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682 discussions

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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