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682 discussions

Problems compiling minimad.c
by Håvard Sperle Engebretsen 29 Sep '05

29 Sep '05

Hi! I am running Cygwin and have installed the libid3tag and libmad libraries. After installation I got the following message in the command window: Libraries have been installed in: /usr/local/lib If you ever happen to want to link against installed libraries in a given directory, LIBDIR, you must either use libtool, and specify the full pathname of the library, or use the `-LLIBDIR´ flag during linking and do at least one of the following: -add LIBDIR to the `PATH´ environment variable during execution -add LIBDIR to the `LD_RUN_PATH´ environment variable during linking -use the `-Wl,--rpath -Wl,LIBDIR´ linker flag When I try to compile the minimad.c file in libmad with: gcc -static -pg -minimad.c -o minimad -lmad (I'm using -static to be able to do a profiling later on ) I get an error message that says that: .../i686-pc-cygwin/bin/ld: cannot find -lmad When I am triyng to use just: gcc -static -pg minimad.c -o minimad I get several errors telling me that there are "undefined references to `name_of_function´". I'm guessing both of these problems have something to do with me not linking against installed libraries? As you probably already have figured out, I'm kind of a newbie to all of this and I would really appreciate your help. Tanks. H. --

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Problems with libmad and minimad.c
by Haavard Engebretsen 29 Sep '05

29 Sep '05

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Problem while decoding higher Kbps files
by Swapnil Wagle 27 Sep '05

27 Sep '05

Hello, There is an unusal problem that I am facing now.I have ported 'Madlib' on to PalmOS.I am able to connect to a shoutcast server and decode the mp3 data using madlib and that too for any Kbps. This works fine on simualtor. But when I ported this onto the device it works well for 32Kbps data. But when the Kbps rate is higher than this the applciation crahes. When I debugged I found out that there is a funcion in 'Layer3.c' called 'III_stereo' which gets called when the Kbps is higher than 32. if (header->mode == MAD_MODE_JOINT_STEREO && header->mode_extension) { WinDrawChars("Namaste",7,30,90); error = III_stereo(xr, granule, header, sfbwidth[0]); if (error) return error; } This causes the crash.If I comment this function that appliation doesnt crashes but the decoding doesnt takes place. This is the final phase of my project.Please help me. Regards, Swapnil

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Synchronization error near the end of file
by Marcello Maggioni 15 Sep '05

15 Sep '05

Hi all, I'm tring to get started with "libmad" so I've tried to make an application very similar to "minimad.c" , the one included in the source code, only that instead of reading and writing from STDIN to STDOUT it reads and writes from and to files. The problem that I get is that I get a strange synchronization error near the end of the file : Decoding /balthasar/****.mp3 0 <-- ID3Tad size decoding error 0x0101 (lost synchronization) at byte offset 3072836 The actual size of the file is : 3072949 I get this almost with every file I try to decode. I manage the input by creating with "malloc()" a memory area of "unsigned chars" of the size of the file , then I fread the mp3 file into memory and pass the pointer to the callback input function . I miss something or I made any mistakes? Thanks for your help. Bye Marcello PS= Do you know how to check the correctness of a Raw 16bit PCM stream?? I've tried to play my file with many players , but it doesn't work. Sorry but I'm a newbie of digital audio :)

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Synchronization error near the end of file
by Marcello Maggioni 14 Sep '05

14 Sep '05

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buffering madlib samples
by Swapnil Wagle 05 Sep '05

05 Sep '05

Hello, I am trying to play the converted PCM samples from Madlib on PalmOS..But the speed at which Madlib decodes the samples is too high.I am not able to buffer all the samples at one time whicih results in buffer overlapping.I wanted to know whether it is possible to make the Madlib libarary wait for a while till all the current samples have been played. Regards, Swapnil

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Quyery regarding using madlib for decoding streaming mp3 chunks
by Swapnil Wagle 02 Sep '05

02 Sep '05

Hello, Thanks to all you people finally half of the job has been done.I have been able to convert the mp3 file into a wave file on PALM device.And the conversion quality is superb. However my main intension for using Madlib is a different one.So I had some doubts regarding whether this can be done or not. I want to use madlib for decoding the mp3 data that comes from the shoutcast servers.So I wanted ot know whether this can be accomplished using madlib. And whether it will be able to decode chunks of data if given in any order? Regards, Swapnil

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Verificattion for reducing the size of rq_table.dat
by Swapnil Wagle 30 Aug '05

