mad-dev May 2001

mad-dev@lists.mars.org

15 participants
14 discussions

Re: [mad-dev] porting mad
by Andre 03 Jun '01

03 Jun '01

Antti, (I have copied the mad-dev list as I think Rob might be able to help out with the inline assembler syntax if it really is needed... :-) For fast decoding, the situation you are trying to achieve is that after every 32x32 hardware multiply instruction, only the upper 32bits of the answer are referenced in order to generate the mad_fixed_t result. (You will have to check the compiler output to be sure...) Please give this patch a try. It _may_ achieve the desired effect without needing to resort to assembler.... ? --- fixed.h.original Wed May 30 04:11:29 2001 +++ fixed.h Wed May 30 04:42:12 2001 @@ -398,10 +398,16 @@ (((hi) << (32 - (MAD_F_SCALEBITS - 1))) | \ ((lo) >> (MAD_F_SCALEBITS - 1)))) + 1) >> 1) # else -# define mad_f_scale64(hi, lo) \ - ((mad_fixed_t) \ - (((hi) << (32 - MAD_F_SCALEBITS)) | \ - ((lo) >> MAD_F_SCALEBITS))) +# if defined(OPT_SPEED) +# define mad_f_scale64(hi, lo) \ + ((mad_fixed_t) \ + (((hi) << (32 - MAD_F_SCALEBITS)))) +# else +# define mad_f_scale64(hi, lo) \ + ((mad_fixed_t) \ + (((hi) << (32 - MAD_F_SCALEBITS)) | \ + ((lo) >> MAD_F_SCALEBITS))) +# endif # endif # define MAD_F_SCALEBITS MAD_F_FRACBITS # endif Andre -- --- Antti Antinoja <antti(a)neonzion.fi> wrote: > > I'll try to add the asm multiply function yet then we see.. the gap > reduced greatly after i turned debugging off (..really).. It took > ~ 72 s for a 60 s mp3.... > > I'm not sure if there would be any gain in case I implement an asm > multiply function... I tryed that, but there are some things I > can't figure out with the cris asm while in extended inline.... > cris offers muls.m and mulu.m (m for format, b=byte, w=word, > d=double word) functions.. I thing i should use the signed one.. > ok.. but there seems to be something with the registers... > the lo variable is read correctly from the registers, but hi will > contain only some grap.. ( I made a little test.c program to > evaluate this asm thingy easily...). > > this is a non working attempt: > > # define MAD_F_MLX(hi, lo, x, y) \ > __asm__ ("muls.d %2, %3" \ > : "=r" (lo), "=r" (hi) \ > : "%r" (x), "r" (y) \ > : "0") > > The manual describes following : > > Assembler syntax: MULS.m Rs,Rd > Size: The operands are byte, word, or dword. The result is 64 bits. > Operation: MOF = ((m)Rs * (m)Rd) >> 32; > Rd = (dword)((m)Rs * (m)Rd); > Description: Both operands are sign extended from the size (m) to > dword, and the extended operands are multiplied, generating a > 64-bit result. > The lower 32 bits of the result are written to Rd, and the upper 32 > bits are written to the multiply overflow register (MOF). > N and Z flags are set depending on the 64-bit result. > The V flag is set if the result is more than 32 bits: > V-flag = ((Rd >= 0) && (MOF != 0)) || > ((Rd < 0) && (MOF != -1)) > ____________________________________________________________ Do You Yahoo!? Get your free @yahoo.co.uk address at http://mail.yahoo.co.uk or your free @yahoo.ie address at http://mail.yahoo.ie

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WIn32 library
by steve.lhomme＠free.fr 01 Jun '01

01 Jun '01

Hi everyone, I'd like to use MAD as a library (a DLL would be better) to read from MP3 buffers. The idea is to create a Windows codec (coding-decoding) with a complete free software solution. I'm currently working on having lame (as a DLL, since my project has the BSD license and not GPL) to do the encoding part, and I'd like to use MAD to do the decoding. Is that possible ? Any way of building a DLL without using cygwin (because I'm don't want to be obliged to use Cygwin on the user PC) ?

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Re: [mad-dev] WIn32 library
by Chad Austin 31 May '01

31 May '01

First, what's standard etiquette for using a mailing list? Do I respond to both the person and the list, or just the list, or what? > It actually works? what modification did you make to the code, > especially huffman.c ? Yup, works fine. The main patches to huffman.c were changing PTR for huffquad and huffpair: // huffquad PTR macro # define PTR(offs, bits) { { \ 0, \ bits, \ (offs & 1) >> 0, \ (offs & 2) >> 1, \ (offs & 4) >> 2, \ (offs & 8) >> 3, \ (offs & 0xFF0) >> 4 \ } } // huffpair PTR macro # define PTR(offs, bits) { { \ 0, \ bits, \ (offs & 0x000F) >> 0, \ (offs & 0x00F0) >> 4, \ (offs & 0x0F00) >> 8 \ } } This'll probably only work on little-endian machines. Of course, there were several other minor changes. One of these days I'll build a patch and submit it to Rob. For now, use anonymous CVS: CVSROOT=:pserver:anonymous@pyrallis.aegisknight.org:d:/cvsroot cvs login Password: guest cvs co acoustique One last note... I *attempted* to port FPM_INTEL to VC++ (renaming it to FPM_INTELVC)... However, my ports of the asm() directives never worked. :) Also, FPM_64BIT was about four or five times slower than FPM_DEFAULT. So if you're compiling with VC++, you *definitely* want FPM_DEFAULT. -- Chad Austin http://aegisknight.org/

