: %u\n", metadata->data.stream_info.channels);
|
||||
sh->channels = metadata->data.stream_info.channels;
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "bits_per_sample: %u\n", metadata->data.stream_info.bits_per_sample);
|
||||
((flac_struct_t*)client_data)->bits_per_sample = metadata->data.stream_info.bits_per_sample;
|
||||
sh->samplesize = (metadata->data.stream_info.bits_per_sample<=8)?1:2;
|
||||
/* FIXME: need to support dithering to samplesize 4 */
|
||||
sh->sample_format=(sh->samplesize==1)?AFMT_U8:AFMT_S16_LE; // sample format, see libao2/afmt.h
|
||||
sh->o_bps = sh->samplesize * metadata->data.stream_info.channels * metadata->data.stream_info.sample_rate;
|
||||
sh->i_bps = metadata->data.stream_info.bits_per_sample * metadata->data.stream_info.channels * metadata->data.stream_info.sample_rate / 8 / 2;
|
||||
// input data rate (compressed bytes per second)
|
||||
// Compression rate is near 0.5
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "total_samples: %llu\n", metadata->data.stream_info.total_samples);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "md5sum: ");
|
||||
for (i = 0; i < 16; i++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%02hhx", metadata->data.stream_info.md5sum[i]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
|
||||
|
||||
break;
|
||||
case FLAC__METADATA_TYPE_PADDING:
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "PADDING block (%u bytes)\n", metadata->length);
|
||||
break;
|
||||
case FLAC__METADATA_TYPE_APPLICATION:
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "APPLICATION block (%u bytes):\n", metadata->length);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "Application id: 0x");
|
||||
for (i = 0; i < 4; i++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%02hhx", metadata->data.application.id[i]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "\nData: \n");
|
||||
for (i = 0; i < (metadata->length-4)/8; i++)
|
||||
{
|
||||
for(j = 0; j < 8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.application.data[i*8+j]<0x20?'.':metadata->data.application.data[i*8+j]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " | ");
|
||||
for(j = 0; j < 8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.application.data[i*8+j]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
|
||||
}
|
||||
if (metadata->length-4-i*8 != 0)
|
||||
{
|
||||
for(j = 0; j < metadata->length-4-i*8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.application.data[i*8+j]<0x20?'.':metadata->data.application.data[i*8+j]);
|
||||
for(; j <8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " ");
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " | ");
|
||||
for(j = 0; j < metadata->length-4-i*8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.application.data[i*8+j]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
|
||||
}
|
||||
break;
|
||||
case FLAC__METADATA_TYPE_SEEKTABLE:
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "SEEKTABLE block (%u bytes):\n", metadata->length);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d seekpoints:\n", metadata->data.seek_table.num_points);
|
||||
for (i = 0; i < metadata->data.seek_table.num_points; i++)
|
||||
if (metadata->data.seek_table.points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " %3d) sample_number=%llu stream_offset=%llu frame_samples=%u\n", i,
|
||||
metadata->data.seek_table.points[i].sample_number,
|
||||
metadata->data.seek_table.points[i].stream_offset,
|
||||
metadata->data.seek_table.points[i].frame_samples);
|
||||
else
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " %3d) PLACEHOLDER\n", i);
|
||||
break;
|
||||
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "VORBISCOMMENT block (%u bytes):\n", metadata->length);
|
||||
{
|
||||
char entry[metadata->data.vorbis_comment.vendor_string.length+1];
|
||||
memcpy(&entry, metadata->data.vorbis_comment.vendor_string.entry, metadata->data.vorbis_comment.vendor_string.length);
|
||||
entry[metadata->data.vorbis_comment.vendor_string.length] = '\0';
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "vendor_string: %s\n", entry);
|
||||
}
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%d comment(s):\n", metadata->data.vorbis_comment.num_comments);
|
||||
for (i = 0; i < metadata->data.vorbis_comment.num_comments; i++)
|
||||
{
|
||||
char entry[metadata->data.vorbis_comment.comments[i].length];
|
||||
memcpy(&entry, metadata->data.vorbis_comment.comments[i].entry, metadata->data.vorbis_comment.comments[i].length);
|
||||
entry[metadata->data.vorbis_comment.comments[i].length] = '\0';
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%s\n", entry);
|
||||
}
|
||||
{
|
||||
double gain, peak;
|
||||
if(grabbag__replaygain_load_from_vorbiscomment(metadata, album_mode, &gain, &peak))
|
||||
{
|
||||
((flac_struct_t*)client_data)->has_replaygain = true;