30 Aug '05

Hi, For reducing the size of rq_tbale.dat as suggested by you people I have done following changes within the libray.Can you please verify whether this is correct or not. The changed 'III_requantize' is as follows static mad_fixed_t III_requantize(unsigned int value, signed int exp) { mad_fixed_t requantized; signed int frac; struct fixedfloat const *power; frac = exp % 4; /* assumes sign(frac) == sign(exp) */ exp /= 4; value = value / 8; // Dividing the input value by 8 power = &rq_table[value]; power = power * 16; // Multiplying the result after finding the array index by 16 requantized = power->mantissa; exp += power->exponent; if (exp < 0) { if (-exp >= sizeof(mad_fixed_t) * CHAR_BIT) { /* underflow */ requantized = 0; } else { requantized += 1L << (-exp - 1); requantized >>= -exp; } } else { if (exp >= 5) { /* overflow */ # if defined(DEBUG) fprintf(stderr, "requantize overflow (%f * 2^%d)\n", mad_f_todouble(requantized), exp); # endif requantized = MAD_F_MAX; } else requantized <<= exp; } return frac ? mad_f_mul(requantized, root_table[3 + frac]) : requantized; } Reagrds, swapnil

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output WAV file contains woblling sound
by Swapnil Wagle 22 Aug '05

22 Aug '05

Hello there, I have ported Madlib on Palm OS platform.I am using the 'FPM_DEFAULT' setting for using the Madlib library because usng any other setting the Codewarrior compiler gives an error.The program runs fine but the output WAV file which I am writing conatins wobbling sound.The tune for the song is fine but the output wobbles a lot. I wanted to know whethe this is due to the FPM_DEFAULT setting we are using ,if not then what could be the reason for this kind of output. Regards, Swapnil

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Problem while decoding large size files
by Swapnil Wagle 20 Aug '05

20 Aug '05

Hello, Finally I have been able to run the Madlib on Codewarrior with this forrums help.Well there is still two major problem.I am trying to write a WAV file from the source mp3 file. 1.If the size of the source mp3 file becomes greater than 800KB then a crash occurs within the 'II_huffdecode' function.So what could be the problem over here.I have useing the hogh level API function from minimad for cnversion. 2.If the file is small then the decoding takes place but the data within the output file is all corrupt. Will there be a problem where I am witring the data to a WAV file becoz I alway get output samples? Regards, Swapnil

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problem with the 'rq_table.dat' during compilation
by Swapnil Wagle 20 Aug '05

20 Aug '05

Hello, I am trying to compile Madlib for PalmOS.Since Palm OS has the 68K limitation I removed last 20 entires from the rq_table.dat.Will this work w/o these entires,atleast a garbage output. Secondly the program is compiling now bu constantly crashes in the 'III_huffdecode' function.Can someone please help me I dont know whther I am doing the right thing or not. Regards, Swapnil

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Re: Re: [madplay]:audio_win32.c::struct buffer.event_handle: Can not find the time that sets the event object to the signaled state (Rob Leslie)
by cnmmd 20 Aug '05

20 Aug '05

1'st Init() { ... output[i].event_handle = CreateEvent(0, FALSE /* manual reset */, FALSE /* initial state */, 0); ----->the event object initial state is FALSE,--nonsignaled. .. } After this ,the event_handle is nonsignaled state. ~~~~~~~~~~~~ but in function: int wait(struct buffer *buffer) { MMRESULT error; DWORD tick; if (!buffer->playing) return 0; switch (WaitForSingleObject(buffer->event_handle, INFINITE)) { .. } why the event_handle is signaled state? I only find this time to setevent: void CALLBACK callback(HWAVEOUT handle, UINT message, DWORD data, DWORD param1, DWORD param2) { WAVEHDR *header; struct buffer *buffer; switch (message) { case WOM_DONE: header = (WAVEHDR *) param1; buffer = (struct buffer *) header->dwUser; if (SetEvent(buffer->event_handle) == 0) { ... } case WOM_OPEN: case WOM_CLOSE: -----> nothing to do in this. break; } ------------------------------ Message: 2 Date: Fri, 19 Aug 2005 11:39:24 -0700 From: Rob Leslie <rob(a)mars.org> Subject: Re: [mad-dev] [madplay]:audio_win32.c::struct buffer.event_handle: Can not find the time that sets the event object to the signaled state To: mad-dev(a)lists.mars.org Message-ID: <DE968453-D275-4D13-9D99-F285590788C2(a)mars.org> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed On Aug 18, 2005, at 10:26 PM, cnmmd wrote: > why the event_handle is signaled state? > > I only find this time to setevent: > void CALLBACK callback(HWAVEOUT handle, UINT message, DWORD data, > DWORD param1, DWORD param2) > { > WAVEHDR *header; > struct buffer *buffer; > > switch (message) { > case WOM_DONE: > header = (WAVEHDR *) param1; > buffer = (struct buffer *) header->dwUser; > if (SetEvent(buffer->event_handle) == 0) { > > ... > } That's it. This callback routine is passed to waveOutOpen(), which allows it to be called asynchronously after each data block sent via waveOutWrite() has finished playing. -------->yes, This callback routine will to be called after each data block sent via [waveOutWrite()] has finished playing. but after waveOutOpen(),the callback do nothing.the event object is nonsignaled also. -- Rob Leslie rob(a)mars.org End of mad-dev Digest, Vol 3, Issue 5 ************************************* -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

