Nabble - MAD

Re: Converting fixed point numbers

2008-07-12T13:07:04Z

On Jul 12, 2008, at 10:56 AM, Reiner wrote:

> madplay uses the following in audio_linear_round
>
> /* quantize and scale */
> return sample >> (MAD_F_FRACBITS + 1 - bits);
>
> if bits is 16 (unsigned short e,g) we will shift 13 bits, whats
> about the 3 MSB bits ???
>
> With this I think only the first digit of the whole part is taken,
> what is with the sign and the other 2 digits from the whole part,
> are they never used?

The whole part is used to represent samples that are above full scale
[-1.0, 1.0) but madplay clips these prior to scaling, so at this point
the whole part just contains copies of the sign bit.

--
Rob Leslie
rob@...

From forum: Mad - Dev

Converting fixed point numbers

2008-07-12T10:56:50Z

Hello,

madplay uses the following in audio_linear_round

/* quantize and scale */
return sample >> (MAD_F_FRACBITS + 1 - bits);

if bits is 16 (unsigned short e,g) we will shift 13 bits, whats about
the 3 MSB bits ???

With this I think only the first digit of the whole part is taken, what
is with the sign and the other 2 digits from the whole part, are they
never used?

Thanks

Reiner

From forum: Mad - Dev

Re: libmad newbie questions

2008-06-27T17:17:34Z

Alan Wolfe wrote:
> Heya!

First off, IANAL but more/less a casual observer. I'd
encourage you to secure professional legal advice to
definitively answer your questions.

> I was wondering; since mp3 is a closed format that you need to pay
> royalties for, how does libmad get around this?

It doesn't. Technically you are on the hook to reconcile
your license liability with the holder of such in the
geography your product will be marketed:

http://en.wikipedia.org/wiki/MP3#Licensing_and_patent_issues

That said enforcement has been selective and you conceivably
could choose your own comfortability level relative to your
situation.

> Also I was looking at the libmad page and noticed it said this under
> licensing:

That is a separate license issue of Underbit's MAD decoder
software. In a nutshell if you GPL your MAD enabled
application (possibly exposing your own IP) you may
distribute the aggregate under the GPL. Otherwise you'll
need to contact the Underbit folks to negotiate an
alternate license agreement.

> 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.

If you are not bound to mp3 as a compression format, others
such as Ogg/Vorbis (well Tremor) may interest being a
license-free public domain format and offering compression/
quality benefits beyond that available from mp3.

http://vorbis.com/faq/#flic

-john

--
john.cooper@...

From forum: Mad - User

libmad newbie questions

2008-06-27T15:45:56Z

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, 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!

From forum: Mad - User

libmad documentation project

2008-06-09T07:46:21Z

Hi,

I recently started working on a small audio processing project. I chose
to use mad library for dealing with mp3 files.

Since I needed to study it for getting enough understanding of it, and
the package TODO file calls for contributors to documentation project, I
tried to make my notes useful for the programmers community.

One thing I like in java programming is the documentation facility,
which generates browseable API documentation directly from comments
inserted in source files.

The same functionality is available for C/C++ and other languages, with
Doxygen (open-source) program, so I decided to use it.

You'll find all details and files here:
http://m.baert.free.fr/contrib/docs/libmad/

Your comments and suggestions are welcome.
TIA for them.

--
Best regards,
Michelle Baert

From forum: Mad - Dev

Re: madplayer cross-compile on ARM11 compilation error "impossible constraint in 'asm'"

2008-05-21T06:05:11Z

Thank you so much for your help, Rob!

Regards,
Jeff

> From: rob@...

