mad-dev July 2003

mad-dev@lists.mars.org

16 participants
17 discussions

Re: How to know the frequency of a frame
by armin.gerritsen＠philips.com 01 Aug '03

01 Aug '03

mad-dev-request(a)lists.mars.org wrote: > Message: 1 > Date: Wed, 30 Jul 2003 16:56:15 -0700 (PDT) > From: Shiming Dong <dong_sm(a)yahoo.com> > To: mad-dev(a)lists.mars.org > Subject: [mad-dev] How to know the frequency of a frame > > --0-1603577723-1059609375=:7610 > Content-Type: text/plain; charset=us-ascii > > Hi there: > > I have one question: I was supposed to implement a program to detect > a specific marker tone. But my problem is how to know the frequency > of each frame? > > Thank you! > > shiming I guess you mean the samplerate? For instance 44,1kHz. In that case it is easy. Once you have decoded a frame, at some point you will synthesise it to PCM samples. struct mad_frame myFrame ; struct mad_synth mySynth ; [...lots of code...] mad_synth_frame( &mySynth, &myFrame ) ; After that mySynth.pcm.samplerate will contain the samplerate. For instance 44100 for 44,1kHz. Regards, Armin

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How to know the frequency of a frame
by Shiming Dong 31 Jul '03

31 Jul '03

Hi there: I have one question: I was supposed to implement a program to detect a specific marker tone. But my problem is how to know the frequency of each frame? Thank you! shiming --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software

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Configure error
by hank.jan＠billionton.com.tw 31 Jul '03

31 Jul '03

Hi, I encounted some trouble when I did the cross compile. I am not familiar with linux environment. So I may ask a stupid question. Any help will be appreciated. I installed the arm cross compiler toolchian, and tried to compile madplay. But when I ran ./configure, I met an error with the message : checking whether byte ordering is bigendian... configure: error: can not run test program while cross compiling. I have ever succeeded the configuration, but I am not sure what to do. Can anybody help me? Thanks in advance.

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Minimad: output PCM data overflow ?
by Yeh You-Ying 31 Jul '03

31 Jul '03

mad-dev-request(a)lists.mars.org mentioned: > Could be, but in my experience even something as simple as: > >signed int MAD_SimpleScale( mad_fixed_t sample ) >{ > // Round > sample += ( 1L << (MAD_F_FRACBITS - 16) ) ; > > // Clip > if ( sample >= MAD_F_ONE ) > sample = MAD_F_ONE - 1 ; > else > if ( sample < -MAD_F_ONE ) > sample = -MAD_F_ONE ; > > // Quantise > return sample >> (MAD_F_FRACBITS + 1 - 16) ; >} > > >... already produces nice results. >If you forget about the clipping though, the quality s#cks! >I'm not sure, but I believe the standard mad-example 'forgets' the clipping. >(Could be wrong though!) >I don't know what minimad.c uses ... > >If that is not it, check the compiler settings. Some very agressive >optimisations may cause rounding-errors, although this usually occures with >floating point numbers, and MAD's strength is that it doesn't use them ... > >I know, I'm just quessing, but maybe this helps somehow. > >Regards, > >Armin > > > > Dear Armin, Thanks for your information. I've check the scale function in minimad.c and it's the same as the one you mentioned. And I 've tried several ways to change the compiler flags, but still no work. I used dist10 decoder to generate another PCM raw data with the same demo.mp3 I also used Sound Forge to analyze these two PCM raw data. The one generated by minimad seems to be overflow in sample value, the maximum and minimun value are -32768 and 32768 (2^16). Any hint about this ? I can email to statistic information to you if you'd like. Thnaks in advance. Yeh You-Ying

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Re: [mad-dev] Palm OS port
by Boris Raicheff 29 Jul '03