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[madplay]:audio_win32.c::struct buffer.event_handle: Can not find the time that sets the event object to the signaled state
by cnmmd 19 Aug '05

19 Aug '05

1'st Init() { ... output[i].event_handle = CreateEvent(0, FALSE /* manual reset */, FALSE /* initial state */, 0); .. } After this ,the event_handle is nonsignaled state. but in function: int wait(struct buffer *buffer) { MMRESULT error; DWORD tick; if (!buffer->playing) return 0; tick = GetTickCount(); switch (WaitForSingleObject(buffer->event_handle, INFINITE)) { .. } why the event_handle is signaled state? I only find this time to setevent: void CALLBACK callback(HWAVEOUT handle, UINT message, DWORD data, DWORD param1, DWORD param2) { WAVEHDR *header; struct buffer *buffer; switch (message) { case WOM_DONE: header = (WAVEHDR *) param1; buffer = (struct buffer *) header->dwUser; if (SetEvent(buffer->event_handle) == 0) { ... } -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

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Output samples are alway zero
by Swapnil Wagle 16 Aug '05

16 Aug '05

Hi, I am trying to port Madlib on PalmOS using Codewarrior.So far I have been able to compile and run Madlib using CW. The function 'mad_frame_decode' dosnt return any error meaning the data is being decoded properly( isnt it). But after that when the 'mad_synth_frame' is called it doesnt fills the pcm samples.Can I figure out a way using which I can find whether these function are working properly or not.I debugged the 'synth_full' and 'synth_half' function but didnt for ne thing out of it. Regards, Swapnil

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Preprocessor error while compiling Madlib
by Swapnil Wagle 12 Aug '05

12 Aug '05

Hello, I am trying to compile the Madlib library using Codewarriror 9.3 by creating PACE application. But during the compilation process I get errors related to the Macros defined in 'Fixed.h' file. Can anyone please help me out.The error which I am getting is as follows Error : Preprocessor #error directive fixed.h line 450 #error "no FPM selected" I also wanted to know whether this is the proper process of compiling Madlib using Codewarrior's PACE application. Regards, Swapnil

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Help:How to get the time stamp for player (not system time, it need return the position time stamp)
by cnmmd 11 Aug '05

11 Aug '05

Hi, I’m studying the minimad, in this function: static enum mad_flow output(void *data, struct mad_header const *header, struct mad_pcm *pcm) { …. } 1.How to get the total time of a mp3 file? 2.How to get the time stamp for player (not system time, it need return the position time stamp)? Can somebody help me? Thanks a lot. -- Using Opera's revolutionary e-mail client: http://www.opera.com/mail/

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RE: [mad-dev] obtaining stream length from mad decoder
by Nigel Sinfield 30 Jul '05

30 Jul '05

> 4. Scan every frame header of the file.... This does not require a full > decode of each frame, so it can be done relatively quickly. Do you have any sample code to do this? I need to do it for a completely different reason (multiplexing Layer II data into a MPEG-2 transport stream) The contents of this email and any attachments are sent for the personal attention of the addressee(s) only and may be confidential. If you are not the intended addressee, any use, disclosure or copying of this email and any attachments is unauthorised - please notify the sender by return and delete the message. Any representations or commitments expressed in this email are subject to contract. ntl Group Limited

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Trying to Trace Error... need some help
by Matt Weber 30 Jul '05

