Mercurial > mplayer.hg
changeset 11004:d48eccbbb984
FLAC decoding support via imported libmpflac.
TODO: fix FLAC-in-ogg decoding.
| author | lumag |
|---|---|
| date | Sat, 04 Oct 2003 22:00:25 +0000 |
| parents | 6111db8a76b5 |
| children | 3f3396e27ee0 |
| files | Makefile configure etc/.libdeps etc/codecs.conf libmpcodecs/Makefile libmpcodecs/ad.c libmpcodecs/ad_flac.c libmpdemux/demux_audio.c libmpdemux/demux_ogg.c libmpdemux/extension.c |
| diffstat | 10 files changed, 673 insertions(+), 2 deletions(-) [+] |
line wrap: on
line diff
--- a/Makefile Sat Oct 04 20:05:54 2003 +0000 +++ b/Makefile Sat Oct 04 22:00:25 2003 +0000 @@ -36,7 +36,7 @@ VO_LIBS = $(AA_LIB) $(X_LIB) $(SDL_LIB) $(GGI_LIB) $(MP1E_LIB) $(MLIB_LIB) $(SVGA_LIB) $(DIRECTFB_LIB) AO_LIBS = $(ARTS_LIB) $(ESD_LIB) $(NAS_LIB) $(SGIAUDIO_LIB) 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) -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) +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) CFLAGS = $(OPTFLAGS) -Ilibmpdemux -Iloader -Ilibvo $(FREETYPE_INC) $(EXTRA_INC) $(CDPARANOIA_INC) $(SDL_INC) $(X11_INC) $(FRIBIDI_INC) $(DVB_INC) # -Wall @@ -67,6 +67,9 @@ ifeq ($(LIBMENU),yes) PARTS += libmenu endif +ifneq ($(MPFLAC),none) +PARTS += libmpflac +endif ALL_PRG = $(PRG) ifeq ($(MENCODER),yes) @@ -105,6 +108,10 @@ GUI_LIBS = Gui/libgui.a endif +ifneq ($(MPFLAC),none) +COMMON_DEPS += libmpflac/libmpflac.a +endif + .SUFFIXES: .cc .c .o # .PHONY: $(COMMON_DEPS) @@ -186,6 +193,9 @@ libavcodec/libpostproc/libpostproc.so: $(MAKE) -C libavcodec/libpostproc +libmpflac/libmpflac.a: + $(MAKE) -C libmpflac + MPLAYER_DEP = $(OBJS_MPLAYER) $(COMMON_DEPS) ifeq ($(LIBMENU),yes)
--- a/configure Sat Oct 04 20:05:54 2003 +0000 +++ b/configure Sat Oct 04 22:00:25 2003 +0000 @@ -198,6 +198,8 @@ --disable-mad disable libmad (mpeg audio) support [autodetect] --enable-xmms build with XMMS inputplugin support [disabled] --enable-externalfaad use external faad library if available [disabled] + --enable-flac build with FLAC support [autodetect] + --enable-external-flac build with external libFLAC [disable] Video output: --disable-vidix disable VIDIX stuff [enable on x86 *nix] @@ -1087,6 +1089,8 @@ _faad=auto _faad_local=yes _xmms=no +_flac=auto +_external_flac=no _css=auto # dvdnav disabled, it does not work #_dvdnav=no @@ -1252,6 +1256,10 @@ --enable-externalfaad) _faad_local=no ;; --disable-externalfaad) _faad_local=yes ;; --enable-xmms) _xmms=yes ;; + --enable-flac) _flac=yes ;; + --disable-flac) _flac=no ;; + --enable-external-flac) _external_flac=yes ;; + --disable-external-flac) _external_flac=no ;; --enable-css) _css=yes ;; --disable-css) _css=no ;; --enable-dvdread) _dvdread=yes ;; @@ -5168,6 +5176,55 @@ fi echores "$_xmms" +echocheck "FLAC support" +if test "$_flac" = auto ; then + if test "$_external_flac" = yes ; then + cat > $TMPC << EOF +#include <FLAC/stream_decoder.h> +#include <stdlib.h> + +int main() +{ + FLAC__StreamDecoder *fdec = FLAC__stream_decoder_new(); + return fdec != NULL; +} +EOF + _flac=no + if cc_check -lFLAC ; then + _flac=external + fi + else + _flac=yes + fi +fi + +if test "$_flac" = external ; then + _def_flac='#define HAVE_FLAC 1' + #Still use dither.c & replay_gain