|
||||
((flac_struct_t*)client_data)->replay_scale = grabbag__replaygain_compute_scale_factor(peak, gain, (double)preamp, /*prevent_clipping=*/!hard_limit);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "calculated replay_scale: %lf\n", ((flac_struct_t*)client_data)->replay_scale);
|
||||
}
|
||||
}
|
||||
break;
|
||||
case FLAC__METADATA_TYPE_CUESHEET:
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "CUESHEET block (%u bytes):\n", metadata->length);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "mcn: '%s'\n", metadata->data.cue_sheet.media_catalog_number);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "lead_in: %llu\n", metadata->data.cue_sheet.lead_in);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "is_cd: %s\n", metadata->data.cue_sheet.is_cd?"true":"false");
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "num_tracks: %u\n", metadata->data.cue_sheet.num_tracks);
|
||||
for (i = 0; i < metadata->data.cue_sheet.num_tracks; i++)
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "track[%d]:\n", i);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "offset: %llu\n", metadata->data.cue_sheet.tracks[i].offset);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "number: %hhu%s\n", metadata->data.cue_sheet.tracks[i].number, metadata->data.cue_sheet.tracks[i].number==170?"(LEAD-OUT)":"");
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "isrc: '%s'\n", metadata->data.cue_sheet.tracks[i].isrc);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "type: %s\n", metadata->data.cue_sheet.tracks[i].type?"non-audio":"audio");
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "pre_emphasis: %s\n", metadata->data.cue_sheet.tracks[i].pre_emphasis?"true":"false");
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "num_indices: %hhu\n", metadata->data.cue_sheet.tracks[i].num_indices);
|
||||
for (j = 0; j < metadata->data.cue_sheet.tracks[i].num_indices; j++)
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "index[%d]:\n", j);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "offset:%llu\n", metadata->data.cue_sheet.tracks[i].indices[j].offset);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "number:%hhu\n", metadata->data.cue_sheet.tracks[i].indices[j].number);
|
||||
}
|
||||
}
|
||||
break;
|
||||
default: if (metadata->type >= FLAC__METADATA_TYPE_UNDEFINED)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "UNKNOWN block (%u bytes):\n", metadata->length);
|
||||
else
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "Strange block: UNKNOWN #%d < FLAC__METADATA_TYPE_UNDEFINED (%u bytes):\n", metadata->type, metadata->length);
|
||||
for (i = 0; i < (metadata->length)/8; i++)
|
||||
{
|
||||
for(j = 0; j < 8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.unknown.data[i*8+j]<0x20?'.':metadata->data.unknown.data[i*8+j]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " | ");
|
||||
for(j = 0; j < 8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.unknown.data[i*8+j]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
|
||||
}
|
||||
if (metadata->length-i*8 != 0)
|
||||
{
|
||||
for(j = 0; j < metadata->length-i*8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%c", (unsigned char)metadata->data.unknown.data[i*8+j]<0x20?'.':metadata->data.unknown.data[i*8+j]);
|
||||
for(; j <8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " ");
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, " | ");
|
||||
for(j = 0; j < metadata->length-i*8; j++)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "%#02hhx ", metadata->data.unknown.data[i*8+j]);
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "\n");
|
||||
}
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
void flac_error_callback(const FLAC__StreamDecoder *decoder, FLAC__StreamDecoderErrorStatus status, void *client_data)
|
||||
{
|
||||
if (status != FLAC__STREAM_DECODER_ERROR_STATUS_LOST_SYNC)
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "\nError callback called (%s)!!!\n", FLAC__StreamDecoderErrorStatusString[status]);
|
||||
}
|
||||
|
||||
static int preinit(sh_audio_t *sh){
|
||||
// there are default values set for buffering, but you can override them:
|
||||
|
||||
sh->audio_out_minsize=8*4*65535; // due to specs: we assume max 8 channels,
|
||||
// 4 bytes/sample and 65535 samples/frame
|
||||
// So allocating 2Mbytes buffer :)
|
||||
|
||||
// minimum input buffer size (set only if you need input buffering)
|
||||
// (should be the max compressed frame size)
|
||||
sh->audio_in_minsize=2048; // Default: 0 (no input buffer)
|
||||
|
||||
// if you set audio_in_minsize non-zero, the buffer will be allocated
|
||||
// before the init() call by the core, and you can access it via
|
||||
// pointer: sh->audio_in_buffer
|
||||
// it will free'd after uninit(), so you don't have to use malloc/free here!