29 Jul '03

>From: Jesse McGrew <jmcgrew(a)hansprestige.com> >To: mad-dev(a)lists.mars.org >Subject: [mad-dev] Palm OS port >Date: Mon, 28 Jul 2003 21:19:48 -0700 > >I want to write a streaming audio player for Palm OS, so I'm thinking of >trying to port libmad to Palm > OS. Newer Palms have ARM processors but >user apps are mostly emulated 68k code, so theoretically > it'll be >possible as long as I can convert the whole library to an "ARMlet". > >I don't want to reinvent the wheel or step on anyone's toes, though. Is >there already a Palm port in > progress? Yes, there is. In fact, there are at least 4 people (the ones that I know of) working on different Palm OS ports. Unfortunately, your idea to convert the whole library to an ARMlet is not easy at all. So you won't be able to just include the sources in your Codewarrior or whatever you're using, and hit compile. Don't forget the 64k limit. There are many issues to deal with (like the lookup tables, which are too large, and the struct alignment thing, the name a few). Regards, Boris Raicheff _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus

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Re: PCM output sample overflow ?
by Yeh You-Ying 29 Jul '03

29 Jul '03

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Palm OS port
by Jesse McGrew 29 Jul '03

29 Jul '03

I want to write a streaming audio player for Palm OS, so I'm thinking of trying to port libmad to Palm OS. Newer Palms have ARM processors but user apps are mostly emulated 68k code, so theoretically it'll be possible as long as I can convert the whole library to an "ARMlet". I don't want to reinvent the wheel or step on anyone's toes, though. Is there already a Palm port in progress?

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Re: mad-dev digest, Vol 1 #278 - 1 msg
by armin.gerritsen＠philips.com 28 Jul '03

28 Jul '03

> Thanks for your reply. > I knew that the output is raw PCM data. > And the Sound Forge can play raw PCM data no matter what extsion > filename is. > I can hear the PCM data via Sound Forge, just the quality is not good, > too much nosie. > As the decription of scale( ) in minimad.c, there is no noise shaping > for scaling. > It that the reason about the quality issue ? Could be, but in my experience even something as simple as: signed int MAD_SimpleScale( mad_fixed_t sample ) { // Round sample += ( 1L << (MAD_F_FRACBITS - 16) ) ; // Clip if ( sample >= MAD_F_ONE ) sample = MAD_F_ONE - 1 ; else if ( sample < -MAD_F_ONE ) sample = -MAD_F_ONE ; // Quantise return sample >> (MAD_F_FRACBITS + 1 - 16) ; } ... already produces nice results. If you forget about the clipping though, the quality s#cks! I'm not sure, but I believe the standard mad-example 'forgets' the clipping. (Could be wrong though!) I don't know what minimad.c uses ... If that is not it, check the compiler settings. Some very agressive optimisations may cause rounding-errors, although this usually occures with floating point numbers, and MAD's strength is that it doesn't use them ... I know, I'm just quessing, but maybe this helps somehow. Regards, Armin

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Re: mad-dev digest, Vol 1 #277 - 1 msg
by Yeh You-Ying 26 Jul '03

26 Jul '03

>Today's Topics: > > 1. Re: PCM Outout Quality Issue (Rob Leslie) > > > > > Subject: > Re: [mad-dev] PCM Outout Quality Issue > (From): > Rob Leslie <rob(a)mars.org> > Date: > Fri, 25 Jul 2003 13:36:13 -0700 > (To): > mad-dev(a)lists.mars.org > > > On Wednesday, July 23, 2003, at 08:27 PM, Yeh You-Ying wrote: > >> What I did is just to decode a mp3 file with my minimad, and generate >> an output, for instance, pcm.wav. >> I played this raw pcm data with Sound Forge. >> Ohhh...it sounds terrible. > > > It's important to realize minimad doesn't output a WAV file. It just > writes raw PCM data in little-endian format. Make sure Sound Forge > isn't treating the file as a WAV file or reading the PCM as big-endian > samples. > Dear Rob, Thanks for your reply. I knew that the output is raw PCM data. And the Sound Forge can play raw PCM data no matter what extsion filename is. I can hear the PCM data via Sound Forge, just the quality is not good, too much nosie. As the decription of scale( ) in minimad.c, there is no noise shaping for scaling. It that the reason about the quality issue ? Thanks a lot.