> To: mad-dev@...
> Date: Sun, 18 May 2008 19:40:12 -0700
> Subject: Re: [mad-dev] madplayer cross-compile on ARM11 compilation error "impossible constraint in 'asm'"
>
> On May 18, 2008, at 12:33 AM, Jianfeng Li wrote:
> > I am now trying to cross-compile the madplayer on an arm11 based
> > embedded system. The compiler I am using is from montavista
> > mobilinux 5.0.
> >
> > When I tried to build the madplay, for the file resample.c, I got 2
> > errors on line 89 and 104. The error information is "impossible
> > constraint in 'asm'. When I tried to build the libmad, I also met
> > the same error. But it seems that during compilation if "FPM-ARM"
> > option is enabled then the error is solved. But here I checked the
> > Makefile, there is no such option of "FPM" as in the Makefile of
> > libmad.
> ! >
> > So can anybody help to tell me why this error happens, it is due to
> > the compiler's constraint/requirement? How should I try to solve
> > this error? Is "trying to enable the FPM" a good option to try?
>
> Running the libmad `configure' script is the usual way to cause an
> appropriate FPM macro to be defined. You can force the selection using
> the --enable-fpm option (e.g. --enable-fpm=arm) in case the
> automatically chosen default is inappropriate for some reason.
>
> Note that the FPM selection is carried into the mad.h header file
> which is used by madplay et al. When cross-compiling, the resulting
> mad.h could be different from the build system's mad.h so it's
> important to check your include paths.
>
> --
> Rob Leslie
> rob@...
>

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From forum: Mad - Dev

Re: Stream Decode

2008-05-21T05:58:24Z

Alexey V Paramonov wrote:
> Hi!
>
> I have the following question - is it possible to decode a stream using
> MAD?

Yes, of course.

> 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.

Haven't looked at madplay (etc..) in years but originally
it was mmap()'ing an mp3 file into memory which works
fine assuming the mmap() region could grow for as long as
you need to decode the respective file.

If rather you are trying to decode into a fixed buffer
you'll need to do some adjustment when your decode nears
the end of the static buffer. This means detecting when
you don't have enough data left at the buffer tail to
do a valid frame decode (ie: mad_frame_decode() returns
MAD_ERROR_BUFLEN). In that case you need to copy the
partial frame at the buffer's tail, back at/near the head
of the buffer (depending on your stream management logic)
and readjust libmad's notion of where it is via a
mad_stream_buffer() call.

> 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
>
> }
>
> }

I really don't understand what you are trying to
accomplish via the back-to-back decode/encode but
I suppose that is unrelated to your question.

> Is it possible to decode a stream without theese "edge effects" ?

Yes but it can be a little tricky to get right in the
case you allow a user to manipulate the stream pointer
and perform arbitrary seeks within the stream.

> May be I should not re-initialize MAD buffer after each read in some way?

If you are altering the [memory address] seek point
within the current buffer you'll need to reset the
decoder's notion of decode location via a mad_stream_buffer()
call.

-john

--
john.cooper@...

From forum: Mad - User

Stream Decode

2008-05-21T05:07:17Z

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.

From forum: Mad - User

Stream Decode

2008-05-20T10:29:20Z

From forum: Mad - Dev

Re: madplayer cross-compile on ARM11 compilation error "impossible constraint in 'asm'"

2008-05-18T19:40:12Z

On May 18, 2008, at 12:33 AM, Jianfeng Li wrote:

> I am now trying to cross-compile the madplayer on an arm11 based
> embedded system. The compiler I am using is from montavista
> mobilinux 5.0.
>
> When I tried to build the madplay, for the file resample.c, I got 2
> errors on line 89 and 104. The error information is "impossible
> constraint in 'asm'. When I tried to build the libmad, I also met
> the same error. But it seems that during compilation if "FPM-ARM"
> option is enabled then the error is solved. But here I checked the
> Makefile, there is no such option of "FPM" as in the Makefile of
> libmad.
>
> So can anybody help to tell me why this error happens, it is due to
> the compiler's constraint/requirement? How should I try to solve
> this error? Is "trying to enable the FPM" a good option to try?

Running the libmad `configure' script is the usual way to cause an
appropriate FPM macro to be defined. You can force the selection using
the --enable-fpm option (e.g. --enable-fpm=arm) in case the
automatically chosen default is inappropriate for some reason.

Note that the FPM selection is carried into the mad.h header file
which is used by madplay et al. When cross-compiling, the resulting
mad.h could be different from the build system's mad.h so it's
important to check your include paths.

--
Rob Leslie
rob@...

From forum: Mad - Dev

madplayer cross-compile on ARM11 compilation error "impossible constraint in 'asm'"

2008-05-18T00:33:50Z

Hi, All,

I am now trying to cross-compile the madplayer on an arm11 based embedded system. The compiler I am using is from montavista mobilinux 5.0.