|
||||
|
||||
return 1; // return values: 1=OK 0=ERROR
|
||||
}
|
||||
|
||||
static int init(sh_audio_t *sh_audio){
|
||||
flac_struct_t *context = (flac_struct_t*)calloc(sizeof(flac_struct_t), 1);
|
||||
|
||||
sh_audio->context = context;
|
||||
context->sh = sh_audio;
|
||||
if (context == NULL)
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "flac_init: error allocating context.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
context->flac_dec = FLAC__stream_decoder_new();
|
||||
if (context->flac_dec == NULL)
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "flac_init: error allocaing FLAC decoder.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (!FLAC__stream_decoder_set_client_data(context->flac_dec, context))
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting private data for callbacks.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (!FLAC__stream_decoder_set_read_callback(context->flac_dec, &flac_read_callback))
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting read callback.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (!FLAC__stream_decoder_set_write_callback(context->flac_dec, &flac_write_callback))
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting write callback.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (!FLAC__stream_decoder_set_metadata_callback(context->flac_dec, &flac_metadata_callback))
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting metadata callback.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (!FLAC__stream_decoder_set_error_callback(context->flac_dec, &flac_error_callback))
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error setting error callback.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (!FLAC__stream_decoder_set_metadata_respond_all(context->flac_dec))
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "error during setting metadata_respond_all.\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
if (FLAC__stream_decoder_init(context->flac_dec) != FLAC__STREAM_DECODER_SEARCH_FOR_METADATA)
|
||||
{
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Error initializing decoder!\n");
|
||||
return 0;
|
||||
}
|
||||
|
||||
context->buf = NULL;
|
||||
context->minlen = context->maxlen = 0;
|
||||
context->replay_scale = 1.0;
|
||||
|
||||
FLAC__stream_decoder_process_until_end_of_metadata(context->flac_dec);
|
||||
|
||||
FLAC__plugin_common__init_dither_context(&(context->dither_context), sh_audio->samplesize * 8, noise_shaping);
|
||||
|
||||
return 1; // return values: 1=OK 0=ERROR
|
||||
}
|
||||
|
||||
static void uninit(sh_audio_t *sh){
|
||||
// uninit the decoder etc...
|
||||
FLAC__stream_decoder_finish(((flac_struct_t*)(sh->context))->flac_dec);
|
||||
FLAC__stream_decoder_delete(((flac_struct_t*)(sh->context))->flac_dec);
|
||||
// again: you don't have to free() a_in_buffer here! it's done by the core.
|
||||
}
|
||||
|
||||
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen){
|
||||
FLAC__StreamDecoderState decstate;
|
||||
FLAC__bool status;
|
||||
|
||||
// audio decoding. the most important thing :)
|
||||
// parameters you get:
|
||||
// buf = pointer to the output buffer, you have to store uncompressed
|
||||
// samples there
|
||||
// minlen = requested minimum size (in bytes!) of output. it's just a
|
||||
// _recommendation_, you can decode more or less, it just tell you that
|
||||
// the caller process needs 'minlen' bytes. if it gets less, it will
|
||||
// call decode_audio() again.
|
||||
// maxlen = maximum size (bytes) of output. you MUST NOT write more to the
|
||||
// buffer, it's the upper-most limit!