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PCM Outout Quality Issue
by Yeh You-Ying 25 Jul '03

25 Jul '03

Hello Everyone, I'm new for MEPG decode. I just finished struggling from set-up MinGW/GCC environmnet. I've successfully compiled and linked minimad. There must be something wrong with me. I got terrible PCM output. Any hint to improve it ? What I did is just to decode a mp3 file with my minimad, and generate an output, for instance, pcm.wav. I played this raw pcm data with Sound Forge. Ohhh...it sounds terrible. My platform is x86 (P4), so here is the compiling CFLAG in my makefile, HAVE_CONFIG_H FPM_INTEL. and in comfig.h #define inline __inline #define STDC_HEADERS 1 #define HAVE_ERRNO_H 1 #define HAVE_FCNTL_H 1 #define HAVE_LIMITS_H 1 #define HAVE_SYS_TYPES_H 1 #define SIZEOF_INT 4 #define SIZEOF_LONG 4 #define SIZEOF_LONG_LONG 8 Did I miss something important ? Thanks in advance.

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Quantizing mad_fixed_t?
by Arve Knudsen 20 Jul '03

20 Jul '03

I see that the algorithm for 32 bit output in madplay quantizes to 24 bit; I'm a little daft when it comes to fixed point representations, but why are some of the fractional bits discarded even when you're outputting in full 32 bit? There's also one more thing I don't understand; the original sign bit of the fixed point number isn't kept (as far as I can see), instead the least significant whole part bit occupies the sign position. Does this whole part bit represent the sign (a little confused here)? Regards Arve Knudsen

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MAD Documentation
by Zeon 16 Jul '03

16 Jul '03

Hello, I'm looking for MAD documentation. Thanks

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samplerate/channels computation on layer 2
by Patrick Cernko 10 Jul '03

10 Jul '03

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm one of the developers of NMM (thanks for your link to use on your Website). We are using libmad to decode MPEG Audio Layer 2 as received by Digital Video Broadcast (DVB) via satelite. After remuxing the received MPEG transport stream to a MPEG 2 PES stream (first part: DVBReadNode) it is demultiplexed into its audio and video parts by our MPEG Demultiplexer Node (= demuxer unit). The resulting MPEG Audio part is MPEG Audio layer 2 with 48000 Hz sample rate (at least we did not found any other one yet) and one ore 2 channels. It is decoded by the MPEGAudioDecodeNode based on libmad version 0.14.2b. The code was inspired by the minimad example from your distribution although we had do make many changes, e.g. unlooping the while loop to process the incoming buffers or computing timestamps for the decoded audio buffers for later syncronisation with the video frames. Now back to my problem. I discovered, that some few times, the mad_pcm struct seems to contain invalid values for samplerate and number of channels. Although the mad_header struct in the mad_frame says, that the frame has 48000 Hz and 1 channel, in the mad_pcm struct, the values are 24000 Hz and 2 channels (see: the total number of samples is 96000 in both cases). I already tried the newer version 0.15 of the mad library, but I get the same effects. We also have mad a "simulation", where we use a recorded DVB-Stream. The effect apears here also indeterministically. Also playing the demulitplexed audio stream is affected. The "accident" occures not so often and most times, it is not reproduceable directly after it occured, except if the application is left running. When the app is left running we can reproduce it a few times and after 4-5 tries, the decoder gets it right. We also tried not to use the values from the mad_pcm struct but the ones from the mad_header struct, but as we heard, the decoded audio data was also affected (or wrong interleaved). Can you give me some hints where I can search for the error or do you think, this is a bug in the library perhaps? I would like to test it also with the current development library. Do you have something like anonymous CVS access? CU/all P.S.: If you want to see our implementation, you can download the latest NMM release. The Code in this release is (at least for the MPEGAudioDecodeNode) nearly identical with our current CVS head! - -- Patrick Cernko | mailto:errror@errror.de | http://www.errror.de Quote of the Week: "Is net scheen, das Bild, awwer einmalisch!" (Markus Sand) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.0 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQE/DatgB6g1EUxThvcRAg0OAKCsYBqKTWuN2suqQlIlVxszLavSPQCfYxrT TMXLriDskDnecWV4KNCcH8M= =LvIq -----END PGP SIGNATURE-----