When I tried to build the madplay, for the file resample.c, I got 2 errors on line 89 and 104. The error information is "impossible constraint in 'asm'. When I tried to build the libmad, I also met the same error. But it seems that during compilation if "FPM-ARM" option is enabled then the error is solved. But here I checked the Makefile, there is no such option of "FPM" as in the Makefile of libmad.

So can anybody help to tell me why this error happens, it is due to the compiler's constraint/requirement? How should I try to solve this error? Is "trying to enable the FPM" a good option to try?

Many thanks in advance!

Regards,
Jeffen

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From forum: Mad - Dev

decoding error while using minimad.c

2008-04-20T19:16:35Z

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！

From forum: Mad - User

Minimad.c

2008-02-27T01:00:07Z

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

From forum: Mad - User

Re: Initite loop bug in libid3tag-0.15.0b

2008-01-30T04:18:22Z

Rahul Banerjee wrote:
> Hi,
>
> I am new to the usage of this mad library. I have a question regarding
> this. Is the fast forward playback of mp3 is supported with this library?

Well yes, sort of. libmad will simply decode a stream.
The burden is placed on the user however to direct the
decoder to whatever bits you'd like to decode. So to
implement fast fwd, trick play, etc.. you'll need to
point the decoder at the bits you'd like to render which
say constitute snippets of audio to be rendered from a
stream in fast forward.

This requires repositioning the stream pointer ahead in
the stream and rendering frames until you have an
error free decode and are synchronized. Render as much
of decoded output to the audio device for a single hop,
reposition the decoder in the stream and repeat.

-john

--
john.cooper@...

From forum: Mad - Dev

Re: Initite loop bug in libid3tag-0.15.0b

2008-01-29T18:49:36Z

Hi,

I am new to the usage of this mad library. I have a question regarding this. Is the fast forward playback of mp3 is supported with this library?

Thanks and Regards,

Rahul Banerjee

On 1/13/08, Kentaro Oda <odaken@...> wrote:

Hi, for the maintainer libid3tag.

I found an infinite loop bug in libid3tag-0.15.0b library, which
causes memory overflow.

The problem occurs when parsing an ID3_FIELD_TYPE_STRINGLIST field,
specifically when data to be parsed is ended with '\0'.
In this case, **ptr == 0, but the condition end - *ptr is 1 so loop
continues infinitely.

*** field.c     2003-04-19 09:14:33.000000000 +0900
--- field-patched.c     2008-01-13 16:08:22.000000000 +0900
***************
*** 291,297 ****

       end = *ptr + length;

!       while (end - *ptr > 0) {
       ucs4 = id3_parse_string(ptr, end - *ptr, *encoding, 0);
       if (ucs4 == 0)
         goto fail;
--- 291,297 ----

       end = *ptr + length;

!       while (end - *ptr > 0 && **ptr != '\0') {
       ucs4 = id3_parse_string(ptr, end - *ptr, *encoding, 0);
       if (ucs4 == 0)
         goto fail;

--
Kentaro Oda

From forum: Mad - Dev

Initite loop bug in libid3tag-0.15.0b

2008-01-12T23:17:36Z

Hi, for the maintainer libid3tag.

I found an infinite loop bug in libid3tag-0.15.0b library, which
causes memory overflow.

The problem occurs when parsing an ID3_FIELD_TYPE_STRINGLIST field,
specifically when data to be parsed is ended with '\0'.
In this case, **ptr == 0, but the condition end - *ptr is 1 so loop
continues infinitely.