|
||||
// note: maxlen will be always greater or equal to sh->audio_out_minsize
|
||||
|
||||
// Store params in private context for callback:
|
||||
((flac_struct_t*)(sh_audio->context))->buf = buf;
|
||||
((flac_struct_t*)(sh_audio->context))->minlen = minlen;
|
||||
((flac_struct_t*)(sh_audio->context))->maxlen = maxlen;
|
||||
((flac_struct_t*)(sh_audio->context))->written = 0;
|
||||
|
||||
status = FLAC__stream_decoder_process_single(((flac_struct_t*)(sh_audio->context))->flac_dec);
|
||||
decstate = FLAC__stream_decoder_get_state(((flac_struct_t*)(sh_audio->context))->flac_dec);
|
||||
if (!status || (
|
||||
decstate != FLAC__STREAM_DECODER_SEARCH_FOR_METADATA &&
|
||||
decstate != FLAC__STREAM_DECODER_READ_METADATA &&
|
||||
decstate != FLAC__STREAM_DECODER_SEARCH_FOR_FRAME_SYNC &&
|
||||
decstate != FLAC__STREAM_DECODER_READ_FRAME
|
||||
))
|
||||
{
|
||||
if (decstate == FLAC__STREAM_DECODER_END_OF_STREAM)
|
||||
{
|
||||
/* return what we have decoded */
|
||||
if (((flac_struct_t*)(sh_audio->context))->written != 0)
|
||||
return ((flac_struct_t*)(sh_audio->context))->written;
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_V, "End of stream.\n");
|
||||
return -1;
|
||||
}
|
||||
mp_msg(MSGT_DECAUDIO, MSGL_WARN, "process_single problem: returned %s, state is %s!\n", status?"true":"false", FLAC__StreamDecoderStateString[decstate]);
|
||||
FLAC__stream_decoder_flush(((flac_struct_t*)(sh_audio->context))->flac_dec);
|
||||
return -1;
|
||||
}
|
||||
|
||||
|
||||
return ((flac_struct_t*)(sh_audio->context))->written; // return value: number of _bytes_ written to output buffer,
|
||||
// or -1 for EOF (or uncorrectable error)
|
||||
}
|
||||
|
||||
static int control(sh_audio_t *sh,int cmd,void* arg, ...){
|
||||
switch(cmd){
|
||||
case ADCTRL_RESYNC_STREAM:
|
||||
// it is called once after seeking, to resync.
|
||||
// Note: sh_audio->a_in_buffer_len=0; is done _before_ this call!
|
||||
FLAC__stream_decoder_flush (((flac_struct_t*)(sh->context))->flac_dec);
|
||||
return CONTROL_TRUE;
|
||||
case ADCTRL_SKIP_FRAME:
|
||||
// it is called to skip (jump over) small amount (1/10 sec or 1 frame)
|
||||
// of audio data - used to sync audio to video after seeking
|
||||
// if you don't return CONTROL_TRUE, it will defaults to:
|
||||
// ds_fill_buffer(sh_audio->ds); // skip 1 demux packet
|
||||
((flac_struct_t*)(sh->context))->buf = NULL;
|
||||
((flac_struct_t*)(sh->context))->minlen =
|
||||
((flac_struct_t*)(sh->context))->maxlen = 0;
|
||||
FLAC__stream_decoder_process_single(((flac_struct_t*)(sh->context))->flac_dec);
|
||||
return CONTROL_TRUE;
|
||||
}
|
||||
return CONTROL_UNKNOWN;
|
||||
}
|
||||
#endif
|
||||
@ -17,6 +17,7 @@
|
||||
|
||||
#define MP3 1
|
||||
#define WAV 2
|
||||
#define fLaC 3
|
||||
|
||||
|
||||
#define HDR_SIZE 4
|
||||
@ -79,6 +80,10 @@ int demux_audio_open(demuxer_t* demuxer) {
|
||||
} else if((n = mp_get_mp3_header(hdr,&mp3_chans,&mp3_freq)) > 0) {
|
||||
frmt = MP3;
|
||||
break;
|
||||
} else if( hdr[0] == 'f' && hdr[1] == 'L' && hdr[2] == 'a' && hdr[3] == 'C' ) {