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mad_frame subbands and filtering
by Bill Zissimopoulos 08 Jul '03

08 Jul '03

I need to process frame subbands before synthesis occurs. Libmad conveniently makes them available through its filtering feature. However the information around filtering is rather limited. I have studied the source code in filter.c and (think that I) have read all the relevant articles in this list, so I understand the basics, however I do have a few more questions: 1. What are the units and range of values for each subband? Must these values be scaled before they can be used? [I saw in a list article that subband values range from -MAD_F_ONE to MAD_F_ONE. Have these values been scaled?] 2. What are the frequency ranges that each subband covers? Many thanks. Bill

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question about libmad install
by Shiming Dong 06 Jul '03

06 Jul '03

Hi there: I downloaded the source code for libmad and try to install it on Windows. But I don't know how to install it, do I need to write a make file ? I have no idea at all. From the document, it said I can use either MSVC++ or cygwin. But there is no readme file for how to install using MSVC++. I installed cygwin, but I don't know how to use it to install? Sorry for my questions, but I'm a beginer and I really have no idea. If anyone could help me, I'd appreciate it very much. Thank you! shiming --------------------------------- Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month!

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mad ogg support
by Joe Drew 03 Jul '03

03 Jul '03

Hi Rob, I remember you mentioning that you might implement a sort of mad-ogg which was a integer-only ogg vorbis decoder, but you were stymied by the lack of a proper specification for the ogg vorbis format. Now that Tremor has been released, do you still plan on making such a project? Thanks, Joe

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Crosscompiling for ARM - Zaurus
by Dave 03 Jul '03

03 Jul '03

Hi I've Successfully compiled madplay and it's libs for the Zaurus. Everything works. I used the following options for configure: CC=/usr/local/Embedix/tools/bin/arm-linux-gcc \ RANLIB=/usr/local/Embedix/tools/bin/arm-linux-ranlib \ ./configure --host=arm-linux \ --enable-speed \ --enable-fpm=arm \ --libdir=/opt/Qtopia/sharp/lib \ --includedir=/opt/Qtopia/sharp/include The playback sounds great. The only problem I've got is occasionally "skipping" which I'm assuming are some sort of buffering problem. The skips only occur when I run other apps while playing an MP3. They occur more often if the MP3 file is stored on an SD card or fetched from an SMB mounted file system. I haven't yet taken a good look at how madplay reads and buffers the data. I originally built it with mmap. After thinking about it - if my understanding of mmap is correct - it makes little sense on a device like the Zaurus. I rebuilt madplay without mmap (is the correct way to do this to edit config.h ?) and the skipping remains. So it seems that mmap had little effect. If mmap works on the Zaurus - when it's enabled madplay will request to have the full MP3 file "mapped" into memory to avoid disk reads while playing it (is this correct?). On the Zaurus, the MP3 will already be stored in memory, and mmap would just copy it from one chunk of memory to another. Perhaps this would save the time it take to execute the filesystem code but it would chew up memory on a limited device. Buffering - I don't know how the buffering in madplay works. I assume that buffering would be better for the Zaurus. Is madplay meant to be a robust generic MP3 player or is just to demonstrate how to use libmad? Finally, are there any other config options I should try? I'm not familiar with the ARM family of processors - I've noticed other posts that specify a specific arm chip in the config options. How about other options for libmad/madplay? Is there a way to get configure to tell you the full list of options? I tried --help but people here are mentioning options that aren't shown. I'm impressed with the work thats been done on MAD - it compiled and worked right out of the box and sounds great. The only trouble I had was figuring out how to cross compile it, and those questions were answered right here in other posts. Thanks in advance for any help offered. Dave ---------------------------------------------------------------- Dave Belser (mad(a)newww.com)

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mad-dev July 2003