from libmpflac + _def_mpflac='#undef USE_MPFLAC_DECODER' + _mpflac='process' + _ld_flac='-lFLAC -Llibmpflac -lmpflac' + _codecmodules="flac(external) $_codecmodules" + echores "yes (using external libFLAC)" +else + if test "$_flac" = yes ; then + _def_flac='#define HAVE_FLAC 1' + #use decoder, dither.c & replay_gain from libmpflac + _def_mpflac='#define USE_MPFLAC_DECODER 1' + _mpflac='full' + _ld_flac='-Llibmpflac -lmpflac' + _codecmodules="flac(internal) $_codecmodules" + echores "yes (using internal libmpflac)" + else + _def_flac='#undef HAVE_FLAC' + _def_mpflac='#undef USE_MPFLAC_DECODER' + _mpflac='none' + _ld_flac='' + _nocodecmodules="flac $_nocodecmodules" + echores "no" + fi +fi +echores "$_flac" echocheck "inet6" if test "$_inet6" = auto ; then @@ -5615,6 +5672,8 @@ XMMS_LIB = $_xmms_lib MACOSX = $_macosx MACOSX_FRAMEWORKS = $_macosx_frameworks +FLAC_LIB = $_ld_flac +MPFLAC = $_mpflac # --- Some stuff for autoconfigure ---- $_target_arch @@ -5833,6 +5892,12 @@ */ $_def_lircc +/* + * FLAC decoding + */ +$_def_flac +$_def_mpflac + /* DeCSS support using libcss */ $_def_css
--- a/etc/.libdeps Sat Oct 04 20:05:54 2003 +0000 +++ b/etc/.libdeps Sat Oct 04 22:00:25 2003 +0000 @@ -24,4 +24,7 @@ linux/libosdep.a: $(wildcard linux/*.[ch]) postproc/libpostproc.a: $(wildcard postproc/*.[ch]) input/libinput.a: $(wildcard input/*.[ch]) +ifneq ($(MPFLAC),none) +libmpflac/libmpflac.a: $(wildcard libmpflac/*.[ch]) +endif
--- a/etc/codecs.conf Sat Oct 04 20:05:54 2003 +0000 +++ b/etc/codecs.conf Sat Oct 04 22:00:25 2003 +0000 @@ -2067,3 +2067,13 @@ format 0x1104 ; SBC driver acm dll "lhacm.acm" + +audiocodec flac + info "Free Lossless Audio Codec" + status untested + flags seekable + comment "using libmpflac or libFLAC. Internal format No" + format 0x43614C66 ; fLaC with mmioFOURCC + driver flac + dll "libmpflac" +
--- a/libmpcodecs/Makefile Sat Oct 04 20:05:54 2003 +0000 +++ b/libmpcodecs/Makefile Sat Oct 04 22:00:25 2003 +0000 @@ -6,7 +6,7 @@ AUDIO_SRCS_LIB=ad_liba52.c ad_hwac3.c ad_mp3lib.c 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 -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 +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 AUDIO_SRCS=dec_audio.c ad.c $(AUDIO_SRCS_LIB) $(AUDIO_SRCS_NAT) $(AUDIO_SRCS_OPT) VIDEO_SRCS_LIB=vd_libmpeg2.c vd_nuv.c vd_lzo.c @@ -38,6 +38,9 @@ OBJS2=$(SRCS2:.c=.o) CFLAGS = $(OPTFLAGS) -I. -Inative -I.. -I../libmpdemux -I../loader $(EXTRA_INC) -D_GNU_SOURCE +ifneq ($(MPFLAC),none) +CFLAGS += -I../libmpflac +endif .SUFFIXES: .c .o
--- a/libmpcodecs/ad.c Sat Oct 04 20:05:54 2003 +0000 +++ b/libmpcodecs/ad.c Sat Oct 04 22:00:25 2003 +0000 @@ -39,6 +39,7 @@ extern ad_functions_t mpcodecs_ad_libdv; extern ad_functions_t mpcodecs_ad_qtaudio; extern ad_functions_t mpcodecs_ad_ra1428; +extern ad_functions_t mpcodecs_ad_flac; ad_functions_t* mpcodecs_ad_drivers[] = { @@ -87,5 +88,8 @@ &mpcodecs_ad_libdv, #endif &mpcodecs_ad_ra1428, +#ifdef HAVE_FLAC + &mpcodecs_ad_flac, +#endif NULL };
--- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/libmpcodecs/ad_flac.c Sat Oct 04 22:00:25 2003 +0000 @@ -0,0 +1,538 @@ +/* + * This is FLAC decoder for MPlayer using stream_decoder from libFLAC + * (directly or from libmpflac). + * This file is part of MPlayer, see http://mplayerhq.hu/ for info. + * Copyright (C) 2003 Dmitry Baryshkov <mitya at school.ioffe.ru> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * parse_double_, grabbag__replaygain_load_from_vorbiscomment, grabbag__replaygain_compute_scale_factor + * functions are imported from FLAC project (from grabbag lib sources (replaygain.c)) and are + * Copyright (C) 2002,2003 Josh Coalson under the terms of GPL. + */ + +/* + * TODO: + * in demux_audio use data from seektable block for seeking. + * support FLAC-in-Ogg. + */ + +#include <stdio.h> +#include <stdlib.h> +#include <unistd.h> +#include <math.h> + +#include "config.h" +#ifdef HAVE_FLAC +#include "ad_internal.h" +#include "mp_msg.h" + +static ad_info_t info = { + "FLAC audio decoder", // name of the driver + "flac", // driver name. should be the same as filename without ad_ + "Dmitry Baryshkov", // writer/maintainer of _this_ file + "http://flac.sf.net/", // writer/maintainer/site of the _codec_ + "" // comments +}; + +LIBAD_EXTERN(flac) + +#ifdef USE_MPFLAC_DECODER +#include "FLAC_stream_decoder.h" +#include "FLAC_assert.h" +#include "FLAC_metadata.h" +#else +#include "FLAC/stream_decoder.h" +#include "FLAC/assert.h" +#include "FLAC/metadata.h" +#endif + +/* dithering & replaygain always from libmpflac */ +#include "dither.h" +#include "replaygain_synthesis.h" + +/* Some global constants. Thay have to be configurable, so leaved them as globals. */ +static const FLAC__bool album_mode = true; +static const int preamp = 0; +static const FLAC__bool hard_limit = false; +static const int noise_shaping = 1; +static const FLAC__bool dither = true; +typedef struct flac_struct_st +{ + FLAC__StreamDecoder *flac_dec; /*decoder handle*/ + sh_audio_t *sh; /* link back to corresponding sh */ + + /* set this fields before calling FLAC__stream_decoder_process_single */ + unsigned char *buf; + int minlen; + int maxlen; + /* Here goes number written at write_callback */ + int written; + + /* replaygain and dithering via plugin_common */ + FLAC__bool has_replaygain; + double replay_scale; + DitherContext dither_context; + int bits_per_sample; +} flac_struct_t; + +FLAC__StreamDecoderReadStatus flac_read_callback (const FLAC__StreamDecoder *decoder, FLAC__byte buffer[], unsigned *bytes, void *client_data) +{ + int b = demux_read_data(((flac_struct_t*)client_data)->sh->ds, buffer, *bytes); + mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "\nread %d bytes\n", b); + *bytes = b; + if (b <= 0) + return FLAC__STREAM_DECODER_READ_STATUS_END_OF_STREAM; + return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE; +} + +/*FIXME: we need to support format conversion:(flac specs allow bits/sample to be from 4 to 32. Not only 8 and 16 !!!)*/ +FLAC__StreamDecoderWriteStatus flac_write_callback(const FLAC__StreamDecoder *decoder, const FLAC__Frame *frame, const FLAC__int32 * const buffer[], void *client_data) +{ + FLAC__byte *buf = ((flac_struct_t*)(client_data))->buf; + int channel, sample; + int bps = ((flac_struct_t*)(client_data))->sh->samplesize; + mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "\nWrite callback (%d bytes)!!!!