- mad-user - MARS Mailing Lists

Trying to use the MAD encoder on a SAM3S
by Simon Preuß 23 Jun '15

23 Jun '15

Hello there, I want to build a simple mp3 player. Because I have access to a SAM3S EK2 evaluation board and a SAM-ICE debugger I want to try and use the MAD encoder on a SAM3S (in order to later build my own board with a SAM3S and a proper 16 bit stereo audio DAC). Now while I do have some experience with programming the SAM3S using modules from the Atmel Software Framework, I have not used third party code before. If anyone here would be so kind and share their expertise with me to help me get this show on the road, I would very much appreciate it! This is what (little) I got so far: I have added all the MAD sources to my project (using Atmel Studio) and included the mad.h in my main source file. Looking through the sources, I noticed that in the mad.h, line 27 I should probably change FPM_INTEL to FPM_ARM. When I try to compile, I get an error in fixed.h saying that no FPM was defined. So I added #ifndef FPM_ARM #define FPM_ARM #endif #ifndef SIZEOF_INT #define SIZEOF_INT 4 #endif at the beginning of fixed.h (because after adding only the define for FPM_ARM, I got another error concerning SIZEOF_INT). Now when I compile, I get a lot of errors in inimdct_l_arm.s, most of them 'lo register required'. I have next to no experience with assembler programming so this leaves me with a large question mark over my head. What steps do I have to take to make the encoder available on my ARM? I hope I am not trying something completely stupid and please excuse any ignorance on my part. If anyone out there would be willing to help me, that would be awesome! Best regards, Simon

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Build failed on MacOS X Maverick
by Raymond Blum 17 Dec '13

17 Dec '13

Hi all I'm trying to build madlib under Mavericks (OS 10.9) using XCode 5.02 and after .configure appears to succeed, I am getting the following error from make *error: **unknown target CPU 'i486'* The compile cmd line from make is: gcc -DHAVE_CONFIG_H -I. -I. -I. -DFPM_INTEL -DASO_ZEROCHECK -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -MT version.lo -MD -MP -MF .deps/version.Tpo -c version.c -fno-common -DPIC -o .libs/version.o I think that I see what the problem us on the surface, .configure is generating a machine arch of i486 but gcc does not know that target type, less clear is why and what to do about it. Help appreciated! ---Raymond --- Empathy is the realization that another creature is self aware

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Failed to Compile lib-mad from source
by Anshul 23 Jul '13

23 Jul '13

-------- Original Message -------- Subject: Failed to Compile lib-mad from source Date: Tue, 23 Jul 2013 14:47:44 +0530 From: Anshul <anshul.maheshwari(a)einfochips.com> To: mad-user(a)lists.mars.org -------- Original Message -------- Subject: Failed to Compile lib-mad from source Date: Tue, 23 Jul 2013 14:39:32 +0530 From: Anshul <anshul.maheshwari(a)einfochips.com> To: mad-user(a)lists.mars.org HI I was trying to compile libmad-0.15.0b or libmad-0.15.1b, both of them failed on my computer. Actually -fforce-mem was deprecated option to gcc and has been removed, so I feel There is need to update configure.ac error was (sed -e '1s|.*|/*|' -e '1b' -e '$s|.*| */|' -e '$b' \ -e 's/^.*/ *&/' ./COPYRIGHT; echo; \ echo "# ifdef __cplusplus"; \ echo 'extern "C" {'; \ echo "# endif"; echo; \ if [ ".-DFPM_INTEL" != "." ]; then \ echo ".-DFPM_INTEL" | sed -e 's|^\.-D|# define |'; echo; \ fi; \ sed -ne 's/^# *define *$HAVE_.*_ASM$.*/# define \1/p' \ config.h; echo; \ sed -ne 's/^# *define *OPT_$SPEED\|ACCURACY$.*/# define OPT_\1/p' \ config.h; echo; \ sed -ne 's/^# *define *$SIZEOF_.*$/# define \1/p' \ config.h; echo; \ for header in version.h fixed.h bit.h timer.h stream.h frame.h synth.h decoder.h; do \ echo; \ sed -n -f ./mad.h.sed ./$header; \ done; echo; \ echo "# ifdef __cplusplus"; \ echo '}'; \ echo "# endif") >mad.h make all-recursive make[1]: Entering directory `/home/anshul/work/libmad-0.15.0b' make[2]: Entering directory `/home/anshul/work/libmad-0.15.0b' if /bin/sh ./libtool --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -DFPM_INTEL -DASO_ZEROCHECK -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -MT version.lo -MD -MP -MF ".deps/version.Tpo" \ -c -o version.lo `test -f 'version.c' || echo './'`version.c; \ then mv -f ".deps/version.Tpo" ".deps/version.Plo"; \ else rm -f ".deps/version.Tpo"; exit 1; \ fi mkdir .libs gcc -DHAVE_CONFIG_H -I. -I. -I. -DFPM_INTEL -DASO_ZEROCHECK -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -MT version.lo -MD -MP -MF .deps/version.Tpo -c version.c -fPIC -DPIC -o .libs/version.lo gcc: error: *unrecognized command line option '-fforce-mem'* make[2]: *** [version.lo] Error 1 make[2]: Leaving directory `/home/anshul/work/libmad-0.15.0b' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/anshul/work/libmad-0.15.0b' make: *** [all] Error 2 I am trying to compile by removing -fforce-mem from configure.ac any better idea are welcomed. Thanks Anshul "Sorry if my sentence look arrogant, my english is bad" HI I tried to remove *-fforce-mem* and apply autoreconf but by that my bash itself crash. I am using OpenSuse 12.3 linux 3.7.10-1.16-pae proccessor Intel thanks Anshul HI I was able to compile source code after removing *-fforce-mem *I have to to some additional steps other than just removing *-fforce-mem *I had to run libtoolize before i could complile, because only autoreconf does not update libtool, I dont Know what version You people use but for me i required macro directory * confure.ac Change *32c32 < --- > AC_CONFIG_MACRO_DIR([m4]) 143d142 < optimize="$optimize -fforce-mem" * Makefile.am Change *23a24 > ACLOCAL_AMFLAGS = -I m4 * Please help me if there is any other way than this work around. *Thanks Anshul --------------------------------------------------------------------------------------------- Notice: This message has been scanned by Trend Micro Mail Security scanner and is believed to be clean ---------------------------------------------------------------------------------------------

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libmad cache
by jon 26 Apr '13

26 Apr '13

Hi, i'm trying to run the code on a empeg player. All seems to work ok, but I need to modify the sources a bit and I have to admit I don't fully (ok, mostly at all !) understand it. What I need to do is get madlib to read a fixed size chunk of mp3 data, then when playback is N seconds before the end of that chunk I want it to read the next chunk. IE I am trying read ahead to work round a problem with disk spin up time on the device. Strictly speaking the code does not need a cache, it can do the read twice and throw the data from the first one away - all that matters is that playback does not stop while the disk spins up, IE read of the real data must never block :-) I'm using a slightly modified madplayer at the moment linked against an unmodified libmad. Any advice? Thanks, Jon

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estimate of output size?
by Chris Korda 27 Feb '13

27 Feb '13

Hi, Is there any method to obtain the size of the expected output from a given MP3 file? Even an estimate would be better than nothing. I need to know the size in order to display a progress bar, and also performance will improve considerably if I can allocate the destination array in advance instead of repeatedly growing it. I'm using libmad in Windows XP/Vista/7 with Visual Studio versions 6.0 and 9.0. Cheers, Chris Korda http://waveshop.sourceforge.net/

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Issues installing MAD
by Ron Karroll 22 Feb '12

22 Feb '12

Quick question, has anyone else encountered the issue when installing MAD hat it can't find the libid3tag library even though you've already installed it? This occurs even after using setting LDFLAGS to look in the proper directory. Any suggestions? Sent from my iPad

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Fwd: madplay
by Neil Jones 22 Sep '10

22 Sep '10

Hi, Im trying to play mp3's on a new arch using madplay and alsa. If I pipe the output of madplay into alsa it works fine eg: madplay -R48000 -o wav:- file.mp3 | aplay -cc but if I invoke it directly: madplay - R48000 file.mp3 it sounds horrible, like its clipping and there lots of white/pinknoise mixed in. I have tried attenuating the output (-a -10) and check alasa-mixer settings that doesn't help. I also get the same effect when playing audio via mplayer (using libmad) I have no FPU so cant try mpg123 for instance. Any help or hints welcome. Cheers Neil

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MinGW-w64 Compatibility
by Hoyt, David 18 Sep '10

18 Sep '10

Hello, I'm compiling libid3tag w/ 32-bit GCC 4.5.1 from MinGW-w64 on Windows 7 64-bit and the shared lib build fails due to missing symbols. If you inspect it a bit closer, it fails b/c it's trying to link with a main function (WinMain@16) somewhere and since it's a shared lib, there is none. The following allowed the shared lib to compile and install: make "libid3tag_la_LIBADD=-no-undefined -shared" I'm just letting you know in case you know how/where to make the fix (sorry, but I don't know). Thanks, - David Hoyt

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mad_frame_decode and libao
by Adam Magaluk 27 May '10

27 May '10

Hello List, Im trying to get my program running, im decoding an mp3 frame by frame with mad_frame_decode. Ive got it to the point where it just outputs the decoded mp3 to console. But thats not very helpful how would I hook it up to libao ao_play or anyother output library?? Running on Linux Debian x86 Thanks for the help

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Problem with madplay installation
by William Goldsmith 13 Dec '09

13 Dec '09

Hi there, I have successfully (I think) installed libmad-015.1b and libid3tag-0.15.1b. But when I try to install configure exits with an error: id3tag.h was not found you must first install libid3tag before you can build this package I'm confused. Any help would be appreciated. I'm running CentOS 5 thanks -bg ----- Bill Goldsmith Radio Paradise www.radioparadise.com

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Question about madplay, libmad and libid3tag
by Kiran Biswal 28 Sep '09

28 Sep '09

Hi All , I am new user of MAD. I am trying to compile MAD on my platform (MIPS based). When I try to run the ./confogure from commandline it prompts me to point to the libmad and libid3tag which are in separate directories, so I give path to both libmad and libid3tag like below from the directory where madplay-0.15.2b is kept ./configure CPPFLAGS=-I/root/Music/audioplayer/libmad/libmad-0.15.1b CPPFLAGS=-I/root/Music/audioplayer/libid3tag/libid3tag-0.15.1b LDFLAGS=-L/root/Music/audioplayer/libmad/libmad-0.15.1b LDFLAGS=-L/root/Music/audioplayer/libid3tag/libid3tag-0.15.1b But it still complains that it can not find mad.h and I must first install libmad . Could you please tell me how to point to libmad and lib3tag at the same time? Thanks in advance for the help Regards Kiran

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id3tag pkgconfig: -zf
by hollunder＠gmx.at 24 Mar '09

24 Mar '09

Hi there, I'm investigating an error when attempting to build sonic visualiser 1.5 and we narrowed it down to the -zf option in the pkgconfig. Is this a bug in libid3tag or in my distro? This is the error at compiletime: g++ -Wl,-O1 -Wl,-rpath,/usr/lib -o sonic-visualiser tmp_obj/main.o tmp_obj/OSCHandler.o tmp_obj/MainWindow.o tmp_obj/PreferencesDialog.o tmp_obj/moc_MainWindow.o tmp_obj/moc_PreferencesDialog.o tmp_obj/qrc_sonic-visualiser.o -L../view -L../layer -L../data -L../widgets -L../transform -L../plugin -L../base -L../system -L../framework -L../audioio -L../rdf -L/usr/lib -lsvframework -lsvaudioio -lsvview -lsvlayer -lsvrdf -lsvtransform -lsvwidgets -lsvdata -lsvplugin -lsvbase -lsvsystem -lbz2 -L/home/hollunder/builds/vamp-plugin-sdk/pkg/usr/lib -lvamp-hostsdk -loggz -lfishsound -lvorbisenc -lspeex -lvorbis -logg -lmad -zf -lid3tag -lrubberband -lfftw3 -lfftw3f -lsndfile -lsamplerate -llrdf -lrdf -lrasqal -lraptor -lportaudio -ljack -llo -Wl,-Bdynamic -lQtXml -L/usr/lib -pthread -pthread -lQtGui -L/usr/X11R6/lib -pthread -lpng -lfreetype -lgobject-2.0 -lSM -lICE -pthread -pthread -lXrender -lfontconfig -lXext -lX11 -lQtNetwork -lssl -lcrypto -pthread -pthread -lQtCore -lz -lm -pthread -lgthread-2.0 -lrt -lglib-2.0 -lpthread -ldl g++: unrecognized option '-zf' This is the pkgconfig file including the -zf prefix=/usr exec_prefix=/usr/bin libdir=/usr/lib includedir=/usr/include Name: ID3TAG Description: libid3tag - ID3 tag manipulation library Version: 0.15.0b Libs: -L${libdir} -lid3tag -zf Cflags: Thanks in advance for your help

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libmad-0.15.1b does not compile on Fedora 9
by Joao Moreira 13 Oct '08

13 Oct '08

Hi, [Posted this on mad-dev some days ago, got no answer...] I've just installed a clean Fedora 9 distro, and I tried to compile libmad-0.15.1b (which I had done fine previously, under Fedora 5), I got the following error : gcc -DHAVE_CONFIG_H -I. -I. -I. -DFPM_INTEL -DASO_ZEROCHECK -Wall -march=i486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -MT version.lo -MD -MP -MF .deps/version.Tpo -c version.c -fPIC -DPIC -o .libs/version.o cc1: error: unrecognized command line option "-fforce-mem" make[2]: *** [version.lo] Error 1 gcc version 4.3.0 does not have the "-fforce-mem" option... what can I do ? is there a way (a configure option) to disable this ? Thx, Joao -- Joao Moreira de Sa Coutinho joao.moreira(a)free.fr Tel.: 08.70.23.48.29 Mob.: 06.28.90.83.67

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libmad newbie questions
by Alan Wolfe 28 Jun '08

28 Jun '08

Heya! I was wondering; since mp3 is a closed format that you need to pay royalties for, how does libmad get around this? Also I was looking at the libmad page and noticed it said this under licensing: "MAD is available under the terms of the GNU General Public License, Version 2 <http://www.underbit.com/resources/license/gpl/>, for either permanent use or for evaluation prior to obtaining a commercial license." Just to be clear, if you use libmad commercially you can either do so with the GPL license (satisfying that license) or you can contact the owners of the library to talk about paying for a non GPL commercial license right? I am looking for a way to convert mp3 to wav in a product that is going to be released commercially and libmad really seems like the perfect fit, I just want to make sure I know what's up with licensing before commiting to this path. Thanks a bunch!

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Stream Decode
by Alexey V Paramonov 21 May '08

21 May '08

Hi! I have the following question - is it possible to decode a stream using MAD? For example: I have shoutcast stream and I read it in small portions (say, 40960 bytes), decode it and encode back to mp3 using LAME (chaning bitrate, samplerate, etc...). The problem is that I have to call mad_stream_buffer with each piece of stream data, decode/encode it and then if I listen the resulting mp3 file - I notice "noise" between theese small decoded pieces. Seems like it happen not only with VBR files. Pseudo-code: while ( bytes = read(file_id, buff, SHOUT_BUFFER_LENGTH ) ) { mad_stream_buffer(&stream, buff, bytes ); while(1){ mad_frame_decode(&frame,&stream) .... int ret = lame_encode_buffer( lameGlobalFlags, left, right, nsamples, mp3Buf, mp3Size ); // Play resulting mp3 data } } Is it possible to decode a stream without theese "edge effects" ? May be I should not re-initialize MAD buffer after each read in some way? Please help.

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decoding error while using minimad.c
by 王斐斐 21 Apr '08

21 Apr '08

Hello ! I just got started using libmad. I need to write an application to decode an mp3 file.I tried to run minimad.c, some errors like that: decoding error 0x0101(lost synchronization)at byte offset 0 decoding error 0x0101(lost synchronization)at byte offset 5539905 my soundcard info: Multimedia audio controller: Ensoniq ES1371 [AudioPCI-97] (rev 02) Subsystem: Ensoniq Creative Sound Blaster AudioPCI64V, AudioPCI128 Flags: bus master, ?? devsel, latency 64, IRQ 18 I/O ports at 1400 [size=64] my kernal system( running in VMware workstation) info: 2.6.15-1.2054_FC5 i686 athlon i386 something in /etc/modprobe.conf is:alias eth0 pcnet32 alias scsi_hostadapter BusLogic alias snd-card-0 snd-ens1371 options snd-card-0 index=0 options snd-ens1371 index=0 remove snd-ens1371 { /usr/sbin/alsactl store 0 >/dev/null 2>&1 || : ; }; /sbin/modprobe -r --ignore-remove snd-ens1371 Can anyone please help me? Thank you!! feifei.wang ___________________________________________________________ 雅虎邮箱，您的终生邮箱！ http://cn.mail.yahoo.com/

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Minimad.c
by Sebastiaan Heunis 27 Feb '08

27 Feb '08

Hi I just got started using libmad. I need to write an application to decode an mp3 and write the contents to a data file for FM modulation in Gnuradio. I tried to compile minimad.c using g++ -c minimad.c -lmad, but I can't get it to work. I get the following errors. Can anyone please help me? minimad.c: In function ��int main(int, char**)��: minimad.c:56: error: invalid conversion from ��void*�� to ��const unsigned char*�� minimad.c:56: error: initializing argument 1 of ��int decode(const unsigned char*, long unsigned int)�� minimad.c: In function ��mad_flow input(void*, mad_stream*)��: minimad.c:87: error: invalid conversion from ��void*�� to ��buffer*�� minimad.c: In function ��mad_flow error(void*, mad_stream*, mad_frame*)��: minimad.c:175: error: invalid conversion from ��void*�� to ��buffer*�� Thank you Sebastiaan

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madplay: How to select a specific ALSA card
by Thomas Reitmayr 03 Jan '08

03 Jan '08

Hi, I just started using madplay a short while ago and I was wondering whether it is possible to select an ALSA card different from the default one for audio output. >From looking at the code it seems like the PATH part of --output is used, however madplay --output=alsa:plughw:1 myfile.mp3 chokes as the TYPE alsa does not exist (in audio.c). Did I miss something in the command line or is this a bug / missing feature (for which I could provide a patch)? Thanks, -Thomas

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dual-thread libmad
by Steve Spano 09 Dec '07

09 Dec '07

I have made a lot of progress working with libmad and trying to get two instances running. I have a simple RTOS which has two threads. Each thread starts lib mad with a dedicated mad_decoder buffer for each thread. I am now able to "read" data correcting into the decoders (i.e. I don't see decoding errors streaming everywhere on the debug trace) but the output audio is mixed up. Meaning that the audio from the two threads is being mixed (actually shuffled) during playback. Seems like there may be another global buffer being used somewhere that I have to split out onto two threads or something. Can anyone help with ideas on how to have two threads, play two different MP3s, using libmad? Thanks Steve Spano, President Finger Lakes Engineering

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hd audio
by Regina Davis 29 Nov '07

29 Nov '07

Hi, I am currently using madplay functions to play MP3 voice messages in our product that runs on a PC motherboard. This is in an RTOS embedded environment not windows. Recently the motherboards have a high definition audio chip on them instead of the ac97. Is there support for this? What are my options? Currently the initialization is returning the device not found error. Thanks, Regina ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________

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mad decoder sample rates
by Steve Spano 07 Sep '07

07 Sep '07

Hi Folks, We have the mad library running on the new STR912 processor; 96MHz ARM9 core. We are using a small CPLD to bridge between the SSP serial port and a CS4431 stereo dac. We have an I2C silicon oscillator being used so we can set any sample rate we want. Currently, its working at 22.05KHz. We haven't change the sample rate yet. I want to be able to run 8Khz to 48KHz MP3s. Is there anything in the mad decoder that would prevent decoding MP3s within this range? We don't have a sample issue with our hardware since we can set any sample rate we want. Thanks Steve Spano, President steve(a)fl-eng.com Finger Lakes Engineering 607-277-1614 x223 www.fl-eng.com <http://www.fl-eng.com/>

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Compiling LIBMAD for ARM
by Karthika 02 Apr '07