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VC++ 6.0 and libmad/madplay
by Vladimir Vunukov 31 May '01

31 May '01

Hi all Did anyone port to VC++? I tried to do it but have some troubles with libmad's hangs. -- Best regards, Vladimir mailto:vlad_vunukov@yahoo.com _________________________________________________________ Do You Yahoo!? Get your free @yahoo.com address at http://mail.yahoo.com

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Re: Will the last frame be decoded ?
by Rob Leslie 29 May '01

29 May '01

Eric Peng wrote: > I'm using libmad for MP3 streaming decoding. However, my program > doesn't decode the last frame. You've correctly isolated the problem. The solution is to append enough zero bytes to your stream buffer after the last frame to satisfy the MAD_BUFFER_GUARD. The reason for MAD_BUFFER_GUARD has to do with the way decoding is performed. In Layer III, Huffman decoding may inadvertently read a few bytes beyond the end of the buffer in the case of certain invalid input. This is not detected until after the fact. To prevent this from causing problems, and also to ensure the next frame's main_data_begin pointer is always accessible, MAD requires MAD_BUFFER_GUARD (currently 8) bytes to be present in the buffer past the end of the current frame in order to decode the frame. Cheers, -- Rob Leslie rob(a)mars.org

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porting mad
by Antti Antinoja 29 May '01

29 May '01

Hi! I've yesterday downlaoded the mad source (latest beta).. mad-0.13.0b. ok.. my quest was (or is...) to compile mad to cris (made by Axis, 32 bit RISC, also know as etrax_lx) platform. Some tweaking with env. vars (CC=cris_gcc, etc)... made a little ./configure "wrapper": #! /bin/sh rm config.cache CC='gcc_cris -mlinux -DCRISMMU -muclibc=/home/antti/axis/devboard_lx/eroot' \ OBJCOPY=objcopy-cris \ LD=ld-cris \ AR=ar-cris \ RANLIB=ranlib-cris \ ./configure --host=unknown --target=cris --disable-nls --disable-aso --enabl e-fpm=approx I had to do something bad also with the main configure script to avoid a problem with the big-endian test (system couldn't run the cris code... compiling under intel x86 ... with cris_gcc cross compiler..) Ok.. with this setup (and some additional work before, cause cris uC-lib didn't have any native math suopport, which i had to add ... well... ) I managed to compile the whole thing without any ugly errors... After this I naturally tested the thing with the native cris platform.. Madplay could read the mp3:s very well, but the output (to a wav file) wasn't too good.. I coud hear that the music was there, but there was also some additional noice a lot. any pointers where i shuld start digging... (?) /Antti

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mad output
by 郑灵翔 28 May '01

28 May '01

Hi, I want to compile madplay for ARM720T CPU in Linux. But I find the output sound is terrible. I think the 8-bit PCM output may get better result in this CPU. Can anyone tell me how to get the 8-bit PCM output using madplay? Thanks.

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IMDCT algorithm
by yu peng 28 May '01

28 May '01

Hi,Rob Thank you for your excellent code! Your MAD baby is a great help to me! Would you please tell me, what is your IMDCT algorithm? Or, where can get documentations for this algorithm? Thank you very much! Danny Yu

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mad plugin for xmms
by Sam Clegg 26 May '01

26 May '01

So, I finally got around to writing a plug-in for xmms that uses libmad. http://superduper.net/cvsweb/cvsweb.cgi/xmms-mad/ http://superduper.net/xmms-mad/ Right now it can play files or shout/ice streams. Seeking/configuring/fileinfo/length estimation are all missing. If anyone wants to out I would welcome contributions. I think I will steal the file info and configuration dialogs from the mpg123 plugin. sam -- sam clegg email: samc(a)superduper.net www : http://www.superduper.net pgp key : http://www.superduper.net/~samc/key.gpg

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dct32 workings
by Rein-Peter de Boer 21 May '01

21 May '01

Hi, I'm trying to grasp the inner workings of the dct32 function. I can understand the matrixing stuff of the butterflies, but i don't understand the shifts (<<1) before or after a calculation. Any tips? grts & thx, RP

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madplay in BeOS
by whee 21 May '01

21 May '01

I decided on trying BeOS again and was wondering if anyone had written an output module using the media server (or whatever). Everything compiles fine, but I don't see a way to actually generate sound without having madplay decode to a .wav and then playing that. It should be pretty simple to do and I was going to try it myself, but ended up confusing myself with the internals of mad while looking at the existing audio_* files. I hope someone's bored enough and has BeOS. :D -whee

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Re: Low level programming interface question
by Rob Leslie 09 May '01