*** field.c 2003-04-19 09:14:33.000000000 +0900
--- field-patched.c 2008-01-13 16:08:22.000000000 +0900
***************
*** 291,297 ****

end = *ptr + length;

! while (end - *ptr > 0) {
ucs4 = id3_parse_string(ptr, end - *ptr, *encoding, 0);
if (ucs4 == 0)
goto fail;
--- 291,297 ----

end = *ptr + length;

! while (end - *ptr > 0 && **ptr != '\0') {
ucs4 = id3_parse_string(ptr, end - *ptr, *encoding, 0);
if (ucs4 == 0)
goto fail;

--
Kentaro Oda

From forum: Mad - Dev

madplay: How to select a specific ALSA card

2008-01-08T10:13:45Z

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

From forum: Mad - Dev

[mad-dev] madplay: How to select a specific ALSA card

2008-01-08T10:13:45Z

From forum: Mad - User

(no subject)

2008-01-03T13:09:15Z

unsubscribe

From forum: Mad - User

madplay: How to select a specific ALSA card

2008-01-03T07:22:21Z

From forum: Mad - User

Re: dual-thread libmad

2007-12-08T16:12:52Z

Hi John

Thanks for the reply.

I actually don't want any mixing at all. I want MP3(A) to be played on
channel A and MP3(B) to be played on channel B. Something is occurring,
which appears to be towards the output stage of my libmad, that is
causing the A and B samples to mix together. So I am looking for some
global buffer that is being using by both A and B or something like that
which the RTOS is mixing between both threads.

Steve Spano, President
Finger Lakes Engineering

-----Original Message-----
From: john cooper [mailto:john.cooper@...]
Sent: Saturday, December 08, 2007 9:04 AM
To: Steve Spano
Cc: mad-user@...
Subject: Re: [mad-user] dual-thread libmad

Steve Spano wrote:
> Hi John,
>
> Thanks for the reply.
>
> I did more tracking into this, the mad_decoder_buffer creates the
> instances on the frame/stream/sync data. I already created two
instances
> of the mad_decoder_buffer so I could link them to each RTOS thread.
>
> It looks like the samples are getting mixed up at the output stage. It
> turns out that one of the MP3s playing is a 1-channel and the other
MP3
> is a 2-channel. Any thoughts perhaps on how mixing of sample data,
> between RTOS tasks, could occur on the output stages?

You'll need to make output the data rates agree potentially
converting one to the other. And you will need to
synchronize and mix the two streams before they hit the
audio device. This should be relatively straightforward
but you may need to scale the data to avoid clipping during
the mix operation.

-john

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.
>
> It is built into the API. You'll need to keep separate
> frame/stream/synth data structures for each instance of
> a decode operation. Sounds like you are rendering to the
> same set of structures multiplexed by the task scheduler.
>
> -john
>

--
john.cooper@...

From forum: Mad - User

Re: dual-thread libmad

2007-12-08T06:04:13Z

Steve Spano wrote:

> Hi John,
>
> Thanks for the reply.
>
> I did more tracking into this, the mad_decoder_buffer creates the
> instances on the frame/stream/sync data. I already created two instances
> of the mad_decoder_buffer so I could link them to each RTOS thread.
>
> It looks like the samples are getting mixed up at the output stage. It
> turns out that one of the MP3s playing is a 1-channel and the other MP3
> is a 2-channel. Any thoughts perhaps on how mixing of sample data,
> between RTOS tasks, could occur on the output stages?

You'll need to make output the data rates agree potentially
converting one to the other. And you will need to
synchronize and mix the two streams before they hit the
audio device. This should be relatively straightforward
but you may need to scale the data to avoid clipping during
the mix operation.

-john

> I figured this part out, that it is occurring at the output stage,
> Friday ~ 5pm. I have to look at it some more.
>
> Thanks
>
> Steve Spano, President
> Finger Lakes Engineering
>
>
> -----Original Message-----
> From: john cooper [mailto:john.cooper@...]
> Sent: Friday, December 07, 2007 10:09 AM
> To: Steve Spano
> Cc: mad-user@...; john cooper
> Subject: Re: [mad-user] dual-thread libmad
>
> Steve Spano wrote:
>
>> 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.
>
> It is built into the API. You'll need to keep separate
> frame/stream/synth data structures for each instance of
> a decode operation. Sounds like you are rendering to the
> same set of structures multiplexed by the task scheduler.
>
> -john
>

--
john.cooper@...

From forum: Mad - User

Re: dual-thread libmad

2007-12-07T19:23:23Z

Hi John,

Thanks for the reply.

I did more tracking into this, the mad_decoder_buffer creates the
instances on the frame/stream/sync data. I already created two instances
of the mad_decoder_buffer so I could link them to each RTOS thread.