02 Apr '07

Helloo, I am trying to compile LIBMAD for ARM7TDMI on Metrowerks Codewarrior tool. The processor specific options for ARM are present in the file fixed.h. I have taken those options which are specific for ARM and included them in a new header file called "asm.h" ( given below).I have included this "asm.h" file in my project. Then, I have removed fixed.h from my project. Now, when I compile my project I get the errors for mad_f_mul and MAD_F_MLA. MAD_F_MLA error is only for III_alias reduce( ) function. A few errors are given below. Error : #29: expected an expression layer3.c line 934 return frac ? mad_f_mul(requantized, root_table[3 + frac]) : requantized; Error : #18: expected a ")" layer3.c line 1591 MAD_F_MLA(hi, lo, -b, ca[i]); Error : #18: expected a ")" layer3.c line 1596 MAD_F_MLA(hi, lo, a, ca[i]); Error : #29: expected an expression layer3.c line 1647 m0 = mad_f_mul(a17, -c3); The "asm.h" file I have included is as follows. # define MAD_F_SCALEBITS 28 # define MAD_F_FRACBITS 28 # define MAD_F(x) ((mad_fixed_t) (x##L)) # define MAD_F_MIN ((mad_fixed_t) -0x80000000L) # define MAD_F_MAX ((mad_fixed_t) +0x7fffffffL) # define MAD_F_ONE MAD_F(0x10000000) # define mad_f_tofixed(x) ((mad_fixed_t) \ ((x) * (double) (1L << MAD_F_FRACBITS) + 0.5)) # define mad_f_todouble(x) ((double) \ ((x) / (double) (1L << MAD_F_FRACBITS))) # define mad_f_intpart(x) ((x) >> MAD_F_FRACBITS) # define mad_f_fracpart(x) ((x) & ((1L << MAD_F_FRACBITS) - 1)) /* (x should be positive) */ # define mad_f_fromint(x) ((x) << MAD_F_FRACBITS) # define mad_f_add(x, y) ((x) + (y)) # define mad_f_sub(x, y) ((x) - (y)) # define mad_f_mul(x, y) \ ({ mad_fixed64hi_t __hi; \ mad_fixed64lo_t __lo; \ mad_fixed_t __result; \ asm ("smull %0, %1, %3, %4\n\t" \ "movs %0, %0, lsr %5\n\t" \ "adc %2, %0, %1, lsl %6" \ : "=&r" (__lo), "=&r" (__hi), "=r" (__result) \ : "%r" (x), "r" (y), \ "M" (MAD_F_SCALEBITS), "M" (32 - MAD_F_SCALEBITS) \ : "cc"); \ __result; \ }) # define MAD_F_MLX(hi, lo, x, y) \ asm ("smull %0, %1, %2, %3" \ : "=&r" (lo), "=&r" (hi) \ : "%r" (x), "r" (y)) # define MAD_F_MLA(hi, lo, x, y) \ asm ("smlal %0, %1, %2, %3" \ : "+r" (lo), "+r" (hi) \ : "%r" (x), "r" (y)) # define MAD_F_MLN(hi, lo) \ asm ("rsbs %0, %2, #0\n\t" \ "rsc %1, %3, #0" \ : "=r" (lo), "=r" (hi) \ : "0" (lo), "1" (hi) \ : "cc") # define mad_f_scale64(hi, lo) \ ({ mad_fixed_t __result; \ asm ("movs %0, %1, lsr %3\n\t" \ "adc %0, %0, %2, lsl %4" \ : "=&r" (__result) \ : "r" (lo), "r" (hi), \ "M" (MAD_F_SCALEBITS), "M" (32 - MAD_F_SCALEBITS) \ : "cc"); \ __result; \ }) mad_fixed_t mad_f_abs(mad_fixed_t); mad_fixed_t mad_f_div(mad_fixed_t, mad_fixed_t); Can anybodu help me? Thanks, Karthika Lakshmi. __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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Synthesis filter bank in LIBMAD
by Karthika 28 Mar '07

28 Mar '07

Helloo, I am trying to understand the synthesis filter bank in the LIBMAD. I could not understand in what order the D coefficients are used to obtain the PCM samples. Also,half of these coefficients have some relation to the other half. Then, why all the coefficeints are being used in D.dat file? What is the significance of even and odd pointers? How is the windowing operation between U vector and D vector taking place? Also, I would like to know if we can further optimize the LIBMAD source code. For e.g, by using Cooley-Tukey algorithm for the calculation of fastdct() and dct32() . Could anyone suggest any further optimization techniques for porting LIBMAD on an ARM processor? Any help will be greatly appreciated. Thanks and Regards, Karthika Lakshmi. __________________________________________________________ Yahoo! India Answers: Share what you know. Learn something new http://in.answers.yahoo.com/

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Help :)
by Khalid Aallouche 09 Dec '06

09 Dec '06

Hi all, I wanted to decode an MP3 file in order to read it. But I'm novice and there is so much file...Is someone can help to create a basic MP3 player (console application:user enters where the MP3 is, and after read the valid MP3 file) Thank you very much Khalid

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madplay vs. mpg123
by Aaron 05 Dec '06

05 Dec '06

I ran more tests. madplay is fine on Mac. I used both madplay and mpg123 to decompress a file to wav. The wav sounds the same on both outputs. It sounds fine from aplay. It sounds bad (again) from audacity. It seems that on this machine, different programs/device drivers yield very different results. Any ideas what is different between these programs? (which drivers to they use... /dev/dsp? some alsa driver?) Perhaps there is some ioctl or something that isn't being done properly for my card on some programs... aaron

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Playing Internet Streams with Madplay
by Mark Kane 12 Oct '06

12 Oct '06

Hi everyone. This may be a stupid question, but I've read through the madplay man page and online documentation several times and I cannot get an MP3 Internet radio station (SHOUTcast) to play with madplay. Every time I try on the command line, I get: ------------------------------------------------------ [mixx941@amd64:~]% madplay http://stream.com:8000 MPEG Audio Decoder 0.15.2 (beta) - Copyright (C) 2000-2004 Robert Leslie et al. http://stream.com:8000: No such file or directory ------------------------------------------------------ I've tried with and without the trailing slash on the URL, with/without quotes on the URL, and adding the "listen.pls" at the end. Still the same. Tried downloading the .pls and playing it from the file, although that gives this: ------------------------------------------------------ [mixx941@amd64:~]% madplay listen.pls MPEG Audio Decoder 0.15.2 (beta) - Copyright (C) 2000-2004 Robert Leslie et al. error: frame 0: lost synchronization 0 frames decoded (0:00:00.0), -inf dB peak amplitude, 0 clipped samples [mixx941@amd64:~]% ------------------------------------------------------ What am I doing wrong? Thanks in advance. -Mark madplay-0.15.2b -- Internet Radio: Party107 (Trance/Electronic) - http://www.party107.com Rock 101.9 The Edge (Rock) - http://www.rock1019.net IRC: MIXXnet IRC Network - irc.mixxnet.net (Nick: MIXX941)

3 3

Decoding time of MAD
by benix samuel 12 Oct '06

12 Oct '06

Hi everyone , I am using MAD decoder. It takes different time to decode two different frames of same header information and frame length. Is that MAD decoding time even depends on the data in the frame or only on the frame size. Benix

2 1

Where the hell is definition of struct id3_file????
by tj 29 Sep '06

29 Sep '06

In trying to learn how to use libid3, since there sems to be no documentatio. I am looking at soure trying to figure out how to use it. The first problem is the practically every call uses a pointer to a "struct id3_file" . Now, the problem is that there is no definition for id3_file structure in any .h file. I finally found it in file.c, a source file. How the hell are you supposed to use it in your app when the prototype is in a package source file? Seems that something is really broken here. tj

3 4

Madplay & Psion 5 MX: works 95% :-)
by Michael Fritscher 11 Sep '06

11 Sep '06

Hi, first of all: Thanks for the great program :-) madplay works ca. 95% on the psion 5 mx: it would need ca. 5-10% more power... It's a 36,8 Mhz ARM, the sound chip can do 8kHz/16 bit. I used the parameter --downscale, which scales 44.1 kHz mp3s down to 22.1 kHz, which helps much. But ideal would be a --downscale4 or so, which would scale it down to 11.05 kHz... How can I add this? I don't find the location in the code :-( Thanks, Michael Fritscher

1 0

Gapless decoding with LAME tag reading
by Tobias Rundström 11 Aug '06

11 Aug '06

Hello List, I am trying to get our mad decoder in XMMS2 to playback LAME encoded files gapless by reading the decoder delay in the header, but I have some problems to get it working 100%. The test cases I have is confirmed playing gapless in foobar2000 so I guess I just doing something wrong :-) I parse the LAME tag with more or less the same code as in madplay.c and then I do: data->samples_to_skip = lame->start_delay; data->samples_to_skip_end = lame->end_padding; data->samples_to_play = (guint64)(((guint64)xmms_xing_get_frames (data->xing) * (guint64)1152) - (guint64)lame->start_delay); XMMS_DBG ("Samples to skip in the beginning: %d, total: %lld", data- >samples_to_skip, data->samples_to_play); These flags are then used in my decoding function which is attached at the bottom. Still I get around 1000 samples of silence in the beginning when I watch the output wave in a wave editor. My theories are that: 1) maybe MAD is adding silence to the output, I read something like hinted towards this on the interwebb somewhere. 2) that my find first sync code is not working properly. I feed the first frame (that I find the Xing/LAME tag in) back to the decoding routine found below, can that produce silence also? 3) My math is off somewhere :-) Any help on getting this play "perfect" gapless would be very appreciated! The full code is available here: http://git.xmms.se/? p=xmms2-devel.git;a=tree;;f=src/plugins/mad -- Tobias static gint xmms_mad_read (xmms_xform_t *xform, gpointer buf, gint len, xmms_error_t *err) { xmms_mad_data_t *data; xmms_samples16_t *out = (xmms_samples16_t *)buf; gint ret; gint j; gint read = 0; data = xmms_xform_private_data_get (xform); j = 0; while (read < len) { /* use already synthetized frame first */ if (data->synthpos < data->synth.pcm.length) { if (data->samples_to_skip <= 0 && data->samples_to_play >= 0) { out[j++] = scale_linear (data->synth.pcm.samples[0] [data->synthpos]); if (data->channels == 2) { out[j++] = scale_linear (data->synth.pcm.samples [1][data->synthpos]); read += 2 * xmms_sample_size_get (XMMS_SAMPLE_FORMAT_S16); data->samples_to_play -= 2; } else { read += xmms_sample_size_get (XMMS_SAMPLE_FORMAT_S16); data->samples_to_play --; } } else { if (data->channels == 2) { data->samples_to_skip -= 2; } else { data->samples_to_skip --; } } data->synthpos++; continue; } /* then try to decode another frame */ if (mad_frame_decode (&data->frame, &data->stream) != -1) { mad_synth_frame (&data->synth, &data->frame); data->synthpos = 0; continue; } /* if there is no frame to decode stream more data */ if (data->stream.next_frame) { guchar *buffer = data->buffer; const guchar *nf = data->stream.next_frame; memmove (data->buffer, data->stream.next_frame, data->buffer_length = (&buffer[data- >buffer_length] - nf)); } ret = xmms_xform_read (xform, (gchar *)data->buffer + data- >buffer_length, 4096 - data->buffer_length, err); if (ret <= 0) { return ret; } data->buffer_length += ret; mad_stream_buffer (&data->stream, data->buffer, data- >buffer_length); } return read; }

1 0

Announce: MAD now included in the Comprehensive Source Database (CSDB)
by Enrico Weigelt 03 Aug '06

03 Aug '06

Hi folks, I'd like you to know that the MAD packages are now included in the comprehensive source database (CSDB): http://sourcefarm.metux.de/ regards, -- --------------------------------------------------------------------- Enrico Weigelt == metux IT service phone: +49 36207 519931 www: http://www.metux.de/ fax: +49 36207 519932 email: contact(a)metux.de cellphone: +49 174 7066481 --------------------------------------------------------------------- -- DSL ab 0 Euro. -- statische IP -- UUCP -- Hosting -- Webshops -- ---------------------------------------------------------------------

1 0

reading from a from a given time
by tarek allaoua 26 Jul '06

26 Jul '06

Hi all, I want to know if the feature of reading an mp3 file from a given time is already implemented ? if not how hard would be to make it possible. Best regards. T

1 0

Compiling madplay on arm-linux
by Shinta L. 17 Jul '06

17 Jul '06

Hi... I have problem compiling madplay for arm-linux.... I used the arm-linux-gcc 3.2.3, compiled the libmad and libid3tag successfully, but when i compile madplay, i received the following message: ----- compile message starts here ----- checking for id3_tag_parse in -lid3tag... no configure: error: libid3tag was not found *** You must first install libid3tag before you can build this package. *** If libid3tag is already installed, you may need to use the LDFLAGS *** environment variable to specify its installed location, e.g. -L<dir>. ----- compile message ends ----- i got the libid3tag.so.0 at the right folder but it still can't find the file.... If someone knows how to solve the problem, please give me some advise. Thanks. Regards, Shinta

2 1

Sample rate
by tarek allaoua 10 Jul '06

10 Jul '06

Hi, can mad decode mp3 files with samplerate other then 44.1 Khz or 22.05 Khz ? can any one explin me tis behaviour plz. Best regards. Tarek.

2 3

compile error on linux
by Yamamoto Kazuko 09 Jun '06

09 Jun '06

I have problem in compiling madplay on linux(RedHat Linux7.3). I've found similar problem in this archive, but it didn't help. 1. I downloaded libid3tag-0.15.1b.tar.gz, libmad-0.15.1b.tar.gz, madplay-0.15.1b.tar.gz 2. I configured, maked and installed libid3tag and libmad, which were successful. I also confirmed that there are mad.h and id3tag.h under /usr/local/include. 3. I tried to make madplay, then I got following messages. -----compile messages starts here gcc -Wall -march=i486 -g -O2 -pg -pg -o madplay madplay.o getopt.o getopt1.o version.o resample.o filter.o tag.o crc.o player.o audio.o audio_aiff.o audio_cdda.o audio_hex.o audio_null.o audio_raw.o audio_snd.o audio_wave.o audio_esd.o audio_oss.o /usr/lib/libesd.so -L/usr/lib -laudiofile -lmad -lid3tag -lm resample.o: In function `resample_init': /home/kazu/mad/madplay-0.15.1b/resample.c:46: undefined reference to `mad_f_div' player.o: In function `show_id3': /home/kazu/mad/madplay-0.15.1b/player.c:644: undefined reference to `id3_frame_field' /home/kazu/mad/madplay-0.15.1b/player.c:678: undefined reference to `id3_frame_field' /home/kazu/mad/madplay-0.15.1b/player.c:684: undefined reference to `id3_frame_field' player.o: In function `process_id3': /home/kazu/mad/madplay-0.15.1b/player.c:1232: undefined reference to `id3_frame_field' /home/kazu/mad/madplay-0.15.1b/player.c:1278: undefined reference to `id3_frame_field' /home/kazu/mad/madplay-0.15.1b/player.c:1278: undefined reference to `id3_field_getlatin1' /home/kazu/mad/madplay-0.15.1b/player.c:1279: undefined reference to `id3_frame_field' /home/kazu/mad/madplay-0.15.1b/player.c:1342: undefined reference to `id3_frame_field' /usr/lib/libid3tag.a(util.o): In function `id3_util_compress': util.o(.text+0x12f): undefined reference to `compress2' /usr/lib/libid3tag.a(util.o): In function `id3_util_decompress': util.o(.text+0x196): undefined reference to `uncompress' collect2: ld returned 1 exit status make[2]: *** [madplay] エラー 1 make[2]: 出ますディレクトリ `/home/kazu/mad/madplay-0.15.1b' make[1]: *** [all-recursive] エラー 1 make[1]: 出ますディレクトリ `/home/kazu/mad/madplay-0.15.1b' make: *** [all] エラー 2 ----compile message end If someone know the solution, please give me advise. Thank you in advance, Kazu

2 2

MAD/ARM archeology..
by john cooper 14 Apr '06

14 Apr '06

Hello, I've been lugging around a 0.13 version of libmad for about 5 years for use in a back burner project. I've ported the works to PPC about a year ago and just today to ARM. With 0.13 libmad there seems to be some really funky decode artifacts in libmad 0.13/ARM (but not in 0.15.1) which I couldn't explain. I noticed a few ARM-specific changes in fixed.h between 0.15.1 and 0.13 as well as what appear to be some tweaks called out in the CHANGES file. But the echo-like decode artifacts were so dramatic I can't believe they would have gone unnoticed nor unaddressed. I suppose it could be some odd build issue local to my work but that seems rather remote. Anyone encounter any similar issue specific to ARM in or around that vintage? Thanks, -john -- john.cooper(a)third-harmonic.com

1 0

Stayalive Wrapper
by Matthew Rubenstein 19 Mar '06

19 Mar '06

I listen to a stream that plays data retrieved by curl piped to madplay, on my Inspiron8000's Ubuntu GNOME desktop: #!/bin/sh # Clean out old sessions. killall curl madplay sleep 2 # WWOZ.org stream curl http://wwoz-sc.streamguys.com:80 | madplay -Q - Every once in a while the curl process is interrupted by a network timeout. I want to put the script into a loop that tests the playback status, and restarts the pipeline if it's not playing. Any ideas? -- (C) Matthew Rubenstein

1 0

Documentation
by MUNDOALEM.com 14 Mar '06

14 Mar '06

I'm starting to code with audio now. I'm trying to code a basic mp3 player just for studying purposes. I found libmad interesting to my purposes and would like to learn how to use it on my ANSI C programs. But I couldn't find any documentation of it. So if you could send me some links over the internet it would be pretty good!! Thank you very much you all. Egon Braun

1 0

Suitability of use for Embedded Project
by Doug Allender 07 Mar '06

07 Mar '06

I am looking for source code for a no operating system, streamed audio receiver. I am told the stream is MPEG1 Layer2. Is the MAD source suitable for this? , and have people found this easy to port?

2 1

#error "no FPM selected"
by Sonia 27 Dec '05

27 Dec '05

I'm using libmad to decode some mp3 sound. But not in a normal way, i'm using a program in windows platform to compile the code, and I get the folloing error " In file included from TestApp_Memory/src/libmad/frame.h:25, from TestApp_Memory/src/libmad/decoder.c:51: TestApp_Memory/src/libmad/fixed.h:446:3: #error "no FPM selected" " I already use this library on linux and it work ok, with the make. but now i need to integrated with a another program, microblaze, where I only have a normal gcc compiler. Do I need somme specific Flags to compile it and run it? I don't know if it is important but I dont send a file to the input now, I'm reading the streams directly from the memory. that means that the decode function receives a pointer to the mem_address. cyas Sonia Gonçalves

2 1

ask some mp3 decoder problem
by 徐生 08 Aug '05

08 Aug '05

hi: i'm try to decode the mp3 according to the 11172-3 standard,but now i have some problem,can anybody help me? 1.when i decode the huffman code,how can i do if bits need from huffman decode is more/less than the side information infer? 2.when i decode the huffman code,for example,"011001001"form bitstream,but this bitstream cannot find huffman code in huffman table,what should i do?i mean,it has search from least bits from most bits(for example,from 2bits to 9bits),excuse my poor english _________________________________________________________________ 免费下载 MSN Explorer: http://explorer.msn.com/lccn/

1 0

Decoding and playing MP3 while doing something else
by Jeroen Taverne 02 Jul '05

02 Jul '05

Hi. I am implementing a very simple MP3 player on an ARM architecture using a CF harddisk. To safe power I am reading as much data as possible in one time from the harddisk, so it will goto standby mode. What I would like to do is this: Load as much possible MP3 data from multiple files in a 24MB cyclic buffer. Start decoding and playing. Playing is done using double buffered DMA. When 16MB of data is decoded and played, 16MB is read again from harddisk, overwriting the decoded and played data. During loading of these 16MB the decoding and playing continues of the 8MB that's still in the cyclic buffer. This is repeated all the time. I can't use MAD's SYNC mode for this, because the decoder "hangs" and at some time the buffer is empty. I don't want to use multi threading, as I don't have an OS at all. So maybe I have to use ASYNC mode??? Then the decoder directly returns, so I can do some other things like disk reading while decoding. But I can't find any simple example how to use ASYNC mode. When the decoder return I need some way to "kick" the decoder again, so it does a little decoding and then returns again. I don't know how the messaging system works of MAD in ASYNC mode. Does someone have a simple example for me? Best regards, Jeroen

1 0

sample code to obtain mp3 data
by yoshita nanda 16 Jun '05

16 Jun '05

hi all I am a masters student doing project on audio classification. I need the mp3 data as uncompressed floating point values. Since i need to concentrate on classification part of projcet it is very neccessary for me to obtain a code that outputs the mp3 data. If anyone can help me with the code that includes Madlib library and outputs the mp3 data. Regards Yoshita _________________________________________________________________ Be the first to hear what's new at MSN - sign up to our free newsletters! http://www.msn.co.uk/newsletters

1 0

madplay: build problems
by Axel Böhm 18 May '05

18 May '05

hi there, i tried to build libid3tag, mad and madplay. libid3tag and mad worked fine, madplay doesn't seem to work. i did the usual ./configure, make, make check and make install for all of them, except madplay. see below for the output of the make command for madplay. my build environment is a common ubuntu installation. used versions: libid3tag_0.15.1b mad_0.14.2b madplay_0.15.2b output (until here everything worked fine): gcc -Wall -march=i486 -g -O2 -o madplay madplay.o getopt.o getopt1.o version.o resample.o filter.o tag.o crc.o rgain.o player.o audio.o audio_aiff.o audio_cdda.o audio_hex.o audio_null.o audio_raw.o audio_snd.o audio_wave.o audio_oss.o -lmad -lid3tag -lm resample.o(.text+0x1d): In function `resample_init': /home/axel/mydocs/src/madplay_0.15.2b/resample.c:46: undefined reference to `mad_f_div' collect2: ld returned 1 exit status make[2]: *** [madplay] Error 1 make[2]: Leaving directory `/home/axel/mydocs/src/madplay_0.15.2b' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/axel/mydocs/src/madplay_0.15.2b' make: *** [all] Error 2 any help is welcome. thanks in advance. axel

2 1

MAD in C# .Net (libMAD.dll)
by Sondre 15 Apr '05

15 Apr '05

Hallo! I want to make my own music player with MAD decoder in Visual Studio C# .Net. My operating system is Windows XP Pro. Have someone used libMAD.dll in C# .Net to play mp3 files? How can I play mp3 files in C# with MAD? I need help please!!! Yours faithfully Sondre Garmo

1 0

madplay watchdog signals?
by Matthew Rubenstein 23 Feb '05

23 Feb '05

Howdy - I'm running madplay on an iPaq3670/Familiar0.8 (ARM arch) with a CF ethernet card. I play Shoutcast MP3 streams by piping wget -o - <stream-url> | madplay - , and it works really well - realtime decoding, full sound, no fan, otherwise useless HW :). But every 10-30 minutes or so, I get a net/Inet hiccup that interrupts the stream connection. wget hangs, and no data gets to madplay. I want to set up another process to watch madplay, and kill it when it stops playing, then restart it - this always works manually. But madplay doesn't seem to reliably emit output status info. the --verbose flag sends an updated "seconds" counter, but, since it doesn't send any newlines, it can't be parsed by piping it to a "read" in a shell script. And redirecting it to a file doesn't reliably update the file's ctime, either. If I "madplay -o wave:- | tee >/dev/dsp |watchdog", the system performance grinds to a halt, unlistenable. I just want any technique that keeps my wget | madplay pipeline restarted all the time, with a minimum of restarts, which make an audible "clunk". Short of recoding (and cross-compiling) libmad with extra status hooks, is there any way to keep this little engine churning? -- (C) Matthew Rubenstein