|
||||
frmt = fLaC;
|
||||
stream_skip(s,-4);
|
||||
break;
|
||||
}
|
||||
// Add here some other audio format detection
|
||||
if(step < HDR_SIZE)
|
||||
@ -202,6 +207,11 @@ int demux_audio_open(demuxer_t* demuxer) {
|
||||
demuxer->movi_end = s->end_pos;
|
||||
// printf("wav: %X .. %X\n",(int)demuxer->movi_start,(int)demuxer->movi_end);
|
||||
} break;
|
||||
case fLaC:
|
||||
sh_audio->format = mmioFOURCC('f', 'L', 'a', 'C');
|
||||
demuxer->movi_start = stream_tell(s);
|
||||
demuxer->movi_end = s->end_pos;
|
||||
break;
|
||||
}
|
||||
|
||||
priv = (da_priv_t*)malloc(sizeof(da_priv_t));
|
||||
@ -272,6 +282,16 @@ int demux_audio_fill_buffer(demux_stream_t *ds) {
|
||||
ds_add_packet(ds,dp);
|
||||
return 1;
|
||||
}
|
||||
case fLaC: {
|
||||
int l = 65535;
|
||||
demux_packet_t* dp = new_demux_packet(l);
|
||||
l = stream_read(s,dp->buffer,l);
|
||||
resize_demux_packet(dp, l);
|
||||
priv->last_pts = priv->last_pts < 0 ? 0 : priv->last_pts + l/(float)sh_audio->i_bps;
|
||||
ds->pts = priv->last_pts - (ds_tell_pts(demux->audio)-sh_audio->a_in_buffer_len)/(float)sh_audio->i_bps;
|
||||
ds_add_packet(ds,dp);
|
||||
return 1;
|
||||
}
|
||||
default:
|
||||
printf("Audio demuxer : unknown format %d\n",priv->frmt);
|
||||
}
|
||||
|
||||
@ -112,6 +112,7 @@ typedef struct ogg_stream {
|
||||
int hdr_packets;
|
||||
int vorbis;
|
||||
int theora;
|
||||
int flac;
|
||||
} ogg_stream_t;
|
||||
|
||||
typedef struct ogg_demuxer {
|
||||
@ -362,6 +363,11 @@ static unsigned char* demux_ogg_read_packet(ogg_stream_t* os,ogg_packet* pack,vo
|
||||
}
|
||||
}
|
||||
#endif /* HAVE_OGGTHEORA */
|
||||
# ifdef HAVE_FLAC
|
||||
} else if (os->flac) {
|
||||
/* we pass complete packets to flac, mustn't strip the header! */
|
||||
data = pack->packet;
|
||||
#endif /* HAVE_FLAC */
|
||||
} else {
|
||||
// Find data start
|
||||
int16_t hdrlen = (*pack->packet & PACKET_LEN_BITS01)>>6;
|
||||
@ -679,6 +685,16 @@ int demux_ogg_open(demuxer_t* demuxer) {
|
||||
if(verbose>0) print_video_header(sh_v->bih);
|
||||
}
|
||||
# endif /* HAVE_OGGTHEORA */
|
||||
# ifdef HAVE_FLAC
|
||||
} else if (pack.bytes >= 4 && !strncmp (&pack.packet[0], "fLaC", 4)) {
|
||||
sh_a = new_sh_audio(demuxer,ogg_d->num_sub);
|
||||
sh_a->format = mmioFOURCC('f', 'L', 'a', 'C');
|
||||
n_audio++;
|
||||
ogg_d->subs[ogg_d->num_sub].flac = 1;
|
||||
sh_a->wf = NULL;
|
||||
mp_msg(MSGT_DEMUX,MSGL_V,"OGG : stream %d is FLAC\n",ogg_d->num_sub);
|
||||
# endif /* HAVE_FLAC */
|
||||
|
||||
/// Check for old header
|
||||
} else if(pack.bytes >= 142 && ! strncmp(&pack.packet[1],"Direct Show Samples embedded in Ogg",35) ) {
|
||||
|
||||
|
||||
@ -36,6 +36,8 @@ static struct {
|
||||
{ "y4m", DEMUXER_TYPE_Y4M },
|
||||
{ "mp3", DEMUXER_TYPE_AUDIO },
|
||||
{ "wav", DEMUXER_TYPE_AUDIO },
|
||||
{ "flac", DEMUXER_TYPE_AUDIO },
|
||||
{ "fla", DEMUXER_TYPE_AUDIO },
|
||||
{ "ogg", DEMUXER_TYPE_OGG },
|
||||
{ "ogm", DEMUXER_TYPE_OGG },
|
||||
// { "pls", DEMUXER_TYPE_PLAYLIST },
|
||||
|
||||
Loading…
Reference in New Issue
Block a user