It looks like the samples are getting mixed up at the output stage. It
turns out that one of the MP3s playing is a 1-channel and the other MP3
is a 2-channel. Any thoughts perhaps on how mixing of sample data,
between RTOS tasks, could occur on the output stages?

I figured this part out, that it is occurring at the output stage,
Friday ~ 5pm. I have to look at it some more.

Thanks

Steve Spano, President
Finger Lakes Engineering

-----Original Message-----
From: john cooper [mailto:john.cooper@...]
Sent: Friday, December 07, 2007 10:09 AM
To: Steve Spano
Cc: mad-user@...; john cooper
Subject: Re: [mad-user] dual-thread libmad

Steve Spano wrote:

> 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.

It is built into the API. You'll need to keep separate
frame/stream/synth data structures for each instance of
a decode operation. Sounds like you are rendering to the
same set of structures multiplexed by the task scheduler.

-john

--
john.cooper@...

From forum: Mad - User

Re: dual-thread libmad

2007-12-07T07:09:24Z

Steve Spano wrote:

> 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.

It is built into the API. You'll need to keep separate
frame/stream/synth data structures for each instance of
a decode operation. Sounds like you are rendering to the
same set of structures multiplexed by the task scheduler.

-john

--
john.cooper@...

From forum: Mad - User

dual-thread libmad

2007-12-07T05:07:05Z

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

From forum: Mad - User

Re: memory leaking in mad_layer_III?

2007-09-27T03:50:40Z

Rob Leslie wrote:
> On Sep 26, 2007, at 2:55 PM, Niels Roosen wrote:
>> I will post the relevant code tomorrow, should I do that as attachment
>> or inline?
>
> Either is fine unless it's long, in which case attachment is probably
> preferred.

You're not going to believe this, but I seem to have solved the problem.
While scrutinizing the relevant code for posting I discovered a rather
odd construct in my code (don't know why I didn't see that before) and
after some more investigation I discovered that I was lying earlier. In
the initialization function (which is also responsible for calling the
mad_*_finish() functions if necessary) the condition for actually
calling the cleanup functions was not working at all! This was because I
mixed up something during development and clearly never tested this
properly. So I fixed that and didn't get -any- references to lost memory
in libmad anymore \o/.

Anyway, to summarize: yet another case of tunnel vision, so you try to
explain your problem to someone else and during that process find out
what is going wrong. So thanks nevertheless, I guess I needed this :-)

So I humbly apologize for being such a retard. Nevertheless, mad rocks
my socks, it is fast and reliable unlike the mpg123 decoder which often
fails to synchronize frames, miserably failing to decode an entire
stream... :-)

Cheers,

Niels

--
"One of the most frightening things about your true nerd, for many
people, is not that he's socially inept - because everybody's been there
- but rather his complete lack of embarrassment about it." - Randy
Waterhouse in Cryptonomicon

n: Arnie / Arnicus | e: niels@...
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From forum: Mad - Dev

Re: memory leaking in mad_layer_III?

2007-09-26T16:53:11Z

On Sep 26, 2007, at 2:55 PM, Niels Roosen wrote:
> I will post the relevant code tomorrow, should I do that as
> attachment or inline?

Either is fine unless it's long, in which case attachment is probably
preferred.

--
Rob Leslie
rob@...

From forum: Mad - Dev

Re: memory leaking in mad_layer_III?

2007-09-26T15:55:04Z

Rob Leslie wrote:

> On Sep 26, 2007, at 3:40 AM, Niels Roosen wrote:
>> Am I missing something or could it be that using mad this way hasn't
>> been tested excessively enough and is it true that the memory
>> alllocated in mad_layer_III() on lines 2530 and 2538 (libmad version
>> 0.15.1b) is not freed properly? I would be delighted to either get
>> confirmation on this or some advice on how to solve this issue.
>
> The memory allocated in mad_layer_III() is freed in mad_frame_finish()
> and mad_stream_finish(); you should make sure these are definitely being
> called on the same structures you pass to mad_frame_decode().
>
> Since you say you are making these calls, I'm at a loss to offer any
> specific advice without seeing your code.