2 1

Winamp MAD plugin buffer
by Andrei Tarkovsky 01 Feb '05

01 Feb '05

The original Nullsoft Winamp MP3 decoder has a option called "Full file buffering". This option loads the entire MP3 into memory rather than reading it from disc during playback. This completely eliminates skipping as well as allowing MP3s to be deleted while they are still playing back... a very useful function for cleaning up large MP3 collections. I was wondering if there is any way to achieve a similar buffer with the MAD winamp input plug. Any advice would be greatly appreciated. Cheers, Tarkovsky

1 0

Time Duration In MAD.
by Senthil Nathan G 11 Jan '05

11 Jan '05

Hi All, NewBie Question : How does one find the time duration without decoding an MP3. Is it possible. (Without looking at any header also since this might be wrong ). How do I decode a particular time period ( say from 30 sec to 45 secs ) of a given MP3 in MAD. ( or where should I look to find the answers ). Thanks very much for all help. Senthil ( INDIA )

1 0

minimad.c change sampling frequency
by bobby digital 03 Jan '05

03 Jan '05

Hello everyone! How can I change the sampling frequency from 44khz to 48khz easily in minimad.c, that is, in the highlevel api? Thanks, steels. -- ______________________________________________ Check out the latest SMS services @ http://www.linuxmail.org This allows you to send and receive SMS through your mailbox. Powered by Outblaze

3 2

S: faster decoder than libmad
by jens-lippmann＠t-online.de 02 Jan '05

02 Jan '05

Hello all! I'm looking for a MP3 decoding lib faster than libmad, probably solely written in Intel assembler. I currently use libmad 0.15.1b to stream MP3 files into WAV sound buffers and I need to improve the performance. The mere task of streaming a single audio file saps about 11 milliseconds (@1.8GHz). libmad uses inline and asm here and there, but the costliest parts (frame decoding and synthing) are written in C, optimized for speed but still C. So, my question is if there's anything freely available that decodes faster than libmad? kind regards Jens Lippmann

3 2

S: faster decoder than libmad (2)
by jens-lippmann＠t-online.de 30 Dec '04

30 Dec '04

> The mere task of streaming a single audio file > saps about 11 milliseconds (@1.8GHz). Sorry I just realized I need to correct this figure. I mean the task to feed a 176 KB/s wave stream in real-time affords frame decoding and synthing of about 44 milliseconds (per second) on a P4 1800. This means 4-5% of your application is needed to keep a single wave stream going, so I'm looking to improve that performance. kind regards Jens Lippmann

1 0

[help] support multiple instance of libmad
by Sangwoon Chung 08 Dec '04

08 Dec '04

Hello there, I'm trying to decoding and playing multiple mp3 files in a time using libmad. But it does not seem to support multiple instance of libmad. I really want to know the libmad support multiple instance ? If not, please let me know how and what should I customize libmad source to support multiple instance. I'll appreciate any Help, P.S Sorry for posting this articel for mad-dev list... Thanks a lot, Sangwoon swchung(a)rtsolutions.co.kr

2 1

Bad sync frame detection
by Hugues Lismonde 11 Nov '04

11 Nov '04

Hi, I've got a little problem with libmad. On some mp3 I have, libmad find a first frame somewhere it probably shouldn't. The problem appeared with mythtv where some files were played in mono instead of stereo and at half the speed. I made some tests with madlld and I get some strange output like this: [...] madlld: recoverable frame level error (forbidden bitrate value) madlld: recoverable frame level error (forbidden bitrate value) madlld: recoverable frame level error (forbidden bit allocation value) madlld: recoverable frame level error (forbidden bit allocation value) madlld: recoverable frame level error (forbidden bit allocation value) madlld: 25000 kb/s audio MPEG layer I stream without CRC, single channel with no emphasis at 44100 Hz sample rate madlld: recoverable frame level error (lost synchronization) madlld: recoverable frame level error (reserved header layer value) [...] I understand that "lost sync" can occur (and it does on good files too so I guess it's ok) but I can't find 25000 kb/s in bitrate_table (frame.c) so I think the "forbidden bit allocation value" errors offsets something by less than a byte. I wrote a small test program which search for sync word and try to decode the frame header without using libmad. Using it I get the following output: Frame at 504 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 508 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 512 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 516 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 520 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 532 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 536 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 540 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 544 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 548 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 560 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 564 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 568 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 572 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 576 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 588 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 592 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 596 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 600 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 604 - MPEG v1 Layer I No CRC Bad frame! (bitrate) Frame at 48330 - MPEG v1 Layer III No CRC Bitrate 192 Samplerate 44 Padding 1 Stereo No emphasis [...] It was written fast so it might be wrong, but it seems correct (the last line is consistent with iTunes or mp3blaster). I haven't found how to get the current location in the buffer from libmad so I can't tell where it get this strange frame. I'll be glad to give any precision or do any tests one might want but I'm a bit lost right now. From what I can see, libmad resync on the next frame, but for mythtv it's too late since it never rebind the channels or sample rate for the current playing song. If it's not the right list, please let me know so I can ask somewhere else. -- Hugues Lismonde < hugues.lismonde(a)hli.be >

1 0

madlld extension for proper 16bit pcm output
by jens-lippmann＠t-online.de 07 Nov '04

07 Nov '04

Hi all! Those acquainted with madlld (libMAD low-level demo) may have noticed the rather poor PCM output which comprises lots of noticeable sound dropouts. This is related to a quick'n'dirty 16bit truncation done by MadFixedToUshort. I spent several hours into gathering information about this problem and analysing the madplay package which provides better algorithms for this task. Finally I wrote a header to be included in madlld.cpp. I just want to keep you up to date with my implementation, so I've attached the header. Please drop me a note if you find it useful. Note that the copyright of the header still is retained by the madplay authors. Good bye and have fun! Jens Lippmann --- How to incorporate the header: 1. Add these lines below the include section in madlld.cpp: #include "madplay.h" struct audio_stats Audio_stats; struct audio_dither Audio_dither; //#define MadFixedToUshort MadFixedToUshortOrig #define MadFixedToUshort(a) (unsigned short)audio_linear_round(16,a,&Audio_stats) //#define MadFixedToUshort(a) (unsigned short)audio_linear_dither(16,a,&Audio_dither,&Audio_stats) 2. Rename the function MadFixedToUshort into MadFixedToUshortOrig. 3. Clear Audio_stats and Audio_dither before you call MpegAudioDecoder (the decoding loop): ZeroMemory(&Audio_stats,sizeof(Audio_stats)); ZeroMemory(&Audio_dither,sizeof(Audio_dither)); 4. Convert your MP3 stream to PCM and listen with awe. ;-> --- The header follows below (I do not want to add it as an enclosure because it might not show up in the list archive then). 8< 8< 8< 8< 8< 8< 8< 8< 8< 8< 8< 8< 8< 8< 8< /* * madplay - MPEG audio decoder and player * Copyright (C) 2000-2004 Robert Leslie * * 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 */ struct audio_stats { unsigned long clipped_samples; mad_fixed_t peak_clipping; mad_fixed_t peak_sample; }; struct audio_dither { mad_fixed_t error[3]; mad_fixed_t random; }; /* * NAME: clip() * DESCRIPTION: gather signal statistics while clipping */ static inline void clip(mad_fixed_t *sample, struct audio_stats *stats) { enum { MIN = -MAD_F_ONE, MAX = MAD_F_ONE - 1 }; if (*sample >= stats->peak_sample) { if (*sample > MAX) { ++stats->clipped_samples; if (*sample - MAX > stats->peak_clipping) stats->peak_clipping = *sample - MAX; *sample = MAX; } stats->peak_sample = *sample; } else if (*sample < -stats->peak_sample) { if (*sample < MIN) { ++stats->clipped_samples; if (MIN - *sample > stats->peak_clipping) stats->peak_clipping = MIN - *sample; *sample = MIN; } stats->peak_sample = -*sample; } } /* * NAME: audio_linear_round() * DESCRIPTION: generic linear sample quantize routine */ # if defined(_MSC_VER) extern /* needed to satisfy bizarre MSVC++ interaction with inline */ # endif inline signed long audio_linear_round(unsigned int bits, mad_fixed_t sample, struct audio_stats *stats) { /* round */ sample += (1L << (MAD_F_FRACBITS - bits)); /* clip */ clip(&sample, stats); /* quantize and scale */ return sample >> (MAD_F_FRACBITS + 1 - bits); } /* * NAME: prng() * DESCRIPTION: 32-bit pseudo-random number generator */ static inline unsigned long prng(unsigned long state) { return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL; } /* * NAME: audio_linear_dither() * DESCRIPTION: generic linear sample quantize and dither routine */ # if defined(_MSC_VER) extern /* needed to satisfy bizarre MSVC++ interaction with inline */ # endif inline signed long audio_linear_dither(unsigned int bits, mad_fixed_t sample, struct audio_dither *dither, struct audio_stats *stats) { unsigned int scalebits; mad_fixed_t output, mask, random; enum { MIN = -MAD_F_ONE, MAX = MAD_F_ONE - 1 }; /* noise shape */ sample += dither->error[0] - dither->error[1] + dither->error[2]; dither->error[2] = dither->error[1]; dither->error[1] = dither->error[0] / 2; /* bias */ output = sample + (1L << (MAD_F_FRACBITS + 1 - bits - 1)); scalebits = MAD_F_FRACBITS + 1 - bits; mask = (1L << scalebits) - 1; /* dither */ random = prng(dither->random); output += (random & mask) - (dither->random & mask); dither->random = random; /* clip */ if (output >= stats->peak_sample) { if (output > MAX) { ++stats->clipped_samples; if (output - MAX > stats->peak_clipping) stats->peak_clipping = output - MAX; output = MAX; if (sample > MAX) sample = MAX; } stats->peak_sample = output; } else if (output < -stats->peak_sample) { if (output < MIN) { ++stats->clipped_samples; if (MIN - output > stats->peak_clipping) stats->peak_clipping = MIN - output; output = MIN; if (sample < MIN) sample = MIN; } stats->peak_sample = -output; } /* quantize */ output &= ~mask; /* error feedback */ dither->error[0] = sample - output; /* scale */ return output >> scalebits; }

1 0

Libmad and libid3tag
by David Barron 26 Oct '04

26 Oct '04

Is it possible to get libmad and libid3tag as windows .dll's? Thanks

1 0

MAD and Jack.....
by andrew＠fnarg.demon.co.uk 08 Sep '04

08 Sep '04

Hi there, I'm writing a playout system which utilises Jack and MAD to play MP3 files. I've written a simple app which under libao (I was using this prior to Jack) will happily play any MP3 file I throw at it. However, upon adding the relevant code to make it a Jack client I've run across a problem. MAD seems to output audio at either 24bit fixed point or 16bit signed whereas Jack expects audio in 32bit floating point. Is there an easy way to make this work, am I being thick? Please help! Cheers, Andy M

2 1

How to manipulate both V1 and V2 tags in the same file
by Cedric Tefft 07 Sep '04

07 Sep '04

I've been playing around with libid3tag 0.15.1b for a couple days now, and I'm at a loss when it comes to manipulating multiple tags in the same file. When reading a file with only one tag (either V1 or V2), reading/parsing the frames and fields works more or less as I'd expect given the API functions in id3tag.h. Where I run into trouble is when I read an MP3 file with both V1 and V2 tags. The first thing I do after opening the file is call id3_file_tag() to get a pointer to a tag. What I find confusing is that even though the tag's ID3_TAG_OPTION_ID3V1 option is set, walking the field list only gives me the frames from the V2 tag: for(i=0;i<tag->nframes;i++) { frame=tag->frames[i]; /* etc. */ } Poking around a bit in file.c it appears the id3_file structure maintains references to all of the individual tags in the file, but looking through the API functions available in id3tag.h, I don't see how to access any tag other than the primary (via the id3_file_tag() function). So my question is, how do I access the fields of a V1 tag when a file has both V1 and V2 tags? Are the V1 fields stored in the primary tag structure along with the V2 fields? If so, why can't I access them when I walk the list of the other (V2) frames? If they're stored in a tag structure other than the primary, how do I access it? Thanks, - Cedric

2 2

Re: [mad-user] MAD and Jack.....
by jens-lippmann＠t-online.de 07 Sep '04

07 Sep '04

Hi! I'm new to the list and I'm not familiar with libmad, but have a look at this function from the madlld sample. You can easily change it to produce 32-bit floating point: float f; if (Fixed & (1<<31)) { // negative Fixed = Fixed & ((1<<31)-1); // strip sign f = -(float)Fixed / (1<<MAD_F_FRACBITS); } else { f = (float)Fixed / (1<<MAD_F_FRACBITS); } return f; hope this helps Jens /**************************************************************************** * Converts a sample from mad's fixed point number format to an unsigned * * short (16 bits). * ****************************************************************************/ static unsigned short MadFixedToUshort(mad_fixed_t Fixed) { /* A fixed point number is formed of the following bit pattern: * * SWWWFFFFFFFFFFFFFFFFFFFFFFFFFFFF * MSB LSB * S ==> Sign (0 is positive, 1 is negative) * W ==> Whole part bits * F ==> Fractional part bits * * This pattern contains MAD_F_FRACBITS fractional bits, one * should alway use this macro when working on the bits of a fixed * point number. It is not guaranteed to be constant over the * different platforms supported by libmad. * * The unsigned short value is formed by the least significant * whole part bit, followed by the 15 most significant fractional * part bits. Warning: this is a quick and dirty way to compute * the 16-bit number, madplay includes much better algorithms. */ Fixed=Fixed>>(MAD_F_FRACBITS-15); return((unsigned short)Fixed); }

1 0

Why too fast song tact by LibMad on RH9 Linux?
by Maxipoint Rep Office 17 Aug '04

17 Aug '04

Do you have a soundcard that will only play at 48KHz? RE: no, because same mp3s on windows play OK Or perhaps it's a MP3 recorded at 22KHz and you've opened the sound device at 44KHz. RE: no you can see in code that I am setRate (44100); all songs are recorded on the 192kbps 44.1kHz. Maybe I must define kbps too? -GP #include <sys/soundcard.h> #include <sys/types.h> #include <sys/stat.h> #include <sys/ioctl.h> #include <fcntl.h> #include <errno.h> #include <unistd.h> #include "audio.h" #include <qstring.h> CAudioSink::CAudioSink () { // initialize device if ((sound_dev = open ("/dev/sound/dsp", O_WRONLY)) == -1) sound_dev = open ("/dev/dsp", O_WRONLY); if (sound_dev == -1) throw tr ("Failed to initialize OSS device"); if (ioctl (sound_dev, SNDCTL_DSP_SYNC, 0) == -1) throw tr ("Failed to sync with DSP"); // TODO: this must be moved to another method to handle mono data channels = 2; ioctl (sound_dev, SNDCTL_DSP_CHANNELS, &channels); ioctl (sound_dev, SNDCTL_DSP_SYNC, 0); int fmt = AFMT_S16_LE; // set 16-bit data in little-endian format ioctl (sound_dev, SNDCTL_DSP_SAMPLESIZE, &fmt); ioctl (sound_dev, SNDCTL_DSP_SYNC, 0); // set default playback rate setRate (44100); } CAudioSink::~CAudioSink () { close (sound_dev); } // set samplerate of data void CAudioSink::setRate (int _rate) { rate = _rate; ioctl (sound_dev, SNDCTL_DSP_SPEED, &rate); } // reset audio device void CAudioSink::flush () { ioctl (sound_dev, SNDCTL_DSP_RESET, 0); } // play data buffer void CAudioSink::play (unsigned char* ptr, int len) { while (len) { int wrote; wrote = write (sound_dev, ptr, len); // this check handles that fact that write syscall can be // interrupted at any time if (wrote == -1) { if (errno == EINTR) continue; else break; } ptr += wrote; len -= wrote; } }

1 0

Why too fast song tact by LibMad on RH9 Linux?
by Maxipoint Rep Office 17 Aug '04

17 Aug '04

Why too fast song tact by LibMad on RH9 Linux? I am create own MP3 player with C++ and LibMad on the Linux. But song tact is to fast, all songs are faster with humoristic tone like on too fast vinil discs. Songs are min. 10% faster on LibMad and it is on ANY booster SO visible, and on small PC loudspeaker you must make more effort to hear, but when you know about that you can hear on small PC loudspeaker too Any help??

2 1

Qns abt MAD Library support for ARM environment
by balaji shanmugam 20 Jul '04

20 Jul '04

1 0

MAD library for TI-54X DSPs
by balaji shanmugam 20 Jul '04

20 Jul '04

Hi I am trying to port the MAD library for the TI-54X DSP and running into tight timing issues. When analysed, I found out that the 54X DSP takes more instructions to perform LONG calculations at the SYNTHESIS FILTER Bank stage as it is a 16 bit processor [but with a 40 bit accumulator]. As it is required to perform 72 times the DCT32 and the remaining MATRIX calculations at the SYNTH stage, I like to optimize the synthesis filter bank. I am not sure about using the Q28.4 format being used by MAD. Is there anyway we can use the Q15.1 format for the calculations? I like to know is there any hints or clues for the LONG calculations that I can make use of during my 54X porting phase. Also any thought about using the NORM instruction to round the final samples to 16 bit will be helpful. Regards Balaji Shanmugam _________________________________________________________________ Apply to 50,000 jobs now. http://go.msnserver.com/IN/46921.asp Post your CV on naukri.com today.

1 0

MAD library for TI-54X DSPs
by balaji shanmugam 20 Jul '04

20 Jul '04

1 0

Qns abt MAD Library support for ARM environment
by balaji shanmugam 20 Jul '04

20 Jul '04

1 0

Clipping in output
by Deven Phillips 15 Jul '04

15 Jul '04

Hey all, I'm having a problem with some of the MP3s I am decoding having a lot of clipping/popping. Any ideas for fixing this? Deven

1 0

How to use libid3tag to find offset to first mp3 header
by tj 28 Jun '04

28 Jun '04

What call in libid3tag returns the offset to the mp3 header? Plus, how the hell is that four byte sync safe length decoded? tj

2 1

Decoding small blocks of data
by peter＠neocoder.net 28 Jun '04

28 Jun '04

Hi I am trying to decode a mp3-file 100kB at a time. My problem is that with each new block some errors occur "Huffman data overrun" "bad main_data_begin pointer" "lost synchronization" I'm not familiar with the mp3-format structure but i guess that the data in the file should be sent to the decoder in complete blocks. I was hoping that the decoder would save incomplete data from the end of the previous buffer sent and join them with the next buffer sent. But it isn't ? So if I want to decode small blocks of the file. If I look through the file and only send complete blocks of data to the decoder. Will this solve my problem? Is it the way it is supposed to work? Peter Hultqvist peter.hultqvist(a)neocoder.net 2004-06-27, 16:54:17

2 1

Question about data strcutures
by Deven Phillips 24 Jun '04

24 Jun '04

Hi, I have written an application which uses libMAD. I was looking at the code for the library and I find that two things are bothering me: 1. The majority of cleanup functions are just #define'd stubs and actually do nothing. 2. Why aren't the data structures typedef'd for easier use? Just curious, and perhaps you can all provide me with a reaonable explanation that will make me more comfortable with it. Thanks!! Deven Phillips

2 1

libid3 documentation
by tj 22 Jun '04

22 Jun '04

Where can I find out how to use libid3tag library? All I wnat to do is open an mp3 file, see if it has or hasn't a tag. If it does, get song and artist and el me where the start of mp3 audio is.. If no id3v2, let me know so I know the audio is at the front of the file. tj

2 1

Can't configure libid3tag form arm processor
by tj 08 Jun '04

08 Jun '04

I am trying to configure libid3tag.0.15.1b for the arm processor. But it keeps failing on zlib.h. I run ./configure --host=arm and get: checking assert.h usability... yes checking assert.h presence... yes checking for assert.h... yes checking for unistd.h... (cached) yes checking zlib.h usability... no checking zlib.h presence... no checking for zlib.h... no configure: error: zlib.h was not found *** You must first install zlib (libz) before you can build this package. *** If zlib is already installed, you may need to use the CPPFLAGS *** environment variable to specify its installed location, e.g. -I<dir>. I check /usr/local/arm/2.95.3/arm-linux/include and there is zlib.h It finds assert.h, whcih is in the same directory, so why not zlib.h? tj

2 1

Some make problems
by Ming Chi Tsai 05 Jun '04

05 Jun '04

Hi: I have a project that port madplay into embedded system(CPU:ARM, OS:uclinux). I know madplay can be build on uclinux using uclibc and arm cpu using arm-linux-gcc respectively. But, when I port madplay to embedded system with ARM and uclinux using arm-elf-gcc, I encountered some problems. The following is my make step: I use cygwin and arm-elf-gcc to build my project. export CPPFLAGS='-I/usr/local/include/' export LDFLAGS='-I/usr/local/lib/' $cd zlib-1.1.4 modify configure AR=$(AR-"ar rc") ---> AR="arm-elf-ar rc" RANLIB=$(RANLIB-"ranlib") ---> RANLIB=arm-elf-ranlib $./configure ---------------->down $cd libid3tag-0.15.1b $./configure CC=arm-elf-gcc --host=arm-elf --build=arm $make $make install ---------------->down $cd libmad-0.15.1b $./configure CC=arm-elf-gcc --host=arm-elf --build=arm $make $make install ---------------->down $cd madplay-0.15.2b $./configure CC=arm-elf-gcc --host=arm-elf --build=arm ---------------->error, because it can not find libmad and libid3tag. But I am sure the lib path is correct. Where am I wrong with the configure setting?? Whether madplay can't be ported to embedded system with ARM and uclinux or I should use anyother corse compiler. thanks for any reply. Tsaimc