30 Jul '05

Hi all, Thanks for your previous help Rob and others. So here's my situation. I've managed to get a working example of minimad going under MSVC6 which decodes very nicely... I was just doing this so I could have something to compare my other project to. My other project is a minimad example which I am trying to port to a SHARC DSP. In any case, after making some minor modification to the code (replacing some functions in minimad that were unavailable in my DSP's library's) I found that the output pcm was not correct. (I'm inputting a simple kHz sine wave and the output looks like a gradually expanding amplitude sine wave with some randomness..) I have set all the necessary pre-processor defines for my architecture... ex FPM_DEFAULT for now until I write my own asm routine. What I would like to be able to do is see exactly how each individual input sample is being decoded and stored as pcm (pcm->samples[]). My problem is that I can't see where the code is outputting to the pcm->samples location. I'm a relatively inexperienced programmer, but would like to think I can manage to figure things out given time. What I imagine is happening is that these locations aren't being filled directly, but being passed an address somehow. Sorry for this lame question, but I have spent hours stepping through the code and still cannot see how the pcm->samples are being filled. I believe the problem that I am having is related to some of the math which is performed on each sample. Seeing how each sample is being treated and comparing it with a working example should help me figure out a fix for this. If anyone has any tips I would be extremely grateful. Matt

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obtaining stream length from mad decoder
by Langley, Adam 29 Jul '05

29 Jul '05

Hi Guys, I have written a Managed C++ wrapper for the libMad decoder, and I would like to know.... How can I get EITHER, the _decoded_ length in bytes, or the duration in seconds of the whole file, without decoding the whole file? I need to just evaluate a set of mp3 files, and obtain their lengths (preferably in bytes, then I can calculate the length in seconds by the bytes-per-second) Thanks! - Adam

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WAV output has "static"
by Matt Weber 26 Jul '05

26 Jul '05

Hi all, Im using minimad as an example and compiling in MSVC6 (under windows obviously... sorry I know.. windows ugh). Im managing to input and output data, but the output sounds full of static (under which I can still barely make out my original audio). Im just outputting raw pcm data and using cool edit to view and play the waveforms. My input mp3 is just a mono 1khz sine wave for one second, just so I can see what is happening on the output. What I see on the output is nearly the same signal, but with quite a few random spikes that exceed the original waveforms amplitude. Like I said, I can still hear the 1 khz tone but it is garbled full of static. Using winamps demo.mp3 (kicking the llama's ass soundbite) I can still also hear the guys voice, but again it is garbled with static. Has anyone else had this problem? What might be causing this? I have messed around with the preprocessor definitions and believe I have set them all correctly, so Im not sure where to go next. ANY suggestions would help me. Thanks a lot everyone! Matt

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Switching from xaudio
by toftat＠cowshed.us 26 Jul '05

26 Jul '05

Hi, My homebuilt car mp3 player has started acting up on certain mp3's (xaudio simply crashes with signal 32) so I would like to start using libmad. However with the lack of documentation this seems to be a little daunting! I will be using it in an asynchronous setting with a thread to handle the call backs (it will also set up the mmap() for the input and handle writing the output into the appropriate device), this much I have acertained as being possible from minimad. The only thing I think I am going to have problems with is that the decoder doesn't report any status as far as track time and elapsed time which are essential to this project. Am I missing something simple or is this an issue (if it is, how should I work around it?) Thanks Anthony

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Endian Problem
by Matt Weber 11 Jul '05

11 Jul '05

Hi Everyone, I know normally the configure script should take care of the endian-ness of your platform, but in my case I am trying to port Madlib to a DSP chip made by analog devices. In any case, I was unable to compile in unix because my IDE is only available for windoz.... leaving me to configure things on my own. After much effort I realized that my input buffer contains mixed up data.. due to the endian problem. If I view my mp3 in a hex viewer, the first address contains: FFFBB044 But when I view my buffer in memory after mapping I see 44B0FBFF Which shows that the byte order is reversed along with each individual byte (4 of them) being reversed as well. (This is shown by B0 and FB...) In any case, does anyone know how to switch this around? I would like to be able to do this without manually switching the byte order... which could be computationally expensive. thanks for your help everyone. Matt

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Where is the wimamp plugin?
by Vadim Berezniker 10 Jul '05

10 Jul '05

There's references to the MAD Winamp plugin source code but I cannot find it inside the official mad tarballs...

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How to compile TCPMP for palm OS.
by Swapnil Wagle 22 Jun '05

22 Jun '05

Hello, Actaully I have downloaded the soure code for TCPMP.The document says that we need to build the project using 'Microsoft Embedded VC++ 4.0' version.So I opened the project with this IDE.But is doesnt have the files that we need to compile it for PalmOS. So I needed to know how do we compile the source of TCPMP for Palm OS. Any help would be appreciated. Regards, Swapnil

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MAD plug-in for winamp
by Sherwin Huang 22 Jun '05

22 Jun '05

Dear all, Could anyone please tell me where to download the source code of the MAD plug-in for winamp? Thanks in advance. Best regards, Sherwin

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