What I do is initialize the decoder, decode incoming data per frame
until the client disconnects, and then, when another client connects,
run the mad_stream_finish() and mad_frame_finish() on the pointers to
the relevant structures before running the initialization functions
again, in order to clean up any remaining data in the decoder. I took
this from some example code but unfortunately don't know what any more,
but I suppose that is how it should be done.

I will post the relevant code tomorrow, should I do that as attachment
or inline?

Regards,

Niels

--
"One of the most frightening things about your true nerd, for many
people, is not that he's socially inept - because everybody's been there
- but rather his complete lack of embarrassment about it." - Randy
Waterhouse in Cryptonomicon

n: Arnie / Arnicus | e: niels@...
w: http://okkernoot.net/ |
irc: irc.okkernoot.net #geek | icq: 31817478
aim: arniea a arniea | msn: arnie@...

http://jungletrain.net/ - 24/7 Drum and Bass Internet Radio

From forum: Mad - Dev

Re: memory leaking in mad_layer_III?

2007-09-26T10:03:51Z

On Sep 26, 2007, at 3:40 AM, Niels Roosen wrote:
> Am I missing something or could it be that using mad this way
> hasn't been tested excessively enough and is it true that the
> memory alllocated in mad_layer_III() on lines 2530 and 2538 (libmad
> version 0.15.1b) is not freed properly? I would be delighted to
> either get confirmation on this or some advice on how to solve this
> issue.

The memory allocated in mad_layer_III() is freed in mad_frame_finish
() and mad_stream_finish(); you should make sure these are definitely
being called on the same structures you pass to mad_frame_decode().

Since you say you are making these calls, I'm at a loss to offer any
specific advice without seeing your code.

--
Rob Leslie
rob@...

From forum: Mad - Dev

memory leaking in mad_layer_III?

2007-09-26T04:40:27Z

Hello,

In the past while I have been rounding up development for a streaming
server for the radio station I am part of. However, during the last
stress testing phase I noticed that memory usage slowly increases as
streaming clients disconnect and connect. At first I was convinced it
was some leak in my own code but valgrind didn't confirm that this was
happening. However, I noticed that there was quite some lost memory
originating in the mad_layer_III() function in layer3.c:

==5957== 7,701 bytes in 3 blocks are definitely lost in loss record 68 of 85
==5957== at 0x481B72C: malloc (in
/usr/lib/valgrind/x86-linux/vgpreload_memcheck.so)
==5957== by 0x485F1C1: mad_layer_III (in /usr/lib/libmad.so.0.2.1)
==5957== by 0x485429A: mad_frame_decode (in /usr/lib/libmad.so.0.2.1)
==5957== by 0x7775: decode_data (server.c:318)
==5957== by 0x9387: server (server.c:980)
==5957== by 0x4874164: pthread_start_thread (in /lib/libpthread-0.10.so)
==5957== by 0x4A176E9: clone (in /lib/libc-2.3.6.so)

==5957== 13,824 bytes in 3 blocks are definitely lost in loss record 71
of 85
==5957== at 0x481CE9E: calloc (in
/usr/lib/valgrind/x86-linux/vgpreload_memcheck.so)
==5957== by 0x485F215: mad_layer_III (in /usr/lib/libmad.so.0.2.1)
==5957== by 0x485429A: mad_frame_decode (in /usr/lib/libmad.so.0.2.1)
==5957== by 0x7775: decode_data (server.c:318)
==5957== by 0x9387: server (server.c:980)
==5957== by 0x4874164: pthread_start_thread (in /lib/libpthread-0.10.so)
==5957== by 0x4A176E9: clone (in /lib/libc-2.3.6.so)

One would assume this is just caused by memory not being free()'d up
when the program exits, but the longer the server has run, the more
memory was definitely lost, indicating that the memory loss originates here.

I use the mad library in such a way that I manually line up data and
decode this data per frame, which is required to ensure certain
functionality of the streaming server. I made sure that when I
reinitialize the decoder every time I run the mad_*_finish() operations
on all the structs I use before running mad_*_init() again. However, I
suspect that not all allocated memory is freed as it should.