2 4

The madplay can't play mp3.
by kobi.yang＠billionton.com.tw 19 May '04

19 May '04

I have three mp3 songs and the madplay can't play these songs. Maybe are mp3 itself problems but i can play them in the windows media player. I don't know the reason and i hope to solve this problem. You can download mp3 songs below. Thanks for help. IP: 203.204.108.161 username: test password: 1234

1 0

ID3v2 tags
by Alexey Panasenko 13 May '04

13 May '04

Hello, All @ mad-user list I wonder why WinAmp MAD plugin does not support ID3v2 tags, displaying and extracting info only from ID3v1. Is it planned to implement ID3v2 in next versions? P.S. I'm currently using ver. 0.14.2beta P.P.S. Looks like now MAD plugin development is frozen, or goes very slow... Am I right? Thanks. -- Regards, Alexey mailto:morgen_stern@mail.ru n.p: PLAYING / Guns N' Roses - Don't Cry

1 0

hi
by rob＠mars.org 24 Apr '04

24 Apr '04

that is bad

1 0

*.pc files
by Pash-man 16 Apr '04

16 Apr '04

Hi all! Why not created files mad.pc and libmad.pc after compiling? I have compiled libmad-0.15.0b. -- Fri Apr 16 14:37:44 UTC 2004

1 0

dB gain --> command-line volume switch?
by Ert 05 Mar '04

05 Mar '04

Hiya. This isn't really a MAD question, but it was suggested to me to try this list: I'm trying to implement some support for the output of the 'normalize' package in my mp3 player of choice, but ran into the problem that it doesn't seem that either mpg123 or mpg321 currently support the RVA2 volume adjustment tag in ID3v2.4 tags. So I wrote a little routine in the jukebox wrapper program I'm using ('Tunez') to pull out normalize's backwards-compatible ID3v2.3 "XRVA" tag data. Practical upshot is that before I call mpg321 I know that I want my file to be played, say, 5 dB louder or 10 dB quieter, but don't know how to translate that into the " -gain " argument's 0-100 scale on mpg123 or mpg321. I could figure this out by trial and error, but was hoping there might be a more definitive conversion. Anyone have any thoughts? Thanks! - Ert Dredge

2 2

madplay 0.15.2b released
by Rob Leslie 23 Feb '04

23 Feb '04

Aleksander Korzynski pointed out to me the recent LAME heresy with respect to Replay Gain: rather than using the 83 dB SPL reference specified by the Replay Gain proposed standard, LAME uses a reference of 89 dB as of version 3.95.1. Combined with the default pre-amp gain of +6 dB implemented in madplay, this often results in an over-amplified signal and clipping when Replay Gain is enabled. I've just released version 0.15.2b of madplay that compensates for this problem: ftp://ftp.mars.org/pub/mpeg/madplay-0.15.2b.tar.gz Cheers, -- Rob Leslie rob(a)mars.org

1 0

a patch for using the last alsa
by Florin Braescu 19 Feb '04

19 Feb '04

Hi everybody, I am a user of the source based Lunar Linux distro (www.lunar-linux.org). This distro contains the mad stuff. In order to use the alsa module (last version 1.0.2c) i use this simple patch. Please incorporate it in the package. Regards, Florin patch begins here --> --- ./audio_alsa.c.old 2003-05-30 09:26:17.000000000 +0300 +++ ./audio_alsa.c 2004-01-08 17:13:39.329443480 +0200 @@ -27,7 +27,11 @@ #include "global.h" #include <errno.h> -#include <sys/asoundlib.h> + +#define ALSA_PCM_OLD_HW_PARAMS_API +#define ALSA_PCM_OLD_SW_PARAMS_API +#include <alsa/asoundlib.h> + #include <mad.h> #include "audio.h" <-- and ends here. __________________________________ Do you Yahoo!? Yahoo! Mail SpamGuard - Read only the mail you want. http://antispam.yahoo.com/tools

1 0

MAD suite 0.15.1b released
by Rob Leslie 18 Feb '04

18 Feb '04

MAD 0.15.1b is now available. Here is a summary of the important changes: libmad - New Layer III IMDCT routine based on Szu-Wei Lee's fast algorithm. - Improved robustness of Layer II bitrate/mode combinations. - Other improved portability and robustness. libid3tag - New id3_field_gettextencoding() API routine. - Other slightly improved robustness. madplay - New Replay Gain support (-G, --replay-gain). - Tags can now be shown without decoding any audio (-T, --show-tags-only). - New support for reading and displaying encoder tag information written by e.g. lame(1). For complete details, see the CHANGES file in each package as well as the updated man page for madplay(1). You will find the new releases here: ftp://ftp.mars.org/pub/mpeg/ Enjoy, -- Rob Leslie rob(a)mars.org

3 2

(no subject)
by wayne＠dfw-sites.com 06 Feb '04

06 Feb '04

Hi All Does anyone know how I can make madplay access the number pad of a keyboard instead of the q p s keys for play and pause stop etc....? thanks

1 0

Rendering 2.3 tags with libid3tag-0.15.0b
by Vince Busam 23 Jan '04

23 Jan '04

Im trying to render 2.3 tags with libid3tag-0.15.0b and having problems. I understand this may not be a feature of this code but Im close. Ill try to make this short, but some detail might be useful. I have a LGPL library (http://www.busam.com/skyland/iavc) for which I am writing a GPL front end. My development environment is Windows. When I use libid3tag-0.15.0b to render tags in an output MP3 file, I see what are almost certainly 2.4 version tags. Neither stock Windows Media Player 9 nor the PhatNoise Music Manager (PMM) (http://www.phatnoise.com) recognizes these tags. Since I use the PMM extensively, I require compatibility with it. The files ripped by PMM have 2.3 tags at the beginning of the mp3 file and version 1 tags at the end. When I do m_libmad.getID3tags(), I get a id3_tag structure with the ID3_TAG_OPTION_ID3V1 bit set. When I rendered the tags I got version 1 tags (no surprise because of the set bit). After removing that bit and defining ID3_TAG_VERSION as 0x0300, I get ID3 tags that are close to 2.3 tags. At least PMM recognizes that there are tags and shows most of them. Windows Media Player 9 still doesnt show any sign of recognizing these tags. Next I found that in compat.gperf that TYER is set to obsolete and that translate_TCON removes parenthesis. Before I get any deeper, I thought Id check to see if there is a better suggestion as to how to get 2.3 tags rendered. Thanks, Vince

1 0

Re: mad-user digest, Vol 1 #176 - 1 msg
by Florin Braescu 08 Jan '04

08 Jan '04

--- mad-user-request(a)lists.mars.org wrote: > Message: 1 > Date: Wed, 7 Jan 2004 13:25:03 -0800 (PST) > From: Florin Braescu <braescu_f(a)yahoo.com> > To: mad-user(a)lists.mars.org > Subject: [mad-user] compile fail for madplay with > alsa-1.0.0.rc2 > > Hi everybody, > > I have tried to compile the madplay app using > alsa-lib > and alsa-utils 1.0.0rc2 and linux kernel 2.6.1-rc1. > I > use also gcc 3.3.2. The compile process failed. It > seems that the audio_alsa.c use sys/asoundlib.h > instead alsa/asoundlib.h. However modifying that it > didn't help either. I want to mention that the 0.9.8 > version of alsa-lib and alsa-utils permit the > compiling of madplay and work just fine. Maybe some > dev have some time to take a look at this situation. > > Regards, > Florin > Hi again, I have find a patch for this problem. I attach it here. Yours, Florin __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

1 0

compile fail for madplay with alsa-1.0.0.rc2
by Florin Braescu 07 Jan '04

07 Jan '04

Hi everybody, I have tried to compile the madplay app using alsa-lib and alsa-utils 1.0.0rc2 and linux kernel 2.6.1-rc1. I use also gcc 3.3.2. The compile process failed. It seems that the audio_alsa.c use sys/asoundlib.h instead alsa/asoundlib.h. However modifying that it didn't help either. I want to mention that the 0.9.8 version of alsa-lib and alsa-utils permit the compiling of madplay and work just fine. Maybe some dev have some time to take a look at this situation. Regards, Florin __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

1 0

Decode directories, how to?
by jrx＠telia.com 02 Jan '04

02 Jan '04

Hi list, I would like to decode a whole directory in a bunch. Probably I have to use some kind of ordinary *nix command (I'm from the other side), but I can't figure it out, and I can't find any description of this presumably easy matter in man madplay Thanks in advance. jrx

4 4

Complemetary questions about decoding a dir and about formats
by jrx＠telia.com 01 Jan '04

01 Jan '04

Hi list and thanks Rob for your answer, A few complementary questions, again maybe a bit OT: I do not really need wav format (by the way, is wav in some way a ms invention?). What I'm trying to do is to decode a lot of mp3s to a format usable for listening with a non-mp3-capable cd player, i.e in my car, and then burn them to CDs with k3b. At the first stage (before burning) I would like to save the mp3s as individual output files in another directory than the mp3s. 1. Is .cdr a format for my purposes? 2. Shell scripts are a bit beyond my capacity for the moment (but I'm learning fast). What do you mean with "The above is for Bourne-derived shells; hopefully yours is not of Csh variety."? What about this? If i navigate to the directory where the mp3s are and then: for file .mp3; do madplay -o -v $file.cdr $file; done Where in this little script should I put the target directory name (called "cdr)? What does "done" mean? Close? TIA /jrx, Stockholm -----Original Message----- From: mad-user-admin(a)lists.mars.org [mailto:mad-user-admin@lists.mars.org] On Behalf Of mad-user-request(a)lists.mars.org Sent: Wednesday, December 31, 2003 3:27 PM To: mad-user(a)lists.mars.org Subject: mad-user digest, Vol 1 #172 - 4 msgs Send mad-user mailing list submissions to mad-user(a)lists.mars.org To subscribe or unsubscribe via the World Wide Web, visit http://www.mars.org/bin/mailman/listinfo/mad-user or, via email, send a message with subject or body 'help' to mad-user-request(a)lists.mars.org You can reach the person managing the list at mad-user-admin(a)lists.mars.org When replying, please edit your Subject line so it is more specific than "Re: Contents of mad-user digest..." Today's Topics: 1. Decode an old mp3, originally recorded in June-50 (jrx(a)telia.com) 2. Decode directories, how to? (jrx(a)telia.com) 3. Re: Decode an old mp3, originally recorded in June-50 (Rob Leslie) 4. Re: Decode directories, how to? (Rob Leslie) --__--__-- Message: 1 From: <jrx(a)telia.com> To: <mad-user(a)lists.mars.org> Date: Tue, 30 Dec 2003 22:00:19 +0100 Subject: [mad-user] Decode an old mp3, originally recorded in June-50 Hi, maybe OT. I have tried to decode an old mp3 with various settings to wav with mad, and the decoding seems to work fine. But when I try to burn the wav with k3b, k3b says "wrong format". madplay -v -i -m --output=wave gives "Layer III, 64 kbps, 22050 Hz, single channel, no CRC, 7095 frames decoded, -7.4 dB peak amplitude, 0 clipped samples" Does anybody have any suggestions about what options I should choose when decoding. I would like to have it in wav. Or maybe I should look in my k3b settings for this particular file..? Best regards /jrx --__--__-- Message: 2 From: <jrx(a)telia.com> To: <mad-user(a)lists.mars.org> Date: Tue, 30 Dec 2003 22:00:19 +0100 Subject: [mad-user] Decode directories, how to? Hi list, I would like to decode a whole directory in a bunch. Probably I have to use some kind of ordinary *nix command (I'm from the other side), but I can't figure it out, and I can't find any description of this presumably easy matter in man madplay Thanks in advance. jrx --__--__-- Message: 3 From: Rob Leslie <rob(a)mars.org> Subject: Re: [mad-user] Decode an old mp3, originally recorded in June-50 Date: Tue, 30 Dec 2003 18:09:11 -0800 To: mad-user(a)lists.mars.org On Dec 30, 2003, at 1:00 PM, <jrx(a)telia.com> wrote: > I have tried to decode an old mp3 with various settings to wav with > mad, and the decoding seems to work fine. > > But when I try to burn the wav with k3b, k3b says "wrong format". > madplay -v -i -m --output=wave gives "Layer III, 64 kbps, 22050 Hz, > single channel, no CRC, 7095 frames decoded, -7.4 dB peak amplitude, 0 > clipped samples" > > Does anybody have any suggestions about what options I should choose > when decoding. I would like to have it in wav. If you're trying to burn an audio CD, try: madplay -o cdda:track.cdr The result is a raw CD audio track ready to be burned. If you really must have a WAV file, you'll have to specify all the parameters yourself: madplay -o foo.wav --stereo --bit-depth=16 --sample-rate=44100 -- Rob Leslie rob(a)mars.org --__--__-- Message: 4 From: Rob Leslie <rob(a)mars.org> Subject: Re: [mad-user] Decode directories, how to? Date: Tue, 30 Dec 2003 18:17:10 -0800 To: mad-user(a)lists.mars.org On Dec 30, 2003, at 1:00 PM, <jrx(a)telia.com> wrote: > I would like to decode a whole directory in a bunch. Probably I have > to use some kind of ordinary *nix command (I'm from the other side), > but I can't figure it out, and I can't find any description of this > presumably easy matter in man madplay You can either have madplay decode everything in one go: madplay -o out.wav *.mp3 (in this case there is only one output) or if you need individual output files you can use some shell scripting to decode files one-at-a-time: for file in *.mp3; do madplay -o $file.wav $file; done The above is for Bourne-derived shells; hopefully yours is not of Csh variety. -- Rob Leslie rob(a)mars.org End of mad-user Digest

2 1

Decode an old mp3, originally recorded in June-50
by jrx＠telia.com 31 Dec '03

31 Dec '03

Hi, maybe OT. I have tried to decode an old mp3 with various settings to wav with mad, and the decoding seems to work fine. But when I try to burn the wav with k3b, k3b says "wrong format". madplay -v -i -m --output=wave gives "Layer III, 64 kbps, 22050 Hz, single channel, no CRC, 7095 frames decoded, -7.4 dB peak amplitude, 0 clipped samples" Does anybody have any suggestions about what options I should choose when decoding. I would like to have it in wav. Or maybe I should look in my k3b settings for this particular file..? Best regards /jrx

2 1

Problem with 24 bit output on M-Audio Revolution 7.1
by Jason Clouse 21 Dec '03

21 Dec '03

Greetings! I just got hold of an M-Audio Revolution 7.1 and decided to start playing around with it. I've got the latest CVS of ALSA and AlsaPlayer installed and I can successfully get sound of the card. I'm having some trouble, however. I read that the Revolution has a pure 24/192 signal path from start to finish, and that madplay can take advantage of this. I tried running madplay with the '--bit-depth=24' option but it told me that the card couldn't handle 24 bit output, and that it was dropping back to 16. Did I miss something? Jason Clouse

2 2

Madlib Slow?
by Tomas 15 Dec '03

15 Dec '03

Hi! I have compiled the madlib and tried to decode MPEG2 frames on a armv4 chip (AT91RM9200) with processor clock ~180 MHz but it seems too slow. I need to decode a MPEG2 (layerII) frame (say 224kHz) each 24 ms. I have cross-compiled the madlib with the following configuration CC=arm-elf-gcc RANLIB=arm-elf-ranlib LD=arm-elf-ld AR=arm-elf-ar ../mad- 0.14.2b/configure --enable-speed --enable-fpm=arm --disable-nls --disable- mmap --disable-debugging --host=arm-elf arm-elf Correct configuration to enable speed or can I try something else? Is any body sucessfully running this on this processor or similar? /Tomas

1 0

Questions about how to build a ARM based madplay
by Kzhang 10 Dec '03

10 Dec '03

Hi All, I download both of mad-0.14.2b and madplay-0.15.0b, and I try to build a ARM based player, but I met problem to change gcc to arm-linux-gcc with ./configure command. I work on a ReHat 9.0 PC as host, and use arm cross compiler 2.95.3 toolchain. Please give me some tips how to build it, I very appreciate for your help. Thanks. Keji Zhang kzhang(a)neomagic.com

1 0

Build problem with cygwin/mingw
by Jedediah Smith 14 Nov '03

14 Nov '03

Following steps were followed: ./configure --host=mingw32 --prefix==/cygdrive/c/source/vc/mixer/libmad Configure step goes fine. The prefix is the dir I want to install to e.g. c:\source\vc\mixer\libmad which already exists. make also works fine make install goes like this: -------------------------------------------------------------------------------- $ make install make install-recursive make[1]: Entering directory `/cygdrive/c/source/vc/mixer/libmad-0.15.0b/libmad-0 .15.0b' make[2]: Entering directory `/cygdrive/c/source/vc/mixer/libmad-0.15.0b/libmad-0 .15.0b' make[3]: Entering directory `/cygdrive/c/source/vc/mixer/libmad-0.15.0b/libmad-0 .15.0b' /bin/bash ./mkinstalldirs /cygdrive/c/source/vc/mixer/libmad/lib /bin/bash ./libtool --mode=install /usr/bin/install -c libmad.la /cygdrive/c/s ource/vc/mixer/libmad/lib/libmad.la /usr/bin/install -c .libs/libmad.lai /cygdrive/c/source/vc/mixer/libmad/lib/libm ad.la /usr/bin/install -c .libs/libmad.a /cygdrive/c/source/vc/mixer/libmad/lib/libmad .a ranlib /cygdrive/c/source/vc/mixer/libmad/lib/libmad.a c:\mingw\bin\ranlib.exe: /cygdrive/c/source/vc/mixer/libmad/lib/libmad.a: No suc h file or directory make[3]: *** [install-libLTLIBRARIES] Error 9 make[3]: Leaving directory `/cygdrive/c/source/vc/mixer/libmad-0.15.0b/libmad-0. 15.0b' make[2]: *** [install-am] Error 2 make[2]: Leaving directory `/cygdrive/c/source/vc/mixer/libmad-0.15.0b/libmad-0. 15.0b' make[1]: *** [install-recursive] Error 1 make[1]: Leaving directory `/cygdrive/c/source/vc/mixer/libmad-0.15.0b/libmad-0. 15.0b' make: *** [install] Error 2 -------------------------------------------------------------------------------- It looks like ranlib is being passed the cygwin version of the path but it doesn't recognize it as it's part of mingw and only allows normal Windows paths. What shall I do about this?

2 1

Question about MAD on ARM
by hover0815＠yahoo.com.tw 25 Sep '03

25 Sep '03

Hi, I have several questions about MAD plays mp3 on ARM-based platform. When I play mp3 with MAD on ARM, there is an unusual phenomenon. 1. While playing a mp3 file which frequency is lower than 44100 hz, the music sounds like downsampled. Which means the sound is faster such that the time of play is fewer than original. 2. When applying the attenuation command (ex, --attenuate=10), some mp3 files will play intermittently. It seems CPU cannot support such high calculation. (CPU clock rate is 96khz) There are additional 2*32*36 multiplication per frame. Can MAD be simplified so that it can play mp3 more smoothly? 3. Why do some files play intermittently, and some don't? I tried several files, and still can not get its rule. bit rate freq. channel mode result 112k 44100 JS ok 128k 32000 JS X 128k 44100 S ok 192k 44100 JS ok 192k 44100 S X It seems irrelative to bit rate, sample frequency or channel mode. What's the rule? Can someone help me? Thank you --------------------------------- 每天都 Yahoo!奇摩海的顏色、風的氣息、愛你的溫度，盡在信紙底圖信紙底圖

2 1

sox and dithering
by DervishD 21 Aug '03

21 Aug '03

Hi all :) I've noticed that the cdda output of 'sox' from a mp3 file is *very* different from the output of madplay, given of course the same mp3 file. In fact the diference is even noticeable in the waveform. I wonder if the cause is that sox does not perform dithering, but disabling dithering in madplay doesn't solve the problem. Maybe sox is just truncating the samples? I'm not asking to the sox author yet because this can be a MAD issue (configuration options of the library generating different output but not lesser quality, for example). Thanks a lot in advance. BTW, I use madplay as mp3 player, because I think that it is what a player *must* be: simple, easy, fast, etc... Great, Rob, great. Raúl Núñez de Arenas Coronado -- Linux Registered User 88736 http://www.pleyades.net & http://raul.pleyades.net/

1 0

Madplay capture output to file - how to?
by Gurpal S. Bhachu 20 Aug '03

20 Aug '03

Hi, I wish to capture the output of madplay to an external file eg: $ madplay abc.mp3 -o abc.wav > abc.txt It seems that abc.txt contains nothing! Is there some weird non standard out thing happening here? Please could someone advise how to do this... Many thanks, Gurpal

2 2

Re: mad-user digest, Vol 1 #160 - 1 msg
by dj nano 25 Jul '03

25 Jul '03

mad-user-request(a)lists.mars.org wrote: >Send mad-user mailing list submissions to > mad-user(a)lists.mars.org > >To subscribe or unsubscribe via the World Wide Web, visit > http://www.mars.org/bin/mailman/listinfo/mad-user >or, via email, send a message with subject or body 'help' to > mad-user-request(a)lists.mars.org > >You can reach the person managing the list at > mad-user-admin(a)lists.mars.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of mad-user digest..." > > >Today's Topics: > > 1. Re: MAD play will not compile (Rob Leslie) > >--__--__-- > >Message: 1 >Date: Thu, 10 Jul 2003 11:39:04 -0700 >Subject: Re: [mad-user] MAD play will not compile >From: Rob Leslie <rob(a)mars.org> >To: mad-user(a)lists.mars.org > >On Wednesday, July 9, 2003, at 10:51 PM, dj nano wrote: > > >>Hello, I've searched the archives and no one else seems to have this >>problem: When I attempt to compile the 4th of the tarballs, >>madplay-0.15.0b it finds errors in the 'libid3' links. Libid3 already >>compiled successfully. This occurs even when I use the syntax >>"./configure --libdir=/usr/local/lib", (the configure completes). Is >>this error somehow related to all the info in the archives about MAD >>not handling ID3 tagging properly? Is there a workaround? Unlike >>most I don't really care if the ID info shows up; I'm looking for >>better playback. I use Red Hat 9 on a dual athlon 1900+ with an >>m-Audio delta 1010 (run by ALSA). Can anyone advise? >> >> > >The problem seems to be you are linking against older versions of >libmad and libid3tag. Make sure you run 'make install' on both of those >after you build them, before you build madplay. > >FYI, --libdir only affects the installation directory for libraries, >not where to look for them for building/linking. If you've installed >the libraries in an unusual place, you'll need to tell the 'configure' >for madplay where they are like this: > > ./configure ... CPPFLAGS="-I/some/include/dir" >LDFLAGS="-L/some/lib/dir" > > > I did a make install on all items in question and they are all in the default locations. I will look for updates but I downloaded the latest from sourceforge. Thanks!