\n", bps*frame->header.blocksize*frame->header.channels); + if (buf == NULL) + { + /* This is used in control for skipping 1 audio frame */ + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; + } +#if 0 + for (sample = 0; sample < frame->header.blocksize; sample ++) + for (channel = 0; channel < frame->header.channels; channel ++) + switch (bps) + { + case 3: + buf[bps*(sample*frame->header.channels+channel)+2] = (FLAC__byte)(buffer[channel][sample]>>16); + case 2: + buf[bps*(sample*frame->header.channels+channel)+1] = (FLAC__byte)(buffer[channel][sample]>>8); + buf[bps*(sample*frame->header.channels+channel)+0] = (FLAC__byte)(buffer[channel][sample]); + break; + case 1: + buf[bps*(sample*frame->header.channels+channel)] = buffer[channel][sample]^0x80; + break; + } +#else + FLAC__plugin_common__apply_gain( + buf, + buffer, + frame->header.blocksize, + frame->header.channels, + ((flac_struct_t*)(client_data))->bits_per_sample, + ((flac_struct_t*)(client_data))->sh->samplesize * 8, + ((flac_struct_t*)(client_data))->replay_scale, + hard_limit, + dither, + &(((flac_struct_t*)(client_data))->dither_context) + ); +#endif + ((flac_struct_t*)(client_data))->written += bps*frame->header.blocksize*frame->header.channels; + return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE; +} + +#ifdef local_min +#undef local_min +#endif +#define local_min(a,b) ((a)<(b)?(a):(b)) + +static FLAC__bool parse_double_(const FLAC__StreamMetadata_VorbisComment_Entry *entry, double *val) +{ + char s[32], *end; + const char *p, *q; + double v; + + FLAC__ASSERT(0 != entry); + FLAC__ASSERT(0 != val); + + p = (const char *)entry->entry; + q = strchr(p, '='); + if(0 == q) + return false; + q++; + memset(s, 0, sizeof(s)-1); + strncpy(s, q, local_min(sizeof(s)-1, entry->length - (q-p))); + + v = strtod(s, &end); + if(end == s) + return false; + + *val = v; + return true; +} + +FLAC__bool grabbag__replaygain_load_from_vorbiscomment(const FLAC__StreamMetadata *block, FLAC__bool album_mode, double *gain, double *peak) +{ + int gain_offset, peak_offset; +static const FLAC__byte *tag_title_gain_ = "REPLAYGAIN_TRACK_GAIN"; +static const FLAC__byte *tag_title_peak_ = "REPLAYGAIN_TRACK_PEAK"; +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
--- a/libmpdemux/demux_audio.c Sat Oct 04 20:05:54 2003 +0000 +++ b/libmpdemux/demux_audio.c Sat Oct 04 22:00:25 2003 +0000 @@ -17,6 +17,7 @@ #define MP3 1 #define WAV 2 +#define fLaC 3 #define HDR_SIZE 4 @@ -79,6 +80,10 @@ } 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 @@ 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 @@ 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); }
--- a/libmpdemux/demux_ogg.c Sat Oct 04 20:05:54 2003 +0000 +++ b/libmpdemux/demux_ogg.c Sat Oct 04 22:00:25 2003 +0000 @@ -112,6 +112,7 @@ int hdr_packets; int vorbis; int theora; + int flac; } ogg_stream_t; typedef struct ogg_demuxer { @@ -362,6 +363,11 @@ } } #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 @@ 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) ) {
--- a/libmpdemux/extension.c Sat Oct 04 20:05:54 2003 +0000 +++ b/libmpdemux/extension.c Sat Oct 04 22:00:25 2003 +0000 @@ -36,6 +36,8 @@ { "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 },