09 May '01

Hi Lionel, > If I put aside the licensing issues ( :-) ), my biggest problem is to really > understand how the low-level MAD interface is working. After looking at the > WinAMP plug-in source, I somewhat understood how it works (except for the > timers, where I am completely lost). My understanding for now is : > > - read some data from the stream > - call mad_stream_buffer These steps: > - call mad_frame_decode > - call mad_synth_frame > - send the data to the sound device should be repeated until MAD returns the error MAD_ERROR_BUFLEN. > - copy the data remaining in the buffer (ie the 'next_frame' pointer) to > the start of my input buffer > > For a stand-alone player, this works wonderfully. The problem comes when I > try to incorporate this in my AVI player. Indeed, when I added some printfs > to my small player, I saw that MAD always : > > - output the same amount of data (0.006 seconds of sound) > - always 'progressed' of the same amount of data in my stream Each MPEG audio frame generally contains the same number of PCM samples, and (in the case of constant bitrate streams) will usually consume the same amount of data from the stream. Each call to mad_frame_decode() and mad_synth_frame() will decode only a single frame. > On the other hand, in my AVI stream, for each video frame, the amount of MP3 > data is not AT ALL constant.... Moreover, it is sometimes smaller than the > amount of data read by MAD => I get (as soon as the first video frame) a > decode error from MAD (I did not check exactly what error it was). You will have to demultiplex the MPEG audio data from the video data and stream this independently to MAD. > I could do 'buffering' (ie concatenate some sound AVI chunks) but then I > would completely lose my synchro (and I do not want to rely on timers or to > retrieve the synchro, it would start to get bad with long films). You will probably have to deal with the synchronization somehow. MAD's timers can help you keep track of how much audio data has been decoded. They might seem intimidating but they're really quite simple to use. I'll post some details in a follow-up message. > PS: I hope this is the right mailing list, I did not know if I should send > this to the devel list or the user list :-) I think mad-dev is probably more appropriate; I've adjusted the Reply-To header but you might need to subscribe before you can reply. Cheers, -rob

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Resampling 44.1k to 48k
by Ira Weiny 09 May '01

09 May '01

I have an application which needs to decode mp3's to 48kHz. As most mp3's are 44.1 I am going to have to resample. I am using libmad to decode the mp3 just fine by starting with the minimad source. I then integrated the resampling algorithim in resample.c. I noticed from the comments that this was not the best resampling algorithim available. Has anyone else done something better with fixed point math? Ira

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MAD timer API
by Rob Leslie 08 May '01

08 May '01

The basic timer datatype is mad_timer_t; it represents a unit of time. You can initialize a timer by setting it either to a zero constant or to some other explicit value: mad_timer_t timer; timer = mad_timer_zero; /* or */ mad_timer_set(&timer, seconds, frac_numer, frac_denom); Timers have a precision of 1/352800000 second, so you can use the frac_numer and frac_denom arguments to specify a fractional value as a rational number. The resulting timer value, in seconds, is: seconds + (frac_numer / frac_denom) You can add one timer value to another with: mad_timer_add(&timer, timer2); You can also multiply a timer value by an integer: mad_timer_multiply(&timer, scalar); Timer values can be either positive or negative; to change the sign of a timer: mad_timer_negate(&timer); To get the absolute value: timer2 = mad_timer_abs(timer); Or, to determine the sign: sign = mad_timer_sign(timer); /* < 0 negative; = 0 zero; > 0 positive */ In general you can compare any two timer values: comp = mad_timer_compare(timer1, timer2); /* comp < 0 : timer1 < timer2 */ /* comp = 0 : timer1 = timer2 */ /* comp > 0 : timer1 > timer2 */ Finally, there are methods for extracting the value of a timer. To convert a timer value to selected UNITS (see libmad/timer.h for unit constants): count = mad_timer_count(timer, UNITS); /* rounded down */ If you assume a basic unit of seconds, you can get the remaining fractional part of a timer in arbitrary terms with: frac_numer = mad_timer_fraction(timer, frac_denom); The precision of the timers is such that frac_denom can be any valid MPEG audio sampling frequency to get sample-granular results. Finally, you can get a string representation by supplying a sprintf-style format and selecting various units: mad_timer_string(timer, dest, format, UNITS, frac_UNITS, subparts); For example, to get HH:MM:SS.DDD you could use: mad_timer_t timer; char dest[13]; mad_timer_string(timer, dest, "%02lu:%02u:%02.%03u", MAD_UNITS_HOURS, MAD_UNITS_MILLISECONDS, 0); (MAD_UNITS_HOURS implies the next two format specifiers are for minutes and seconds.) There are also selectors for video frame units, and even drop-frame time coding is supported: mad_timer_string(timer, dest, "%02lu:%02u:%02u;%02u", MAD_UNITS_HOURS, MAD_UNITS_29_97_FPS, 0); After each call to mad_frame_decode() (or mad_header_decode()), frame.header.duration is filled with a timer value representing the playing time of the audio in the frame. You can accumulate these times with mad_timer_add() to keep an accurate time position as you play the audio. -- Rob Leslie rob(a)mars.org

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mad-dev May 2001