1 0

MAD play will not compile
by dj nano 10 Jul '03

10 Jul '03

Hello, I've searched the archives and no one else seems to have this problem: When I attempt to compile the 4th of the tarballs, madplay-0.15.0b it finds errors in the 'libid3' links. Libid3 already compiled successfully. This occurs even when I use the syntax "./configure --libdir=/usr/local/lib", (the configure completes). Is this error somehow related to all the info in the archives about MAD not handling ID3 tagging properly? Is there a workaround? Unlike most I don't really care if the ID info shows up; I'm looking for better playback. I use Red Hat 9 on a dual athlon 1900+ with an m-Audio delta 1010 (run by ALSA). Can anyone advise? error out put: make all-recursive make[1]: Entering directory `/home/morpha/MAD/madplay-0.15.0b' Making all in intl make[2]: Entering directory `/home/morpha/MAD/madplay-0.15.0b/intl' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/home/morpha/MAD/madplay-0.15.0b/intl' Making all in po make[2]: Entering directory `/home/morpha/MAD/madplay-0.15.0b/po' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/home/morpha/MAD/madplay-0.15.0b/po' make[2]: Entering directory `/home/morpha/MAD/madplay-0.15.0b' /bin/sh ./libtool --mode=link gcc -Wall -march=i486 -g -O2 -o madplay madplay.o getopt.o getopt1.o version.o resample.o filter.o xing.o player.o audio.o audio_aiff.o audio_cdda.o audio_hex.o audio_null.o audio_raw.o audio_snd.o audio_wave.o audio_esd.o audio_oss.o -lesd -lmad -lid3tag -lm gcc -Wall -march=i486 -g -O2 -o madplay madplay.o getopt.o getopt1.o version.o resample.o filter.o xing.o player.o audio.o audio_aiff.o audio_cdda.o audio_hex.o audio_null.o audio_raw.o audio_snd.o audio_wave.o audio_esd.o audio_oss.o -lesd /usr/local/lib/libmad.a /usr/local/lib/libid3tag.a -lm resample.o(.text+0x29): In function `resample_init': /home/morpha/MAD/madplay-0.15.0b/resample.c:46: undefined reference to `mad_f_div' player.o(.text+0xb56): In function `show_id3': /home/morpha/MAD/madplay-0.15.0b/player.c:630: undefined reference to `id3_frame_field' player.o(.text+0xc11):/home/morpha/MAD/madplay-0.15.0b/player.c:674: undefined reference to `id3_frame_field' player.o(.text+0xc36):/home/morpha/MAD/madplay-0.15.0b/player.c:680: undefined reference to `id3_frame_field' player.o(.text+0x10dd): In function `process_id3': /home/morpha/MAD/madplay-0.15.0b/player.c:801: undefined reference to `id3_frame_field' player.o(.text+0x112e):/home/morpha/MAD/madplay-0.15.0b/player.c:846: undefined reference to `id3_frame_field' player.o(.text+0x1137):/home/morpha/MAD/madplay-0.15.0b/player.c:846: undefined reference to `id3_field_getlatin1' player.o(.text+0x114a):/home/morpha/MAD/madplay-0.15.0b/player.c:847: undefined reference to `id3_frame_field' /usr/local/lib/libid3tag.a(util.o)(.text+0x105): In function `id3_util_compress': /home/morpha/MAD/mad-0.14.2b/libid3tag/util.c:116: undefined reference to `compress2' /usr/local/lib/libid3tag.a(util.o)(.text+0x160): In function `id3_util_decompress': /home/morpha/MAD/mad-0.14.2b/libid3tag/util.c:142: undefined reference to `uncompress' collect2: ld returned 1 exit status make[2]: *** [madplay] Error 1 make[2]: Leaving directory `/home/morpha/MAD/madplay-0.15.0b' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/home/morpha/MAD/madplay-0.15.0b' make: *** [all] Error 2

2 1

Cutting wave file
by mukesh harjai 02 Jul '03

02 Jul '03

Hi All, I need to cut the wave file and store it as another file, Please tell me cutting wave file is just like cutting any binary file or some care needs to be taken care of while cutting wave file. Thanks and Regards, Mukesh Harjai _________________________________________________________________ Dress up your desktop! Get the best wallpapers. http://server1.msn.co.in/msnchannels/Entertainment/wallpaperhome.asp Just click here!

2 1

madplay keyboard controls
by Naomi Bristol 24 Jun '03

24 Jun '03

What are the interactive keyboard controls for madplay? Such as for pause, stop, increase/decrease volume, etc...I've been looking in all the documentation and can't seem to find it...maybe my eyes are just tired. Thank you. ********************************* * naomi(a)winlab.rutgers.edu * *********************************

2 1

MAD Output Format
by mukesh harjai 23 Jun '03

23 Jun '03

Hi All, I have followed madlld application for Implementing MAD Decoder. MAD Creates Output wav file of desired size, but when i tried to play it using Windows Media Player or Winamp, it does not recognize the wav format. Please tell me,how this problem can be corrected. Thanks in advance. Regards, Mukesh Harjai _________________________________________________________________ Are you a geek? Are you a techno freak? http://www.msn.co.in/Computing/

2 1

(no subject)
by mukesh harjai 23 Jun '03

23 Jun '03

Hi All, I have followed madlld application for Implementing MAD Decoder. MAD Creates Output wav file of desired size, but when i tried to play it using Windows Media Player or Winamp, it does not recognize the wav format. Please tell me,how this problem can be corrected. Thanks in advance. Regards, Mukesh Harjai _________________________________________________________________ Are you Unmarried? Register in India's No 1 Matrimony. http://www.bharatmatrimony.com/cgi-bin/bmclicks1.cgi?4d

1 0

help for missing files !! Urgent !!
by Bhumika 18 Jun '03

18 Jun '03

Hi, We are trying to build the VLC code in WinCE. We are not able to bulid as some of the files are missing in MAD library. Following files are missing, 1. libmad.c 2. layer1.c 3. layer2.c 4. errno.h 5. math.c These 5 files are being used in VLC source code. We checked the source code available in FTP as well as in Mirror site. But we didnt fine these files there also. If we get these files it will be helpful for us to compile and build the source code. Please help us as it very urgent for us. Thanks, Bhumika

2 1

help for missing files !! Urgent !!
by Bhumika 18 Jun '03

18 Jun '03

Hi, We are trying to build the VLC code in WinCE. We are not able to bulid as some of the files are missing in MAD library. Following files are missing, 1. libmad.c 2. layer1.c 3. layer2.c 4. errno.h 5. math.c These 5 files are being used in VLC source code. We checked the source code available in FTP as well as in Mirror site. But we didnt fine these files there also. If we get these files it will be helpful for us to compile and build the source code. Please help us as it very urgent for us. Thanks, Bhumika

1 0

MAD suite 0.15.0b released
by Rob Leslie 05 Jun '03

05 Jun '03

At long last, MAD 0.15.0b is now available. This release is a bit different from previous releases: there are now independent packages for libmad, libid3tag, and madplay. There is not at this time an update to the MAD plug-in for Winamp; 0.14.2b is still the current release for the plug-in. Here is a summary of the important changes: General - Shared libraries are now built by default. - Cygwin builds have reverted to Cygwin dependency by default; MinGW builds are still possible with the --host=mingw32 option to 'configure'. libmad - Minor compatibility enhancements: 64th entry to Layer I/Layer II scalefactor table; allow reserved emphasis value in frame headers. - Bug fix for ARM mad_f_scale64() that was tickled by certain optimization options. - API additions: mad_f_div(), MAD_EMPHASIS_RESERVED. - Rewritten PPC fixed-point assembly. libid3tag - Numerous API additions: id3_tag_version(), id3_tag_options(), id3_tag_setlength(), id3_frame_field(), id3_field_getlatin1(), id3_field_getfulllatin1(), id3_genre_index(), id3_genre_number(), id3_latin1_ucs4duplicate(), id3_utf16_ucs4duplicate(), id3_utf8_ucs4duplicate(). - Fixed missing interfaces for other API routines. - Bug fix for segfault rendering ID3v1 tags. - Bug fix for reading ID3v2.2 frames. - Other miscellaneous improvements. madplay - Support for arbitrary resampling ratios (--sample-rate). - Support for RVA2 (relative volume adjustment) ID3 tag frames. - Keyboard controls for dynamic volume control (+/-). - Support for ancillary data output (--ancillary-output). - New audio_qnx, audio_alsa modules (untested). - New audio_carbon module (tested, for Mac OS X). - Other miscellaneous improvements. For complete details, see the CHANGES file in each package. Be sure also to review the README in each package for modified build/install instructions. You will find the new releases here: ftp://ftp.mars.org/pub/mpeg/ Windows developers who need zlib (no longer included with the above) can find a convenient package here: ftp://ftp.mars.org/pub/mpeg/extra/ (Note that zlib is already available as part of Cygwin, so you may only need this package if you are using MSVC++ or want to use MinGW to eliminate the Cygwin dependency.) Enjoy, -- Rob Leslie rob(a)mars.org

1 0

MAD Sample Application
by mukesh harjai 05 Jun '03

05 Jun '03

Hi Everybody, If anybody has sample application which Uses MAD for decoding mp3 files, Please send it to me. Thanks in advance. Regards, Mukesh Harjai _________________________________________________________________ Got a wish? Make it come true. http://server1.msn.co.in/msnleads/citibankpersonalloan/index.asp Best personal loans!

1 0

RE: [mad-user] Require MAD Help
by Scott Parkerson 15 May '03

15 May '03

Mukesh, There is sample code distributed with the MAD tarball. See minimad.c for a really stripped down example. Madplay.c (and its associated files) show a more "full-bodied" example of the high-level interface. Additionally, there is an example of the low-level API on the web here: http://www.bsd-dk.dk/%7Eelrond/audio/madlld/ Hope this helps. --sgp -----Original Message----- From: mukesh harjai [mailto:mukesh_harjai@hotmail.com] Sent: Thursday, May 15, 2003 10:19 AM To: mad-user(a)lists.mars.org Subject: [mad-user] Require MAD Help Hi I need a sample code or Documentation which demonstrate how to use MAD Library. If you have it,Please mail it to me. Thanks Mukesh Harjai _________________________________________________________________ Your life is precious. Protect it. http://www.ingvysyalife.com/LeadDtls_msn.asp Go for ING Vysya Insurance

1 0

Require MAD Help
by mukesh harjai 15 May '03

15 May '03

Hi I need a sample code or Documentation which demonstrate how to use MAD Library. If you have it,Please mail it to me. Thanks Mukesh Harjai _________________________________________________________________ Your life is precious. Protect it. http://www.ingvysyalife.com/LeadDtls_msn.asp Go for ING Vysya Insurance

1 0

MAD Help
by mukesh harjai 15 May '03

15 May '03

Hello Friends, I need a sample code or Documentation which demonstrate how to use MAD Library. If you have it,Please mail it to me. Bye Mukesh Harja _________________________________________________________________ Bridge gaps. Reconnect with old friends. http://www.batchmates.com/msn.asp On batchmates.com

1 0

How do I unregister from this email list?
by Lyndon Tiu 09 May '03

09 May '03

Hello, Sorry for the SPAM. I just need to know how to unregister from this email list. Thank you. Lyndon

2 1

Re: [mad-user] Documentation on MAD
by Senthil Kumar 09 May '03

09 May '03

Dear Sir, Is there any wav player utility for Strong Arm platform (SA1110). If so please proived me some information regarding this. I tried to use madplay, i am able to play only in the x86 platform. Is it possible to use madplay for Strong Arm platform. If so please provide me some idea. with thanks and regards Senthil _________________________________________________________________ Attention Taureans! See what's in store. http://server1.msn.co.in/features/taurus/index.asp Read on.

2 1

Documentation on MAD
by mukesh harjai 05 May '03

05 May '03

Hello Friends, I am working on an audio Editor as part of my MCA. For that i need to decode MP3 files into .WAV files, and i would be using MAD Decoder for that purpose. I don't have documentation for MAD Please send me documentation for using MAD i.e. Decoding functions present in MAD library and their parameters etc. Bye Mukesh Harjai _________________________________________________________________ Win a PC. Every 15 days! http://server1.msn.co.in/sp03/hclbeanstalktour/index.asp Just take this quiz.

2 1

(no subject)
by Senthil Kumar 27 Mar '03

27 Mar '03

Dear All, Producing musical tones through FM Syntheizer is little bit confusing for me. If possible can u please send me any link or example for the generating the musical tones using FM synthesizer. with thanks and regards Senthil _________________________________________________________________ Say it now. Say it online. http://www.msn.co.in/ecards/ Send e-cards to your love

2 1

(no subject)
by Senthil Kumar 26 Mar '03

26 Mar '03

Hi, I want to play the mp3 file on my own. so i extracted the data from the mp3 file leaving the frame information of size 4 bytes. Then i extracted the the bit size, padding value, bit/sample, sampling rate and calculated the frame size from the frame information. After that i have extracted the data to my local buffer of size 4096 and i wrote to the dsp device repeatedly upto the end of the file. But i was able to get annoying sound. Please can u give me solution for this problem.. with thanks and regards Senthil _________________________________________________________________ Get ball by ball action on your desktop. http://server1.msn.co.in/msnspecials/cricketdownload/contest.asp Get Hutch MSN Cricketer

3 2

mad output tested for iso-compliance
by Praseetha K K 22 Mar '03

22 Mar '03

Hi all, I would like to know if anybody has done a testing of MAD decoder output for MPEG1 iso-streams and compared them with the standard iso decoder(l3dec) output. I find that si_huff.bit(this is an iso-stream) shows a difference from iso decoder o/p in the frames 4,5 and 6.Possibly it could be in the tables or in implementaion of huffmann in MAD. If anybody has already cleared out thsi problem, please help. Cheers, Praseetha

2 1

Help installing
by Shane Bauer 05 Feb '03

05 Feb '03

Hi everyone, I need some help. One of my packages requires a libmad.so.0 file. When I installed one of the packages (and the source the second time), a libmad.so.0 file was not created. Can anyone help? Thanks, Shane

2 1

MAD plug-in: Lack of Unicode support
by Sergej Malinovski 29 Jan '03

29 Jan '03

Any plans on adding support for Unicode file names in Win2000? Sergej

1 0

very noisy output from madplay.
by Ivan A. Uemlianin 02 Jan '03

02 Jan '03

Dear all I've just downloaded and installed mad on an intel box running Debian 3. Compilation all went fine (I just used configure/make/make install). I'd like to use mad to play/extract the audio layer from mpeg video files. As a quick test, I ran madplay on an mpeg downloaded from the net. The output was pretty rough - the audio was there, but so was a lot of clicking, squeaking, etc. I got the following output. MPEG Audio Decoder 0.14.2 (beta) - Copyright (C) 2000-2001 Robert Leslie error: frame 0: lost synchronization error: frame 5: lost synchronization error: frame 10: lost synchronization error: frame 15: lost synchronization error: frame 20: lost synchronization error: frame 25: lost synchronization error: frame 30: bad scalefactor index error: frame 34: lost synchronization error: frame 39: lost synchronization error: frame 44: lost synchronization error: frame 49: lost synchronization error: frame 54: lost synchronization error: frame 59: lost synchronization error: frame 63: lost synchronization error: frame 68: lost synchronization error: frame 73: lost synchronization error: frame 78: lost synchronization error: frame 83: lost synchronization error: frame 88: lost synchronization error: frame 92: lost synchronization error: frame 97: lost synchronization error: frame 102: lost synchronization error: frame 107: lost synchronization error: frame 112: lost synchronization error: frame 117: lost synchronization error: frame 121: lost synchronization error: frame 125: lost synchronization error: frame 130: lost synchronization error: frame 135: lost synchronization error: frame 140: lost synchronization error: frame 145: lost synchronization error: frame 150: bad scalefactor index error: frame 154: lost synchronization error: frame 159: lost synchronization error: frame 164: lost synchronization error: frame 169: lost synchronization error: frame 174: lost synchronization error: frame 179: lost synchronization error: frame 183: lost synchronization error: frame 184: forbidden bit allocation value error: frame 185: CRC check failed output: no channel selected for dual channel; using first output: mono output not available; forcing stereo error: frame 190: lost synchronization error: frame 195: lost synchronization error: frame 200: lost synchronization error: frame 205: lost synchronization error: frame 207: bad scalefactor index error: frame 212: lost synchronization error: frame 216: lost synchronization error: frame 221: lost synchronization error: frame 226: lost synchronization error: frame 231: lost synchronization error: frame 236: lost synchronization error: frame 241: bad scalefactor index error: frame 245: lost synchronization error: frame 250: lost synchronization error: frame 255: lost synchronization error: frame 260: lost synchronization error: frame 261: forbidden bit allocation value error: frame 262: forbidden bit allocation value error: frame 263: forbidden bit allocation value error: frame 264: forbidden bit allocation value error: frame 265: forbidden bit allocation value error: frame 266: forbidden bit allocation value error: frame 267: forbidden bit allocation value error: frame 268: forbidden bit allocation value error: frame 269: forbidden bit allocation value error: frame 270: forbidden bit allocation value error: frame 271: forbidden bit allocation value error: frame 272: forbidden bit allocation value error: frame 273: forbidden bit allocation value error: frame 274: forbidden bit allocation value error: frame 275: forbidden bit allocation value error: frame 276: forbidden bit allocation value error: frame 277: forbidden bit allocation value error: frame 278: forbidden bit allocation value error: frame 279: forbidden bit allocation value error: frame 280: forbidden bit allocation value error: frame 281: forbidden bit allocation value error: frame 282: forbidden bit allocation value error: frame 283: forbidden bit allocation value error: frame 284: forbidden bit allocation value error: frame 285: forbidden bit allocation value error: frame 286: forbidden bit allocation value error: frame 287: forbidden bit allocation value 255 frames decoded (0:00:06.6), +18.0 dB peak amplitude, 6965 clipped samples Can anyone tell me what I'm doing wrong? Wrong app? Wrong config? ...? All comments welcome Best wishes Ivan -- ==== ==== ==== ==== ==== Ivan A. Uemlianin, PhD Research & Development Jura Technology Ltd mailto:ivan@jurakm.com http://www.jurakm.com ==== ==== ==== ==== ====

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Re: Direct Sound
by Unorthodox Behaviour 14 Dec '02

14 Dec '02

Dear Steve, My Hercules Game Theater XP is also working perfectly (meaning that I hear sound loud and clear) when MAD is set at 24 bits (the card's chipset features a 20 bit D/A converter). The issue is what the soundcard is actually outputting: 16 bit, 20 (my card)or 24 bit (your card)? I mean that MAD surely outputs 24 bits, but are these actually getting through the OS/drivers/hardware to the D/A? My humble opinion is that without *external* measurement (bitscope, oscilloscope, 2nd PC with 24/96 card really working at 24 bits to record your output) one cannot tell. __________________________________________________ Do you Yahoo!? New DSL Internet Access from SBC & Yahoo! http://sbc.yahoo.com

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Direct Sound
by Steve Roth 13 Dec '02

13 Dec '02

[I hope this doesn't fall into the "Well DUH, we all knew that" category.] Visited Rob's MAD web page to see if there was anything new and noticed this: "24-bit output is not reported to work with the DirectSound plug-in." Well, let me make a report that it does indeed work with Direct Sound. I had done a fresh install of W2k and SP3 a couple of weeks ago on my multimedia PC. Loaded the latest VIA 4-in-1 driver support. The only sound device is M-Audio 24/96. Fresh install of Winamp 2.81. Went into Winamp preferences and set up MAD @ 24 bits. Output apparently defaulted to DS (I didn't check it at the time.), but everything worked perfectly. I became curious what Winamp was using when I saw the above comment about DS, so I looked and sure enough DS was enabled and operational. To double check, I removed "out_wave.dll" and it still functioned normally. I've also confirmed that MAD is operational by the "active" checkbox and by playing with the title options.

1 0

mp3pro support?
by Edwin Eefting 05 Dec '02

05 Dec '02

Will mad have mp3pro support, or is mp3pro an extention to the iso standard, and thus not iso compliant? thanks! -- //||\\ Edwin Eefting || || || DatuX - Linux solutions and innovations \\||// http://www.datux.nl

3 3

How much for use in a commercial App?
by Ismal Ibric 28 Nov '02

28 Nov '02

Rob, I'm working on a DLL that uses DirectSound to play Wav files and was thinking of selling it as shareware. How much does a custom license cost? Thanks, Izzy. (aka I. Ibric) ___________________________________________________________ Sent by ePrompter, the premier email notification software. Free download at http://www.ePrompter.com. _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963

1 0

mp3 import for sweep -- please test :)
by Conrad Parker 14 Oct '02

14 Oct '02

Hi, I've just coded up some mp3 importing to sweep using libmad, so if you want to throw scrubby at your old mp3 collection, now's your chance. mad people: sweep is a Linux/BSD sound editor with vinyly scrubbing, and its fun to dj with; more info at http://sweep.sourceforge.net/ this is a prerelese for all ye elite hackers, this probably needs to be tweaked a little before its unleashed (later this week ?!) so, please test it till your speakers explode and if it breaks please help fix it (eg. show how to reproduce it, supply the file, or even fix the code): http://www.metadecks.org/software/sweep/download/sweep-0.5.7-mad1.tar.gz otherwise, just go crazy with it. Seems to be working ok here so please give it a whirl. non-mad ppl: you need libmad libs and headers (libmad-devel) installed, otherwise the configure script will simply not include mp3 support ... if libmad isn't in your distro it's at: http://www.mars.org/home/rob/proj/mpeg/ enjoy :) Conrad.