Am I missing something or could it be that using mad this way hasn't
been tested excessively enough and is it true that the memory alllocated
in mad_layer_III() on lines 2530 and 2538 (libmad version 0.15.1b) is
not freed properly? I would be delighted to either get confirmation on
this or some advice on how to solve this issue. If more information is
required I will do my best to provide this.

Regards,

Niels Roosen

--
"One of the most frightening things about your true nerd, for many
people, is not that he's socially inept - because everybody's been there
- but rather his complete lack of embarrassment about it." - Randy
Waterhouse in Cryptonomicon

n: Arnie / Arnicus | e: niels@...
w: http://okkernoot.net/ |
irc: irc.okkernoot.net #geek | icq: 31817478
aim: arniea a arniea | msn: arnie@...

http://jungletrain.net/ - 24/7 Drum and Bass Internet Radio

From forum: Mad - Dev

Re: mad decoder sample rates

2007-09-07T11:39:04Z

Hi Rob,

Great - thanks! I see the call, something like set sample rate. It just
goes to a debug trace and I see it decoding a file at 22050. So we will
hook that into the silicon oscillator setup.

We are also going to run MAD on the Xilinx Microblaze later this month
as well.

I have another issue with compiling with the FPM_ARM; but I am working
on resolving that. Right now I am using FPM_DEFAULT.

So I may ask a few other questions. For the STR912 implementation, we
actually want to run two instances of MAD, in mono mode, and then have
one MP3 goto the left channel and the other the right; assuming the
sample rates match.

It doesn't look like MAD performs on the fly resampling; which is fine.
We may add it later.

Thanks

Steve Spano
FLE

-----Original Message-----
From: mad-user-bounces@...
[mailto:mad-user-bounces@...] On Behalf Of Rob Leslie
Sent: Friday, September 07, 2007 1:14 PM
To: mad-user@...
Subject: Re: [mad-user] mad decoder sample rates

On Sep 7, 2007, at 9:57 AM, Steve Spano wrote:
> 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.

MAD supports all MPEG audio sampling frequencies, namely
8000/11025/12000 (MPEG 2.5), 16000/22050/24000 (MPEG-2 LSF), and
32000/44100/48000 (MPEG-1) Hz.

Keep in mind the bitstream determines the sample rate you get from
MAD. If you want to output PCM at some other rate, you'll have to
resample it first.

--
Rob Leslie
rob@...

From forum: Mad - User

Re: mad decoder sample rates

2007-09-07T11:14:19Z

On Sep 7, 2007, at 9:57 AM, Steve Spano wrote:
> 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.

MAD supports all MPEG audio sampling frequencies, namely
8000/11025/12000 (MPEG 2.5), 16000/22050/24000 (MPEG-2 LSF), and
32000/44100/48000 (MPEG-1) Hz.

Keep in mind the bitstream determines the sample rate you get from
MAD. If you want to output PCM at some other rate, you'll have to
resample it first.

--
Rob Leslie
rob@...

From forum: Mad - User

mad decoder sample rates

2007-09-07T10:57:04Z

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@...

Finger Lakes Engineering

607-277-1614 x223

www.fl-eng.com

From forum: Mad - User

Re: Calculation error

2007-09-07T03:33:33Z

On Fri, Aug 24, 2007 at 12:16:12PM +0200, Tim Dylla wrote:
> I did some testing on my own implementation of mp3, and used MAD as
> reference.
> Therefore I generated a sine-sweep from 0Hz to 22kHz ( as far as this is
> possible ;) ) at 0dBFS, and 48kHz Samplerate. Then I encoded it with
> Lame for a single Channel MPEG-1 Layer III Bitstream, 48kHz Samplerate,
> 320kbps Bitrate. (I don't know the exact Lame-version and optionstring
> anymore, but I didn't use anything weird)

> I'm not a friend of posting large files on a mailing list without any
> agreement with the moderator, so if someone is interested in the
> bitstream and the resulting raw file, pleas feel free to send me a message.

Could the generation be done with common software? (sox? clm? nyqvist?)
That could make a sort of full automatic test system if scripted.

Also, a file like that might find a home on Wikimedia or Internet Archive.

--
To know what you prefer, instead of humbly saying Amen
to what the world tells you you ought to prefer,
is to have kept your soul alive.
-- Robert Louis Stevenson

From forum: Mad - Dev