1 0

why is the HQ dsp option unavail? has this not been implemented?
by bangsailio 08 Oct '02

08 Oct '02

Also, I was wondering if there was such a output plugin that supports 96khz output. I found one, but then again, I want to know if there are more out there. shawn

3 3

mad error
by Jesse 01 Oct '02

01 Oct '02

Hello, I'm getting a run time error from layer3.c line 2461: "'ptr' used without being defined.' I'm using win32 and the libmad project to link to. Any ideas what could be causing this? thanks for any ideas and the cool lib, jesse

1 0

for Mac
by Mete BALCI 30 Jul '02

30 Jul '02

Hi, Is there a way to use MAD on Macs? on PowerPC ? --- Mete

2 1

just installed latest ver of MAD
by Markus Sjöström 27 Jun '02

27 Jun '02

i don't think my winamp (2.8) is using the mad plugin. i've deleted the in_mp3.dll file, i've changed the MP31, MP32, MP3 but when i change to 32bit in the mad plugin i still can play mp3s, and that shouldn't work caus i have a shitty built-in soundcard. i've also tried to 'channels' to 'right' only, but still playin in both speakers. how can i see for sure that MAD is used? Regards Markus Sweden

2 1

Re: EgoSys U24
by Unorthodox Behaviour 20 Jun '02

20 Jun '02

Very good card but quite expensive. I just discovered OPTOplay in http://www.audiotrak.net/eng/index.html The web page says: "suggested US retail price (net): $69.00 US" which is equivalent to 71.5 euros... All I can say is "whoa!!!" (now rushing to search online shops!!!) If any MAD-user knows an even cheaper 24 bit solution, please inform us - it seems that these new USB devices are the future of digital audio "cards". I expect that costs for these gadgets will reduce dramatically since there is no PCI bridge chip involved, no big PCB etc - just the DSP chip, audio stream to serial data converters and plastic parts. UB __________________________________________________ Do You Yahoo!? LAUNCH - Your Yahoo! Music Experience http://launch.yahoo.com

2 3

EgoSys U24
by Saboteur 19 Jun '02

19 Jun '02

I don't know if this is inappropriate for the mailing list but since I'm not affiliated w/ the company I think it's a good service for the MAD users. I just bought this little gizmo and am currently testing it w/ the latest MAD plugin and the out_wav from http://www.blorp.com/~peter/ it works like a charm w/ 24bit output and the sound is awesome. it also comes w/ the ASIO low-latency USB audio driver from www.usbaudio.com . I can't wait to connect it to my headphone amp at home and try it w/ my Grados :-)) for all of you in northern america you can use the coupon at www.tracertek.com . for more info http://egosys.net/1-pro/eng/product/u24/u24.html ciao, g. Waveterminal U2A 24 bit USB Sound Module Now Shipping. There has been quite a bit of pent up demand for this product. It was initially scheduled to ship in January and much like all engineers, they missed by just a tiny bit. But the wait is certainly worth it. This high quality USB device will deliver the great USB portable audio at only $239. We had advertised this unit at $219 before we knew the real pricing, so as a special bonus to our newsletter readers, you can now order this product for $20 off at $219 by simply going to our store by July 1, 2002 and using this coupon when you place the order. The U2A is at http://store.yahoo.com/tracertek/wavu24comsoo.html and the coupon number is "newsu2a101".

1 0

Re: Vortex2 and >16 bits playback
by Unorthodox Behaviour 25 May '02

25 May '02

I don't know what your soundard DSP chip is, but I have a Hercules/Guillemot Game Theatre XP which features 20 bit playback. Outwave.dll works for me in both normal and alternate settings for 24 bits, that is I am able to listen to Line Out. So I guess that in my case the 24 bits MAD sends are accepted by the soundcard, processed somehow and outputted. However, I cannot tell whether what I hear is 20 bits, or even the 24 bits downsampled to 16 (!!). So far, nobody I asked could find me a procedure without using an oscilloscope to find out what the real output is. So I'm afraid that even if you manage to get sound out of winamp+MAD to the digital outputs, you won't have any means knowing whether this output is 16, 20 or 24 bits. Perhaps some experts can enlighten us ? In any case I'd recommend mailing Peter Pawlowski - he should know whether his plugin is compatible to your card. __________________________________________________ Do You Yahoo!? Yahoo! - Official partner of 2002 FIFA World Cup http://fifaworldcup.yahoo.com

1 0

Vortex2 and >16 bits playback
by kis＠aha.ru 23 May '02

23 May '02

Hello, I've read all the mailing list archive, and noticed that some people found out how to use MAD at 24 bits (with effective 20) on Vortex2-based cards. But I see no exact instruction on how to do it... My setup: WinAmp 2.80 MAD 0.14.2b out_wave.dll v. 1.03 from Peter Pawlowski Windows 98 SE Aureal SQ Digital (Vortex2 chip, similar to SQ2500, but with optical S/PDIF output) Drivers ver. 4.06.2050 S/PDIF is turned on (actually, I listen via an external DAC) AFAIK there is no way to use WDM driver under Windows 98SE (or at least it is not recommended). When I try to use MAD with 24 bits resolution, I get an error message from the out_wave that this PCM format (44100 Hz, 24 bits, 2 channels) is not supported. So what? How did you manage to work it out? Thanks in advance, Konstantin

1 0

Are there any plans to port the MAD plugin to Sonique 1.9x?
by Roj 24 Apr '02

24 Apr '02

Thanks... Roj "Faith manages." jms

2 4

How to use Libmad?
by naso 24 Apr '02

24 Apr '02

Hi everybody, I'm Simone from Genoa (Italy). I'd like to find a reference guide about libmad. Can you halp me? Thank you.

2 1

20 bit playback and DSP plugins
by Unorthodox Behaviour 16 Apr '02

16 Apr '02

Greetings to Rob and the list! I've been lurking for a while but now I'd like some advice. For the lack of money for a 24/96 card, I just bought a Hercules (ex Guillemot) Game Theatre XP 6.1 card, which features the Crystal CS4630 DSP with 18-bit ADC and 20-bit DAC. (It's the same chip as the Videologic Sonic Fury, and the Turtle Beach Santacruz) Using Mad I can listen to music when I set Mad's resolution in both 24 and 32 bits, when the output plugin is either wave out (Winamp critical update installed). With the directx output plug, I get white noise in both 24 and 32 bits. Question 1 is: Is there a way to certify that the resulting sound is taking advantage of the 20 bit resolution of the card and that it's not just a dithered to 16 bits sound ? (provided of course that the card's drivers do actually support these 20 bits of the ADC...) Question 2 is: Does anybody know of any DSP music enhancer Winamp plug that would support more than 16 bits ? The ones I tried did not work. I know some of you have 24/96 cards and so I expect you might use DSP plugins. Thanks in advance for any tips! By the way, and for your information, Terratec will soon produce a new budget card, the DMX 6fire LT priced below 150 euros. It's a full 5.1 channel 24/96 card (all channels being 5.1) with the same specs as the existing 6fire but without the breakout box - a definite best buy. Amazon.de and computeruniverse.com already offer it as a preorder. Of course i'm not in any way affiliated to these companies, it's just that this will be my next card - cheaper than any 24/96 stereo channel card, even Terratec's. Needless to say, since the time I left the list some months ago, I have recommended Mad to some 100+ people who use it daily. Moreover, Mad is used some 10 hours per day on my PC... Long live MAD ! __________________________________________________ Do You Yahoo!? Yahoo! Tax Center - online filing with TurboTax http://taxes.yahoo.com/

3 3

rendering using libid3tag
by Dominic Mazzoni 11 Apr '02

11 Apr '02

Hi, In developing Audacity (a cross-platform audio editor), I've continued to have nothing but good experiences with libmad. I recently added support for 24-bit and 32-bit samples in Audacity, and it's great that using libmad I can take advantage of these sample formats. I'd like to also switch over to using libid3tag. We had been using id3lib, but it's buggy and has trouble compiling on a lot of systems. Our needs are minimal. We're not trying to be the ultimate ID3 editor; we just want to be able to read and write the most common tags for convenience. I didn't see any documentation on libid3tag...did I miss something? The sample code only uses a small subset of its features. Anyway, I was able to figure out how to use libid3tag to import tags from a file. But I'm stuck trying to render them using id3_tag_render. 1. First of all, the only way I was able to figure out to add new frames was to muck around in the data structures. Is this correct? It might be nice if there were more API functions to make it easier... For example, id3lib has a "simple API" which provides functions that let you add and/or query the most common tags (Title, Artist, etc.), without mucking in the data structure at all. 2. What's the proper way to convert an ASCII string to an id3_ucs4_t string? (I used a for loop, casting chars to longs...is that OK?) 3. When I try to enable the ID3_TAG_OPTION_ID3V1, the rendering fails. Is it supported? 4. Finally, I do get some bytes back from id3_tag_render without the ID3V1 option, but when I prepend them to the beginning of my MP3 file, programs don't recognize them. In fact even libid3tag doesn't even load any tags from the resulting file. Is there something else I need to do to the id3_tag_render output? I'm including some of my source code below. Thanks for any help anyone can offer. - Dominic struct id3_frame *MakeID3Frame(const char *name, const char *data) { struct id3_frame *frame; union id3_field *field; id3_ucs4_t *ustr; int i; ustr = (id3_ucs4_t *)malloc(sizeof(id3_ucs4_t)*(strlen(data)+1)); for(i=0; i<strlen(data)+1; i++) ustr[i] = (id3_ucs4_t)data[i]; field = (union id3_field *)calloc(1, sizeof(union id3_field)); field->type = ID3_FIELD_TYPE_STRINGLIST; id3_field_setstrings(field, 1, &ustr); frame = (struct id3_frame *)calloc(1, sizeof(struct id3_frame)); strncpy(frame->id, name, 4); frame->nfields = 1; frame->fields = field; return frame; } // returns buffer len; caller frees int Tags::ExportID3(char **buffer, bool *endOfFile) { struct id3_tag *tp; struct id3_frame *frame; tp = id3_tag_new(); id3_tag_attachframe(tp, MakeID3Frame(ID3_FRAME_TITLE, mTitle)); id3_tag_attachframe(tp, MakeID3Frame(ID3_FRAME_ARTIST, mArtist)); id3_tag_attachframe(tp, MakeID3Frame(ID3_FRAME_ALBUM, mAlbum)); if (mID3V2) { tp->options |= ID3_TAG_OPTION_COMPRESSION; *endOfFile = false; } else { tp->options |= ID3_TAG_OPTION_ID3V1; *endOfFile = true; } id3_length_t len; len = id3_tag_render(tp, 0); *buffer = (char *)malloc(len); len = id3_tag_render(tp, (id3_byte_t *)*buffer); id3_tag_delete(tp); return len; }

2 1

Re: mad-user digest, Vol 1 #121 - 2 msgs
by Benjamin Lebsanft 03 Apr '02

03 Apr '02

That's what i meant, sorry for the incorrectness :) > I just think it's the way that the MAD plugin was written is at fault, not MAD > itself. > > Benjamin Lebsanft wrote: > > > But when MAD is enabled vorbis streaming does not work, and if its disabled > > it does. So it must be MAD's fault...

1 0

Re: mad-user digest, Vol 1 #120 - 2 msgs
by Benjamin Lebsanft 02 Apr '02

02 Apr '02

But when MAD is enabled vorbis streaming does not work, and if its disabled it does. So it must be MAD's fault...

2 1

Re: MAD and streaming OGG
by Saboteur 01 Apr '02

01 Apr '02

well this is a different problem though, since I'm not talking about an mp3 stream, MAD --as far as I know-- doesn't decode OGG files so I really don't see the reason why it should interact with the Nullsoft in_vorbis plugin. g. > > that's not new. > > For some reason, MAD for winamp has never worked with streaming > anything. > > Usually, I forget to disable it when I want to listen to some > streaming mp3.... > > Too bad MAD can't just "pass" it on to another plugin if it doesn't > support it. > > G.

2 1

MAD and streaming OGG
by Saboteur 01 Apr '02

01 Apr '02

Hi, I was trying the new OGG streams for BBC1 in winamp 2.78 ( http://support.bbc.co.uk/ogg/ ) and noticed that if MAD is enabled they won't work. I'm not saying it's a MAD problem especially considering the early stage of the vorbis plugins and especially the streaming support. I thought I'd mention it here though since I see no reason why the 2 plugins should interact (and clash). I have the latest versions for both plugins. thanks, g.

2 1

xmms-mad and mad_timer_zero
by John Utz 16 Mar '02

16 Mar '02

hi; this error has been defeating me /usr/local/lib/xmms/Input/libxmmsmad.so: Undefined symbol "mad_timer_zero" its in mad.h as an extern const mad_timer_t i cant find for the life of me where the thing is actually supposed to be resolved! it's obviously not getting resolved or else the loader wouldnt be complaining, right :-) what to do? tnx! johnu -- John L. Utz III john(a)utzweb.net Idiocy is the Impulse Function in the Convolution of Life

1 0

Re: [mad-user] MP3 sampled by 22.05kHz, mono or stero
by weipyy 28 Feb '02

28 Feb '02

> Dear davidlee: You can resample the mp3 form 44.1khz to 22.05khz by mad BRs wei ping > > > > > > > Dear All: > Because I should do same test, I need some > MP3s with 22.05kHz sample frequency stero or mono. But after I got some MP3s, I > found all of them are sampled by 44.1kHz, that's to fast for my poor MP3 > hardware. So I wander if anyone can tell where I can find MP3 sample by 22.05Khz > 16bit mono or stero. I appreciate anyone's help > > > > > > BEST REGARD > > > > > davidlee > > ÀîŽóŒœ > > Email: davidlee(a)gv.com.tw >

1 0

MP3 sampled by 22.05kHz, mono or stero
by davidlee 28 Feb '02

28 Feb '02

Dear All: Because I should do same test, I need some MP3s with 22.05kHz sample frequency stero or mono. But after I got some MP3s, I found all of them are sampled by 44.1kHz, that's to fast for my poor MP3 hardware. So I wander if anyone can tell where I can find MP3 sample by 22.05Khz 16bit mono or stero. I appreciate anyone's help BEST REGARD davidlee 李大冀 Email: davidlee(a)gv.com.tw

2 1

Re: [mad-user] madplay control
by Rob Leslie 26 Feb '02

26 Feb '02

On Tuesday, February 26, 2002, at 01:13 AM, Nicolas DEVERGE wrote: > I'm a newbie using mad and I wish to control madplay through its stdin, > let's > say, I wish to pause, resume or quit madplay by sending a P, Q or else > character in its standard input with another program (maybe through a > socket), so is it possible with madplay ? And how works the tty-control > parameter ? madplay reads keyboard input from its controlling terminal, so to make this work you would need to mess with master and slave PTYs, or use a tool like `expect'. An alternative would be to send signals to the madplay process: SIGSTOP would pause, SIGCONT would resume, and SIGINT would quit. -- Rob Leslie rob(a)mars.org

1 0

Porting MAD to MIPSIII
by davidlee 26 Feb '02

26 Feb '02

Dear All: I have a develope board with MipsIII running at 55.8MHz. I wonder if I can port MAD on it. Or if there is anyon had done it can give me some sugguestion. Best Regard 李大冀 Email: davidlee(a)gv.com.tw

2 1

madplay control
by Nicolas DEVERGE 26 Feb '02

26 Feb '02

Hi, I'm a newbie using mad and I wish to control madplay through its stdin, let's say, I wish to pause, resume or quit madplay by sending a P, Q or else character in its standard input with another program (maybe through a socket), so is it possible with madplay ? And how works the tty-control parameter ? Thanks, Nico

1 0

alternative idct implementation question
by Yuxuan Mao 25 Feb '02

25 Feb '02

Hi, I'm doing a school project on mp3 decoders. Other teams in my class is proposing to do an 8-bit cosine lookup table implementation of the idct. Does anyone know if this will work? Yuxuan

4 3

Huuuuuuuge Improvement to My Stereo
by John van Ommen 22 Feb '02

22 Feb '02

If any of you do a lot of listening at your PC, I ran into something today that may be of interest. I've been listening to MP3s on the command line using madplay.exe, cuz winamp sounded like ass, even with the MAD plugin. It turns out that a upgrade of wimamp had replaced my "wave out" output plugin in Winamp. As far as I can tell, the MAD input pluging was outputting a 24bit stream, and I think the wave out plugin was truncating it down to 16bit. Nonetheless, I fixed the output plugin, and the difference is h-u-g-e. 24/96 soundcards playing MP3s upsampled to 24/48 really give CDs a run for their money. Especially at high volumes, the extra dynamics of 24bit are clear, and it seems like you can see 'deeper' into the recording as well. John van Ommen

9 9

only mpeg1 layer 3 code
by Yuxuan Mao 22 Feb '02

22 Feb '02

Hi, We're doing a school project on mp3 decoding with a motorola micro-controller and an altera fpga. We're only interested in mpeg1 layer 3 decoding. So we want to trim the MAD distribution down to only the relevant files pertaining to that. But we're kind of lost on that. Can someone help?

1 0

installation help
by Yuxuan Mao 22 Feb '02

22 Feb '02

Hi, We're doing a school project on building a mp3 decoder on a Motorola processor and an Altera FPGA. We'd like to use MAD because it appears to be the only open source integer decoder available on the net. But when we try to install it on our school unix account by typing ./configure, it gives us this error: ./configure: Permission denied. Can someone help us?

2 1

Re: [mad-user] RE: Huuuuuuuge Improvement to My Stereo
by Andrew Burgess 21 Feb '02

21 Feb '02

>With upsampling, you can interpolate that waveform to *approximate* a sine >wave. Obviously, the best solution would be to do the original recording at >a extremely high sampling rate. I don't think this explains it. A 20KHz square wave sounds like a 20KHz sine wave on your stereo. This is because, in the frequency domain, a square wave is the same as a sum of (even or odd -- I forget, probably odd) harmonics. The first harmonic is 40KHz which your stereo cannot reproduce. The 2nd is 80KHz, etc. All DACs also have low pass filters on the output to attenuate these unwanted harmonics. I don't disbelieve that it sounds better but I don't think this explains it. Andrew

1 0

Is there any WINxx SDK for MAD ???
by S.Sekar 17 Feb '02

17 Feb '02

Hi, Is there any WINxx SDK for MAD. Currently I am using XAUDIO and want to s/o to MAD. How to go about it ??? TIA Yours. #407, 'B' Block, Sterling Park, Kodiga Halli, S.Sekar. Sanjeevini Nagar, Bangalore - 560 092. India. __________________________________________________________ E-Mail : sekar(a)vsnl.com (or) ssekar(a)cheerful.com sekar(a)softhome.net (or) ssekar(a)hotmail.com Phone : 91(80)3636804 (R) 91(80)3412700, 3412792 (O). __________________________________________________________

2 1

Problems when porting to MIPS processor using Greenhills compiler
by Jeroen T 15 Feb '02

15 Feb '02

Hi. I got problems using MADLIB with a MIPS4111 processor and the Greenhills compiler. The compiler is doing OK, and also the decoder seems to do something. When I decode a 128 kbits MP3 containing a stereo 440Hz sinus, I get strange results. The output contains sinus wave forms in multiple frequencies and amplitudes, the output samples get even out off boundaries! I use the FPM_DEFAULT to be 100% compatible. When compiling the MADLIB on the PC with FPM_DEFAULT everything is OK. Does anyone has ported the MADLIB to a MIPS succesfully? With which compiler? Or maybe someone has a suggestion which (free) compiler running on DOS/Windows I can use to compile the MADLIB? Hope someone can help me out. Thanks. Jeroen _________________________________________________________________ Chat with friends online, try MSN Messenger: http://messenger.msn.com

1 0

Re: [mad-user] minimad.c
by Gareth 10 Feb '02

10 Feb '02

Thanks, that worked fine. The only thing is minimad seems to do nothing, I tried ./minimad and it just goes straight back to the command prompt. I tried giving it a filename, ./minimad track.mp3, and the same thing happens. Can you give me some pointers as to how i start using it, or is there some basic documentation that i can read up on it? On 10 Feb 2002 11:50:41 -0500, Mathieu Dube wrote: > you need to tell it to use libmad > > so do gcc minimad.c -o minimad -lmad > > -Mat > On Sunday 10 February 2002 07:50, you wrote: > > Hi, > > I have compiled mad-0.13.0b which went off fine but i am interested in > > playing with minimad.c. The problem is i can't see how to compile it. I > > tried 'gcc minimad.c' but it can't find mad.h. How do i tell the compiler > > how to find all the necessary files? I am using mandrake 8.0. > > > > Thanks. >

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Re: [mad-user] Use on Dreamcast/Linux
by Marcus Kwok 10 Feb '02

10 Feb '02

On Sun, Dec 23, 2001 at 11:01:18PM +0000, Adrian McMenamin wrote: > On Sunday 23 Dec 2001 6:57 pm, you wrote: > > On Sun, Dec 23, 2001 at 01:45:07PM +0000, Adrian McMenamin wrote: > > > Anybody know of a test suite of MP3s that I could use to try to isolate > > > the problem? > > > > ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg1/compliance/ > > > > This page has some test patterns that might be helpful. Good luck. > > Thanks for this - but these were still not much good. None played but that > might have been for a number of different reasons. All the layer III examples > appeared to be MPEG 1s - I really need a range of MPEG 2s as well. ftp://ftp.tnt.uni-hannover.de/pub/MPEG/audio/mpeg2/ This has some mpeg 2 test patterns as well. I don't think they were meant to be played, but rather I think they are there to make sure your decoder works properly. Like, you decode the files and compare the output to another decoder that you know works properly. Sorry if this isn't what you were looking for, but I hope it helps. -- Marcus B. Kwok | 338178 Georgia Tech Station | AOL Instant Messenger: SLuGHEAd78 Atlanta, GA 30332 | ICQ: 20039767 http://gtf.org/ricecake/ | ricecake(a)gtf.org

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precision/quality mode - Winamp
by George Pantazopoulos 08 Feb '02

08 Feb '02

Heya Rob, With the popularity of fast CPU's (a 1 GHz Athlon can now be had for $70) , would you please bump the winamp MAD decoder up to full accuracy decoding. Im sure the extra CPU time consumed will be negligible. Also you could consider adding even higher quality and accuracy options now. Thanks, George Pantazopoulos

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streaming audio support - fix?
by George Pantazopoulos 08 Feb '02

08 Feb '02

Heya Rob, Do you happen to have a build of MAD for winamp that works correctly with shoutcast streams? (ie doesnt drop out) The one I have now is 14.2b Thanks, George Pantazopoulos

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ID3 Tags
by Lindsay Berry 07 Feb '02

07 Feb '02

Is anyone else having a problem with MAD reading the ID3 tags on their MP3s or is it just me? Always the song title. Thanks..

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24/48 MP3s
by John van Ommen 07 Feb '02

07 Feb '02

Anyone know why the LINUX port of madplay can't paly 24/48 MP3s, but the Wintel version can? I upsample all my mp3s from 16/44 to 24/48 to take advantage of the wider bandwidth of modern soundcards. John

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Re: [mad-user] 24/48 MP3s
by John van Ommen 07 Feb '02

07 Feb '02

Have you ever HEARD a upsampler? Have you compared a CD upsampled to 24/48? If you haven't, I suggest you keep your (uninformed) opinions to yourself until you learn more about hi-fidelity. John P.S. If you took a VHS signal sampled at 320x240, and interpolated it to 1280x960, it *would* look better. That's exactly what I'm doing with the audio signal. Hearing is believing, if you have a 24/96 sound card, I'll send you the scripts and you can try it yourself. On Wednesday 06 February 2002 10:15 pm, you wrote: > Upsampling doesn't improve your quality at all...you may also lose some > quality depending on the algorithm used. Tell me this...if you put a VHS > video on DVD, is it gonna look like DVD? > > Ishaan > > ----- Original Message ----- > From: "John van Ommen" <jvanommen(a)verio.net> > To: <mad-user(a)lists.mars.org> > Sent: Wednesday, February 06, 2002 4:14 PM > Subject: [mad-user] 24/48 MP3s > > > Anyone know why the LINUX port of madplay can't paly 24/48 MP3s, but the > > Wintel version can? > > > > I upsample all my mp3s from 16/44 to 24/48 to take advantage of the wider > > bandwidth of modern soundcards. > > > > John

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Re: mad-user digest, Vol 1 #102 - 4 msgs
by Dmitry 06 Feb '02

06 Feb '02

Hello mad-user-request, hey, guys! is it possible not to use html in your mails? 1) it's completely impossible to read digest!!! 2) not all mail-readers understand html format PLEASE USE PLAIN TEXT Best regards, Dmitry mail to: kuts(a)jthz.com http://mitiok.cjb.net/

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Frame skipping
by Mathieu Dube 06 Feb '02

06 Feb '02

Hi, is there a way one can skip a frame in a stream instead of doing mad_frame_decode. Coz I need to make a jump ahead but it takes too much time if I just wait for the say 10 seconds to pass without writing to /dev/dsp but still decoding. I dont know if I attacking the problem in a good way thus any suggestions are welcome. -Mat

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Doc
by Mathieu Dube 01 Feb '02

01 Feb '02

Hi, is there some kind of documentation on how to use mad for decoding mp3s or is there only the code and headers(mad.h)? Thanks -Mat

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24-bit and 32-bit output works!
by George Pantazopoulos 26 Jan '02

26 Jan '02

Finally, Peter Pawlowski's waveout 1.0b9 waveout plugin works perfectly with 24 or 32-bit sound output! Its found in the Critical Updates for Winamp link on this page: http://www.blorp.com/~peter/misc_wa.htm Enjoy! George Pantazopoulos

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RE: [mad-user] 24 and 32 bit
by Jens Rex Bræmhøj 17 Jan '02

17 Jan '02

> From: Gene [mailto:geneh@softhome.net] > > for some reason, after my reinstall, 24 and 32 bit doesn't > work anymore (gives an > error). > > Using 0.14.2b What OS? What hardware? What sound card? What drivers? What error messages? You really need to be a lot more specific, for us to be of any help at all.

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24bit Output in WinXP on a MX300
by Geoff 17 Jan '02

17 Jan '02

Has anybody gotten this combo to work? I ask because 24bit output works for MX300 (and I assume all Aureal cards), under WinME. Any help would be great Thanks, Geoff

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More on Dreamcast/Linux
by Adrian McMenamin 06 Jan '02

06 Jan '02

Solved my general problem with building applications for this platform (including madplay) by rebuilding tools and general set up. But now I (& others) have a different problem - or rather two problems. 1. Firstly, madplay never recognises that the DC and driver are capable of stereo output. This is not a problem for any other application I have tested - any body else have this problem or know how it might be fixed? 2. Secondly the output is "doubled up" - ie each fragment is played twice - meaning each MP3 takes twice as long to play and comes out in a way not all expected - any suggestions? I don't have this problem with, say mpg123, which plays output linerally. Thanks Adrian

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RE: [mad-user] hey! new to this and I wanna say
by John van Ommen 02 Jan '02

02 Jan '02

are you playing them back in Wnamp? I have a Athlon 1333, and I have to use the command line version of madplay. Winamp can't keep up with all the data, it stops and stutters. What's the sonic foundry program you're using? Is there a noticeable improvement? Check out SSRC, it's pretty outrageous, especially if you have a lot of hard drive space ;) -----Original Message----- From: chris georgeson [mailto:hideho55@yahoo.com] Sent: Friday, December 28, 2001 4:50 AM To: John van Ommen Subject: RE: [mad-user] hey! new to this and I wanna say Cool, John, I"m using a demo from sonic foundry, batch converter. chris --- John van Ommen <jvanommen(a)verio.net> wrote: > I wrote a script that upsamples 16/44 MP3s to 24bit > 48khz MP3s. These MP3s > work in my car (24/96 JVC KD-SH99) and on my PC > (24/96 Echo MIA soundcard.) > > The script is quite elegant, I can convert a few > hundred MP3s overnight. > > Probably doesn't sound as good as a kilobuck > audiophile upsampler, but I do > hear a slight improvement in the treble on the 24/48 > mp3s. > > John > > -----Original Message----- > From: mad-user-admin(a)lists.mars.org > [mailto:mad-user-admin@lists.mars.org]On Behalf Of > chris georgeson > Sent: Thursday, December 27, 2001 10:28 PM > To: mad-user(a)lists.mars.org > Subject: [mad-user] hey! new to this and I wanna say > > > anyone else have the M Delta 24/96 sound card, and > wanting to convert WAVs to 96khz, then use MAD to > play > em back as 24 bit ? > > on my AMD/VIA pc, I'm getting halting or stuttering > audio. M Delta site sez the prob, in recording, is > that the USB port is disabled, and the VIA AC 97 > audio > is enabled. . . . but fixing these doesn't do it for > my machine. > > anyone have an idea? > > tia > > __________________________________________________ > Do You Yahoo!? > Send your FREE holiday greetings online! > http://greetings.yahoo.com > __________________________________________________ Do You Yahoo!? Send your FREE holiday greetings online! http://greetings.yahoo.com

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Re: [mad-user] hey! new to this and I wanna say
by Rob Leslie 29 Dec '01

29 Dec '01

John van Ommen wrote: > I wrote a script that upsamples 16/44 MP3s to 24bit 48khz MP3s. These MP3s > work in my car (24/96 JVC KD-SH99) and on my PC (24/96 Echo MIA soundcard.) > > The script is quite elegant, I can convert a few hundred MP3s overnight. > > Probably doesn't sound as good as a kilobuck audiophile upsampler, but I do > hear a slight improvement in the treble on the 24/48 mp3s. How does your script upsample 16/44 MP3s to 24/48? MP3 files have a fixed sample rate and can be decoded to any precision. -- Rob Leslie rob(a)mars.org

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hey! new to this and I wanna say
by chris georgeson 28 Dec '01

28 Dec '01

anyone else have the M Delta 24/96 sound card, and wanting to convert WAVs to 96khz, then use MAD to play em back as 24 bit ? on my AMD/VIA pc, I'm getting halting or stuttering audio. M Delta site sez the prob, in recording, is that the USB port is disabled, and the VIA AC 97 audio is enabled. . . . but fixing these doesn't do it for my machine. anyone have an idea? tia __________________________________________________ Do You Yahoo!? Send your FREE holiday greetings online! http://greetings.yahoo.com

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Re: mad-user digest, Vol 1 #86 - 3 msgs
by Benjamin Lebsanft 27 Dec '01

27 Dec '01

> > >Message: 2 >From: "Carlos Gonzalez" <carlos.gp(a)corevia.com> >To: <mad-user(a)lists.mars.org> >Date: Sun, 23 Dec 2001 22:12:08 +0100 >Subject: [mad-user] How to equalize 24 bits sound > >Anybody knows how to equalize 24 bit sound in winamp with mad and a >terratec ewx 24/96? Any plugin? > >-- >Carlos Gonzalez http://www.milky.ne.jp/~galaxy/superequ.html perhaps...

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Re: mad-user digest, Vol 1 #88 - 1 msg
by Benjamin Lebsanft 26 Dec '01

26 Dec '01

> >From: "Carlos Gonzalez" <carlos.gp(a)corevia.com> >To: <mad-user(a)lists.mars.org> >Subject: RE: [mad-user] Re: mad-user digest, Vol 1 #86 - 3 msgs >Date: Tue, 25 Dec 2001 23:33:00 +0100 > > > >> Anybody knows how to equalize 24 bit sound in winamp with mad > >> and a terratec ewx 24/96? Any plugin? > > > http://www.milky.ne.jp/~galaxy/superequ.html perhaps... > >It seems to work. Anybody knows any site with documentation about this >equalizer? > >-- >Carlos Gonzalez this is the official site... :) perhaps visit hydrogenaudio.org, shibata san is sometimes there :) Bye Benni

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Use on Dreamcast/Linux
by Adrian McMenamin 23 Dec '01

23 Dec '01

I have compiled mad/madplay on SH4 for Linux (Dreamcast) to test the /dev/dsp driver I am writing for this machine. I can play one MP3 - MPEG II at 22k sampling - everything else seems to result in a "Segmentation fault" (that includes trying to play the one working mp3 with option -v). All the others I have tried are MPEG I and all seem to have 44.1k sampling rates. Questions I have: Anybody know of a test suite of MP3s that I could use to try to isolate the problem? Anybody had a similar problem on a different platform and might be able to tell me what's up? Adrian

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Audigy/24Bit/Mad Plugin
by Fleetwood Mac Attack 07 Dec '01

07 Dec '01

Hi, Just wondering if there is a definitive yes/no, on whether the Audigy can be used in 24 bit mode with the Mad Plugin. I've heard of "wrappers" that can be used to achieve this in NT, is there any for XP and possibly Win 98SE? I haven't as yet been able to download one of these files, as the site I was told about has been taken down. Thanks for any info in advance, Jason

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24 bit and alsa 0.9beta10 [linux]
by Karl Ranseier 05 Dec '01

05 Dec '01

Hi, I've a C-Media PCI CMI8738 (model 37) running with alsa 0.9beta10. I realy would like to get 24 bit on the S/P DIF interface. So I do: >madplay --bit-depth=24 music.mp3 MPEG Audio Decoder 0.14.2 (beta) - Copyright (C) 2000-2001 Robert Leslie output: bit depth 24 not available; using 16 Any ideas? Thanks, Karl __________________________________________________________________ Nokia 5510. Verrücktes Design! Toller Sound! Entdecke und gewinne es auf http://de.promotions.yahoo.com/nokia/ Gewinnspiel endet am 16. Dezember 2001.

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Incorrect Time Detected
by Marcus Kwok 25 Nov '01

25 Nov '01

I have a VBR file that MAD does not detect the time of properly in the file info box. The file is 23.9 MB, and is about 44 minutes long. However, in the file info box, MAD displays the time as around 20 minutes long. I scanned the file with MP3Utility, and the file seems to be in good condition (i.e., not corrupted). The correct time is displayed in the playlist though. -- Marcus B. Kwok | 338178 Georgia Tech Station | AOL Instant Messenger: SLuGHEAd78 Atlanta, GA 30332 | ICQ: 20039767 http://gtf.org/ricecake/ | ricecake(a)gtf.org

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cdda format type error
by ltiu 12 Nov '01

12 Nov '01

Hello, madplay --output=cdda 1.cdda 1.mp3 gives me an error: madplay: unknown output format type for "cdda" I'm trying to convert mp3 to cd audio so that I may burn this to a cd and listen to them in my car's cd player. Any help would be appreciated. Thanks.

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NT Drivers with Delta Dio2496 seems to work
by kaiwei 12 Nov '01

12 Nov '01

Hi all, I *seem* to have gotten the Delta Dio 2496 working under Windows 2000 & MAD at 24bit with the NT drivers 4.1.17.13. The thing is I'm not sure if it is at 24bit because switching between 24bit & 8 bit, I can't really hear a difference. Moreover, I tried to play a 24bit 96Hz wav file on winamp & it crashed the whole system, this problem is repeatable every time. I recorded digitally to a Minidisc & the minidisc was able to record fine which leads me to suspect that it is still at 16bit because the minidisc has a 16bt DAC only. (maybe I'm wrong here?) Is there any way to check if the output is at 24bit? 1 more thing, I'm reluctant to try the wrapper posted here because under the WDM drivers my audio playback keeps having pops & clicks. I've tried almost every trick in the book, disable hardware acceleration to no avail. but right now under the NT drivers, at least there is no pops & clicks! But is it 24bit? thanks for helping. kaiwei

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NT Drivers with Delta Dio 2496 seems to work
by Geoff 11 Nov '01

11 Nov '01

Just a general pointer, but if you can, try to get the sound card its own IRQ. This solved a couple of minor problems on my Vortex 2 sound card. Geoff

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MAD 0.14.2b available
by Rob Leslie 09 Nov '01

09 Nov '01

MAD version 0.14.2b is now available. See the CHANGES file in the distribution for full details. In summary... This version adds three new output modules to `madplay': - esd: Enlightened Sound Daemon (EsounD) - cdda: CD audio (16-bit 44100 Hz stereo, padded to CD frame boundary, suitable for recording directly to CD) - aiff: Audio IFF Also, two new options: --bit-depth: select the output sample resolution --ignore-crc: decode frames with bad CRCs Finally, for Win32 users, `madplay' no longer requires any Cygwin DLLs at runtime; the built EXE runs stand-alone using Microsoft's Visual C runtime (MSVCRT.DLL). Also, keyboard controls are now supported under Win32. The MAD plug-in for Winamp has one bug fix and a few other changes in preparation for an ID3v2 tag editor, but no major new developments. The source code is here: ftp://ftp.mars.org/pub/mpeg/ The plug-in for Winamp is here: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ Cheers, -- Rob Leslie rob(a)mars.org

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Audigy
by Michael Deinhardt 03 Nov '01

03 Nov '01

Hi Kjetil, the Delta Audiophile is a great card, but I´ve been exactly thinking about those ways you mention: Gaming, EAX Advanced HD, Firewire, huge software bundle for first time users, a bundled game. Again: We don´t want to sell high end cards, we sell for the mass market. And there, those things are more important than 24/96 recording and the like. So, why the doubts? I wouldn´t be in this mailinglist, if I weren´t interested in high-end sound quality and 24Bit playback. But there are thousands of gamers, novices etc. who either just don´t understand our need for sound quality or who just don´t give a damn. Not only due to my job, I always try to understand both sides... Rgds, Michael

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Specific Card
by Kevin Cody Jr 02 Nov '01

02 Nov '01

On Mon, 29 Oct 2001 mad-user-request(a)lists.mars.org wrote: > Anything with the ICEnsemble ICE1712 ENVY24 chip does excellent. This > includes the Terratec EWX 24/96 and M-Audio Audiophile 2496 that George > Pantazopoulos mentioned earlier. M-Audio makes several other boards that > have the ICE1712, such as the Delta DiO 2496, Delta 66, > etc. Hoontech/STAudio also has a board with this chip called the > Soundtrack Audio DSP24 Value. I might be beating a dead horse here, but maybe someone can save me some research. I'm looking for something that will just plain pass through the PCM data without any kind of resampling or mixing, over (preferably) fiber S/PDIF. Therefore I don't much care about the DA's, but high quality AD's would be nice. Balanced inputs would be a cool bonus, but aren't necessary. I guess the point I'm making is that I don't want to go the traditional pro audio route, I want top-end home stereo, because for some reason I like the way they simulate five channels from stereo music sources. They invariably give a more pleasing and natural soundfield than two-channel setups in rooms with shitty acoustics. On the other hand, Pro Logic II is good enough for that, so if anyone knows where I can get the following equipment at a reasonable cost, please pipe up: 24-bit variable sample rate stereo D/A w/toslink input A sound card that can feed it * note: i want toslink because the EMI in here is outrageous Pro Logic II decoder with center/rear tone,delay,level controls (3) 100w @ 8ohm mono amps (2) 50w @ 8ohm mono amps A rack to mount all this stuff in Now, given those parameters, if I'm better off sticking with the SB Live! let me know. Thanks, - Kevin

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Audigy
by Michael Deinhardt 01 Nov '01

01 Nov '01

Funny, how this list comes to live when a topic like Creative pops up :-) The Audigy is not capable of playing back 24 Bit - But we never positioned it to do so. It´s not a pure musician soundcard and it´s not a professional studio soundcard. But it´s got the best gaming engine you´ll find anywhere, you get really good sound quality (Roj, mute the Mic input and you CAN get 100dB) and beginners can try an excellent ASIO implementation, Soundfonts, Midi etc. Btw, the drivers are the best we made so far, and I hate bad drivers as much as you. With the 1394 on top you´ll never find a card like that for such a price. Sure there are better this and better that cards but: The Audigy is for the mass market and you all are obviously not our core target group, so why bother with it? I got an Audigy AND a Terratec EWX 24/96 and I love it. Rgds, Michael Creative Labs Central Europe

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installing MAD
by Markus Sjöström 01 Nov '01

01 Nov '01

I dont understand this in the readme file: "Disable the Nullsoft MPEG audio decoder plug-in by deleting the file association extension list (version 2.7)" can someone please help me? MAD ROKKS!

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Audigy
by Matt Makauskas 30 Oct '01

30 Oct '01

Hello, just a question about the new Creative Audigy cards... are they able to playback the 24-bit audio that MAD produces. I think I'd be be willing to pick one up (the Value OEM is only 110 Canadian $$s) if it could pass the MAD plug-in for WinAMP output out it's SPDIF at 24-bit. thanks, MAtt

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14.1b and streaming
by George Pantazopoulos 28 Oct '01

28 Oct '01

Heya Rob, 14.1b still handles streaming audio (with the usual gaps) even though I have the option unchecked. I have to disable MAD completely to let Nullsoft's streaming support take effect. Btw, I got your postcard, thanks! :) Did my donation have anything to do with the recent flurry of updates? George

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Re: [mad-user] Live! problems with VIA?
by Ian Scott 28 Oct '01

28 Oct '01

At 01:14 PM 10/28/2001 -0500, you wrote: >I've got a Live! MP3+ on a Via board, and haven't had any problems. Well, >not with audio, anyway. I don't know the model number off the top of my >head, but it's the dual 370 that CompUSA was selling for $100. > >What kind of problems should I be looking for? The big problem is with boards that have VIA VT82686B southbridge chips. The SBLive isn't the nicest card with regards to the PCI bus so VIA enabled some workarounds in their chipset. Unfortunately this workaround causes data corruption with the ATA100 controller. It's best demonstrated by trying to copy a large amount of files (>100MB) between two drives. The latest VIA 4in1 drivers should fix this, but no guarantees of course. >Also, while we're on the subject, what soundcard /would/ pass through >24-bit without dicking with the stream somehow? Anything with the ICEnsemble ICE1712 ENVY24 chip does excellent. This includes the Terratec EWX 24/96 and M-Audio Audiophile 2496 that George Pantazopoulos mentioned earlier. M-Audio makes several other boards that have the ICE1712, such as the Delta DiO 2496, Delta 66, etc. Hoontech/STAudio also has a board with this chip called the Soundtrack Audio DSP24 Value. Ian

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Suggestions for MAD plugin
by leolo＠terra.es 28 Oct '01

28 Oct '01

Dear Sir, Will you please add info about where exactly in the file was the MPEG header found? Winamp does this, and it's very nice, because you can easily check which files have id3v2 tags and which ones don't. Also, it's helpful because there are some strange mp3 files which have a RIFF/WAVE header in them, and the MPEG header is usually found in byte 70 or 72 instead of byte 0. And, if you don't mind, it would be great if your plugin had an option to remove that annoying RIFF/WAVE header. Lastly, I'd love another option to remove id3 tags (both version1 and version2) Thanks a million. Best regards.

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Live! problems with VIA?
by Kevin Cody Jr 28 Oct '01

28 Oct '01

On Sun, 28 Oct 2001 mad-user-request(a)lists.mars.org wrote: > Subject: Re: [mad-user] Audigy > Date: Sun, 28 Oct 2001 15:52:43 +1100 > > Problem is, although the card accepts 24-bit in and out it's processed at > 16bit. AFAIK there's no way to bypass this since you're just passing through > data (no need to convert to analogue). Also, you may not be able to get > 24-bit working if you use NT (NT4/2000/XP). > > Otherwise the Audigy is a good buy, particularly as it doesn't have the same > problems with VIA motherboards as the Live does. I've got a Live! MP3+ on a Via board, and haven't had any problems. Well, not with audio, anyway. I don't know the model number off the top of my head, but it's the dual 370 that CompUSA was selling for $100. What kind of problems should I be looking for? Also, while we're on the subject, what soundcard /would/ pass through 24-bit without dicking with the stream somehow? - Kevin Cody Jr

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file length not displayed
by Matthias Pleines 24 Oct '01

24 Oct '01

hi! has anyone the same effect with winamp mad-plugin v0.14.1b? ...or is it a bug? when starting an mp3-file by doubleclicking on it, so that it will be played immediately after the start of winamp, the file length is not displayed in the playlist. Only when I go into the file info dialog window and exit it, the time is displayed. This effect only occurs on the file that will be played on the start of winamp. Greetings Matthias, Germany --- www.Matthias.Pleines.com

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Stream with a different plugin
by Kenneth Oransky 24 Oct '01

24 Oct '01

Hi, I want to play local mp3 files with MAD but I want to get streams with Winamp's plugin. I thought the instructions said to uncheck streams in MAD. I did this but I still get streams through MAD. I have removed the default file extensions in Winamp's plugin. What am I doing wrong? Thanks, Ken

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MAD 0.14.1b available
by Rob Leslie 23 Oct '01

23 Oct '01

MAD version 0.14.1b is now available. Since the release of 0.14.0b, a number of issues were immediately raised that warrant a new release. Directly from the CHANGES file: - Updated config.guess and config.sub to latest upstream versions. - Enabled libtool versioning rather than release numbering. - Renamed `libid3' to `libid3tag' and enabled installation as a separate library. - Enabled the plug-in for Winamp to use libid3tag. - Fixed a NetBSD compatibility issue in audio_sun. - Fixed a FreeBSD compatibility issue in audio_oss. - Modified the audio_null output module to compute peak amplitude. - Changed the peak amplitude computation to avoid evaluating log10(0). - Improved the documentation in minimad.c. - Several other small fixes. The source code is here: ftp://ftp.mars.org/pub/mpeg/ The plug-in for Winamp is here: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ Cheers, -- Rob Leslie rob(a)mars.org

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MAD 0.14.0b available
by Rob Leslie 19 Oct '01

19 Oct '01

MAD version 0.14.0b is now available. Highlights of this release include: - a new ID3 tag manipulation library implementation - a new dithering algorithm for (hopefully) even better sound quality - keyboard controls for `madplay' (see the man page for details) - a fix for `madplay' segfaults on files that are a multiple of 4K - improved code portability, and new MSVC++ project files - other various small code improvements The new ID3 tag manipulation library is the most significant addition. The library has full support for reading ID3v1, ID3v1.1, ID3v2.2, ID3v2.3, and ID3v2.4 tags, as well as support for writing ID3v1, ID3v1.1, and ID3v2.4 tags. Currently the only thing to make use of the new library is `madplay' for displaying tag information, however I also intend to use this as the basis for implementing ID3v2 support in the MAD plug-in for Winamp. Yes, it's coming soon! The new dithering algorithm is intended to address the slight audio deficiencies of the old one. It should produce better sound in theory, but your ears are the best judge. Please test it and share your feedback. The source code is here: ftp://ftp.mars.org/pub/mpeg/ The plug-in for Winamp is here: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/ Cheers, -- Rob Leslie rob(a)mars.org

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compilation problem for arm
by Wayne F Davis 08 Oct '01

08 Oct '01

Hi, I'm trying to compile for the arm, but am having problems: make[2]: Entering directory `/usr/src/build/mad-0.13.0b/libmad' /bin/sh ./libtool --mode=compile /skiff/local/bin/arm-linux-gcc -DHAVE_CONFIG_H -I. -I/usr/src/mad-0.13.0b/libmad -I. -DFPM_ARM -DASO_ZEROCHECK -Wall -m486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -c /usr/src/mad-0.13.0b/libmad/version.c /skiff/local/bin/arm-linux-gcc -DHAVE_CONFIG_H -I. -I/usr/src/mad-0.13.0b/libmad -I. -DFPM_ARM -DASO_ZEROCHECK -Wall -m486 -g -O -fforce-mem -fforce-addr -fthread-jumps -fcse-follow-jumps -fcse-skip-blocks -fexpensive-optimizations -fregmove -fschedule-insns2 -fstrength-reduce -c /usr/src/mad-0.13.0b/libmad/version.c -o version.o cc1: Invalid option `486' make[2]: *** [version.lo] Error 1 make[2]: Leaving directory `/usr/src/build/mad-0.13.0b/libmad' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/src/build/mad-0.13.0b' make: *** [all-recursive-am] Error 2 How do I override the -m486 option? I configured with: /usr/src/mad-0.13.0b/configure --prefix=/skiff/local --target=arm-linux --enable-fpm=arm Thanks. -- Wayne Davis - wfdavis(a)seas.upenn.edu - PGP Key Available

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one more time
by John van Ommen 26 Sep '01

26 Sep '01

>John van Ommen wrote: > >> >> drive, whereas I listen to MP3 CDs on my laptop almost exclusively. And of >> >> course, a hard drive would have *much* lower jitter than a CD-ROM, because >> >> it's a much more stable mechanism. >> >> >> > >> >Ummmm, ever heard of a thing called a "buffer" .... >> Some output devices are better than others. You may have a buffer, but how >> well does it work? You're not seriously saying you can't hear the >> difference between a low-jitter, well engineered CD player and a $80 Wal >> Mart special? My Harmon Kardon only cost $105 but it clearly sounds better >> than your average CD player. But it doesn't sound as good as my Echo MIA. >> According to the documentation on the modest ($200) Monarchy Audio Digital >> Interface Processor, jitter is reduced by about 90%. And believe me, you >> can hear a difference immediately. One thing that is very interesting about >> this change is that it's not very dependent on your speakers. I've hooked >> up the D.I.P. to Cambridge Audio's $80 set of multimedia speakers and heard >> a notable difference (they have a digital input.) Naturally, the difference >> is even more pronounced on my 'real' speakers. >> > >Yes. But why are you claiming that all CD-ROMs have cheap and nasty buffers, >and that all hard disks have top of the line buffers? The purpose of the buffer is to >remove the instability in the mechanical system, and then the timing of the data >is *only* dependant on the read clock to the buffer itself. >I agree, I can hear the difference between a cheap & expensive player, but it >is not due to jitter. There are many more components after the buffer that are >much easier to stuff up. Actually, it's not that hard drives have a better buffer (thought it's a given they have a MUCH larger buffer.) It's that the hard drive mechanism itself is waaaay more stable, and therefore much less likely to require error correction. And therefore, the data being streamed to the buffer is more stable. >> >> >> Bottom line is that your MAD setup may sound good now, but you'd be amazed >> >> how much better it CAN sound. >> > >> >If you're into maximum quality, why are you even bothering with lossy >> >compression? >> > >> The same reason EVERYONE is into MP3s. You can download them for free off >> the internet!? doh! > >Complete with patch and parrot :) >I have personally found mp3's off the net to be of low quality, usually poorly ripped, >and seldom well encoded, even at a decent bit rate! Very dissapointed with net mp3s. >You only get what you pay for! > I get most of my MP3s off of Usenet, and I find that the ppl who are willing to go to the trouble of posting on Usenet actually take a lot of pride in their RIPs. They even boast about what they used to rip & encode. Not to mention some also rip off of vinyl, which is a lot of work. I used to rip my own MP3s, but after 2-3 years of ripping, I'm finished nearly all of my CDs. >> >> I'm being sincere when I say a good MP3 rig can give CDs a run for their >> money. I'll admit they don't have as much resolution, generally. And CDs >> have more 'air' in the soundstage, due to their superior channel seperation. >> But a lot of the irritating aspects of CD playback such as the timing errors >> and the crummy treble is NOT present in a MP3 that's being played off a hard >> drive on a good (24/96) soundcard. Getting the timing right really makes >> the music 'swing', especially if you're into rhythmically intense music, >> like rock & roll, electronica, or hip-hop. > >A *good* mp3 (rare from the 'net) sounds very good. But even with my bunged >out ears, I can hear the artifacts present in a 160kbps mp3 encoded direct from >a CD. That bitrate is slowly becoming more popular on the 'net, but good >encoding is also required! > >What if you rip a .wav from the CD to the hard disk? Does it sound like the >original... or has the 'jitter' disappeared? > I'd expect that it *would* sound better. But I don't want to offer a opinion until I do a A/B comparison. Generally I play CDs on my CD player, and the MP3s on the PC. Same amp & speakers. I'll copy a wav to HD and see if it sounds better than the original CD, that would be educational. >> >> >PS: CD's are mastered for price, not quality. Check out how many newer CD's have >> >*serious* clipping. All the tubes and monster cables you like won't make your >> >end of the chain fix those ruined links at the start (screw-ups - deliberate or >> >otherwise - during mastering). >> I don't agree that clipping ruins a mix. I've clipped mixes on *purpose*, >> because it can produce a lot of interesting effects. I listen to mostly >> electronic music, so I'm pretty comfortable with outrageous mixing. > >Yes! But mixing for effects is different than mixing for a 'minimum' distortion >master. If the original is not clipped, you can always add clipping later to your >hearts content... but not the other way around :)

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RE: [mad-user] Howdy
by Andrew Burgess 26 Sep '01

26 Sep '01

On Tue, 25 Sep 2001, John van Ommen wrote: > I'm glad to hear that MAD sounds good on your PC. You stated that you were > closer to audio nirvana because you had bypassed the DACs in your SB Live. > While improving DACs is certainly a good thing, note that this doesn't > eliminate what is potentially the biggest problem with digital audio, > jitter. > > A definition of jitter: > http://www.feurio.com/English/faq/faq_vocable_jitter.shtml This is jitter during ripping. Ripping software does mutliple reads of the same data area until it finds its place again to eliminate it. See e.g. the cdparanoia docs. > I have a box that re-clocks the digital signal, a Monarchy Audio Digital > Interface Processor. This seems to restore the pace to music, and also > smooths out the treble. I used to use it with my desktop PC, but found that > their was a greater improvement when I hooked it up to my laptop. This is > primarily because my computer's audio colletions is mostly on my 60gig hard > drive, whereas I listen to MP3 CDs on my laptop almost exclusively. And of > course, a hard drive would have *much* lower jitter than a CD-ROM, because > it's a much more stable mechanism. This is different. The mass storage device of an mp3 file is irrelevent as long as it (and the system) is fast enough to keep the sound card fed (or you get dropouts, BIG gaps of silence). The clocking of the DACs occurs on the soundcard and is dependent on the stability of the soundcard's crystal oscillator. As for what you hear, if you are taking digital output from your laptop and reclocking then you are bypassing the laptops DACs completely. You may be hearing low quality DACs or related D/A hardware on the laptop.

1 0

Howdy
by Kevin Cody Jr 26 Sep '01

26 Sep '01

Mad + mpg321 just became my mp3 playback tool of choice based on output quality alone. FYI, my setup is a dual-boot RH7.1/Win98, with an SB Live and the optical i/o daughtercard, feeding into a JVC RX-1028V A/V deck. Under Linux that optical S/PDIF output seems to be a nearly-transparent passthrough, meaning it isn't mixed. I'm not sure if it's the card or ALSA doing the sample rate corrections. Probably the card. This has yielded what I believe is the cleanest possible pathway for digital audio, because it's using the much higher quality DA's in the JVC stereo unit. I believe that for a digital recording, the only point of significant signal loss is the MP3 encoding and decoding. Thanks to MAD, the vast majority of that is now the encoder. Which leads me to the reason for posting: this implies that much of the perceived fidelity loss of MP3's involves flaws in soundcard DA's getting tickled by something in the MP3 algorithms. I'm available for whatever testing anyone can come up with. Keep up the good work! - Kevin Cody Jr

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RE: [mad-user] Howdy
by John van Ommen 26 Sep '01

26 Sep '01

>> drive, whereas I listen to MP3 CDs on my laptop almost exclusively. And of >> course, a hard drive would have *much* lower jitter than a CD-ROM, because >> it's a much more stable mechanism. >> > >Ummmm, ever heard of a thing called a "buffer" .... Some output devices are better than others. You may have a buffer, but how well does it work? You're not seriously saying you can't hear the difference between a low-jitter, well engineered CD player and a $80 Wal Mart special? My Harmon Kardon only cost $105 but it clearly sounds better than your average CD player. But it doesn't sound as good as my Echo MIA. According to the documentation on the modest ($200) Monarchy Audio Digital Interface Processor, jitter is reduced by about 90%. And believe me, you can hear a difference immediately. One thing that is very interesting about this change is that it's not very dependent on your speakers. I've hooked up the D.I.P. to Cambridge Audio's $80 set of multimedia speakers and heard a notable difference (they have a digital input.) Naturally, the difference is even more pronounced on my 'real' speakers. >> Bottom line is that your MAD setup may sound good now, but you'd be amazed >> how much better it CAN sound. > >If you're into maximum quality, why are you even bothering with lossy >compression? > The same reason EVERYONE is into MP3s. You can download them for free off the internet!? doh! I'm being sincere when I say a good MP3 rig can give CDs a run for their money. I'll admit they don't have as much resolution, generally. And CDs have more 'air' in the soundstage, due to their superior channel seperation. But a lot of the irritating aspects of CD playback such as the timing errors and the crummy treble is NOT present in a MP3 that's being played off a hard drive on a good (24/96) soundcard. Getting the timing right really makes the music 'swing', especially if you're into rhythmically intense music, like rock & roll, electronica, or hip-hop. >PS: CD's are mastered for price, not quality. Check out how many newer CD's have > >*serious* clipping. All the tubes and monster cables you like won't make your >end of >the chain fix those ruined links at the start (screw-ups - deliberate or >otherwise - >during mastering). I don't agree that clipping ruins a mix. I've clipped mixes on *purpose*, because it can produce a lot of interesting effects. I listen to mostly electronic music, so I'm pretty comfortable with outrageous mixing. Owen

1 0

RE: [mad-user] Howdy
by John van Ommen 26 Sep '01

26 Sep '01

RE: [mad-user] Howdywww.soundchaser.com I may have paid $200, I can't remember. Last weekend I noticed that CompUSA is stocking a 24/96 manufactured by Creative. It was well under $100, but I have no idea if it's a quality item. The MIA is killer. Also, I am definitely *not* one of those wackos who thinks he can hear a difference between different kinds of speaker cable and interconnect. But the MIA has these giant 1/4" inch connectors that are absolutely beefy, and I definitely observed a noticeable improvement in sound quality when I swapped out the interconnects between my MIA and my amp. This is the first time I've ever noticed an improvement in sound based on such a small change. BTW, I didn't buy the new interconnects because I was trying to 'tweak' my system. I actually bought them because the interconnects I was using with the MIA originally were grafted together, as it's hard to find a 1/4" to RCA cable. I found these at Guitar Center, they're very nice. -----Original Message----- From: Yura Kloubakov [mailto:yura@opmr.com] Sent: Tuesday, September 25, 2001 3:00 PM To: John van Ommen Subject: RE: [mad-user] Howdy Hi John, I wonder where did you get Echo Mia for $150 from. The best price I was able to find is $190 at "The Midi-Store". Thanks, Yura. -----Original Message----- From: mad-user-admin(a)lists.mars.org [mailto:mad-user-admin@lists.mars.org]On Behalf Of John van Ommen Sent: September 25, 2001 16:56 To: Kevin Cody Jr Cc: mad-user(a)lists.mars.org Subject: RE: [mad-user] Howdy I'm glad to hear that MAD sounds good on your PC. You stated that you were closer to audio nirvana because you had bypassed the DACs in your SB Live. While improving DACs is certainly a good thing, note that this doesn't eliminate what is potentially the biggest problem with digital audio, jitter. A definition of jitter: http://www.feurio.com/English/faq/faq_vocable_jitter.shtml I have a box that re-clocks the digital signal, a Monarchy Audio Digital Interface Processor. This seems to restore the pace to music, and also smooths out the treble. I used to use it with my desktop PC, but found that their was a greater improvement when I hooked it up to my laptop. This is primarily because my computer's audio colletions is mostly on my 60gig hard drive, whereas I listen to MP3 CDs on my laptop almost exclusively. And of course, a hard drive would have *much* lower jitter than a CD-ROM, because it's a much more stable mechanism. One of the big problems with the SB Live is that it outputs a 16bit, 48khz signal. So it's doing some really weird resampling in real time, even with 44.1khz audio. I used to use a SB Live, but I found that a cheap $40 Maxi Sound Fortissimo sounded better when using the SPDIF output. The Fortissimo lasted about a year, but I replaced it four months ago with a Echo Mia, which has 24/96 output. For $150, the Echo Mia sounds stellar. Even without a outboard DAC or any kind of re-clocking or upsampling, it sounds excellent. In some respects, I find that MP3s played on the 24bit Echo Mia actually sound superior to CDs played on my Harmon Kardon CD player. Must have something to do with the low jitter and the exceptional DACs in the MIA. Amplification is tubes. Bottom line is that your MAD setup may sound good now, but you'd be amazed how much better it CAN sound. John P.S. If I can find a few minutes, I'm going to try upsampling a CD to 24/96 bit wavs using Cool Edit 2000, compress the wavs, then play the 24/96 MP3s through the MIA. That should really give my CDs a run for their money. Another thought is to do something similar with the audio track of a DVD. -----Original Message----- From: mad-user-admin(a)lists.mars.org [mailto:mad-user-admin@lists.mars.org]On Behalf Of Kevin Cody Jr Sent: Sunday, September 23, 2001 1:02 PM To: mad-user(a)lists.mars.org Subject: [mad-user] Howdy Mad + mpg321 just became my mp3 playback tool of choice based on output quality alone. FYI, my setup is a dual-boot RH7.1/Win98, with an SB Live and the optical i/o daughtercard, feeding into a JVC RX-1028V A/V deck. Under Linux that optical S/PDIF output seems to be a nearly-transparent passthrough, meaning it isn't mixed. I'm not sure if it's the card or ALSA doing the sample rate corrections. Probably the card. This has yielded what I believe is the cleanest possible pathway for digital audio, because it's using the much higher quality DA's in the JVC stereo unit. I believe that for a digital recording, the only point of significant signal loss is the MP3 encoding and decoding. Thanks to MAD, the vast majority of that is now the encoder. Which leads me to the reason for posting: this implies that much of the perceived fidelity loss of MP3's involves flaws in soundcard DA's getting tickled by something in the MP3 algorithms. I'm available for whatever testing anyone can come up with. Keep up the good work! - Kevin Cody Jr

2 1

MAD plug-in for Winamp 0.13.1b-pre1
by Rob Leslie 24 Sep '01

24 Sep '01

Here is a new pre-release version of the MAD plug-in for Winamp: http://www.mars.org/home/rob/proj/mpeg/mad-plugin/in_mad-0.13.1b-pre1.zip This pre-release has two significant changes, one of which affects sound quality. Note well: THIS VERSION OF THE PLUG-IN PRODUCES DIFFERENT SOUNDING OUTPUT THAN PREVIOUS VERSIONS. This pre-release uses a new, experimental, dithering algorithm to address the dithering deficiencies of previous versions. It may (hopefully) sound better, or it may not. Please listen to it carefully and post your feedback, positive or negative. The other change is a new "Streaming" checkbox in the configuration, to enable or disable streaming support. If you prefer to use another plug-in with better streaming support (http/ftp), you can uncheck this while still leaving the MAD plug-in enabled for normal files. Cheers, -- Rob Leslie rob(a)mars.org

2 4

24 bit output in Win2K/XP....Success at last.....:))
by Hitesh Patel 22 Sep '01

22 Sep '01

Hello all, After numerous attempts at trying to get the Mad plugin to work at 24 bit resolution in Win2K with my Audiophile 2496, I am happy to report that it works perfectly due to the "Wrapper" that Ian Scott was kind enough to point me to. The only issues are: 1. The "wrapped" device name does not show up in Winamp. Actually it does, as a blank entry. Simply select the blank entry. The horrible static/buzz sound that all but drowned out the music was caused by the DSP plugin I was using. Once I disabled this, it worked fine. A real Duh,,,,.... Thanks again to Ian Scott, and to Rob Leslie. Hitesh Patel

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Bug in v0.13.1a?
by Ross Tregaskis 15 Sep '01

15 Sep '01

Check this one out. I'm using Winamp v2.76: 1) Install 0.13.1a, and switch from Nullsoft to this (by removing the MP3;MP2;MP1 list in it). 2) Do an ADD FILE. Note that MP3s don't come up under "supported types", and you have to scroll down the "Files of type:" box to locate them. 3) Disable 0.13.1a and switch back to Nullsoft. Do an ADD FILE, and the MP3s appear. 4) Remove MP3 from the Nullsoft decoder only, and enable MAD. Guess what: not under supported types. It appears that SUPPORTED FILES works with the Nullsoft decoder on this account; and that MAD doesn't work with it properly. It's worked OK with previous MAD versions, however. Hope a workaround can be found (or whatever I'm doing wrong, of course), -- Ross Tregaskis / <r.treg(a)spamhunter.co.uk>

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Dropouts when trying to stream audio still present
by George Pantazopoulos 15 Sep '01

15 Sep '01

Hey, Rob Glad to see a new version of MAD is here! However, I'm still having the same problem when trying to stream 128k mp3's off of mp3.com. a 1/2 to 1 second dropout occurs for every 15-20 seconds of playing time. I'm on a cable modem and this does not happen when I use the default decoder. Any ideas? Thanks, George Pantazopoulos

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Streaming MAD
by Magnus Holmgren 08 Sep '01

08 Sep '01

Hi, Bug found in MAD regarding streaming: it seems to hijack all streaming, making it impossible to recieve e.g. Ogg Vorbis streams when the MAD plugin is installed. Also, MAD seems to always use the proxy setting in "Internet Options" in Windows/Internet Explorer, rather than the setting in WinAmp. -- Magnus Holmgren

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