Hi,

I'm trying to convert a aac file (retrieved from a online radio
station, using fPls) into a mp3 file (to listen to it on my MP3
player).
I've compiled 'mplayer' and 'ffmpeg' (on Windows, using cygwin).
I can do the following basic operations:
1. Listen to the aac file using 'mplayer' (I assume then the aac
decoder is working, right?)
2. Convert a mp3 to another mp3 file using 'ffmpeg' (I assume then
the mp3 coder is working, right?)
3. Convert a mp3 to a aac file
However when trying to:
4. Convert a aac to mp3 file, I get this error:
"Unsupported codec (id=86018) for input stream #0.0"

I pasted below samples for every the above operations. I guess the
answer is in the output these utilities shows, but I cannot decode it
myself.
Any idea what's wrong and what diagnotics/testing I can do more?

Many thanks,
Petru

1. Listen to a aac file is working fine:
-----------------------------------------------------
$> mplayer /tmp/radio/input.aac
MPlayer dev-SVN-r19147-3.4.4 (C) 2000-2006 MPlayer Team
CPU: AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ (Family: 15, Model:
35, Stepping: 2)
CPUflags: MMX: 1 MMX2: 1 3DNow: 1 3DNow2: 1 SSE: 1 SSE2: 1
Compiled for x86 CPU with extensions: MMX MMX2 3DNow 3DNowEx SSE SSE2

Playing /tmp/radio/input.aac.
libavformat file format detected.
==========================================================================
Opening audio decoder: [faad] AAC (MPEG2/4 Advanced Audio Coding)
AUDIO: 44100 Hz, 2 ch, s16le, 35.5 kbit/2.51% (ratio: 4435->176400)
Selected audio codec: [faad] afm: faad (FAAD AAC (MPEG-2/MPEG-4 Audio)
decoder)
==========================================================================
AO: [win32] 44100Hz 2ch s16le (2 bytes per sample)
Video: no video
Starting playback...
A: 2.7 (02.6) of 1233.4 (20:33.4) 4.1%

2. Converting a mp3 to another mp3 file is also working (128Kb => 64Kb)
---------------------------------------------------------------------------------------------------------
$> ffmpeg -i /tmp/radio/input.mp3 -acodec mp3 -ac 2 -ab 64
/tmp/radio/test.mp3
FFmpeg version SVN-r5804, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-mp3lame --enable-faac
libavutil version: 49.0.0
libavcodec version: 51.10.0
libavformat version: 50.5.0
built on Jul 23 2006 02:11:46, gcc: 3.4.4 (cygming special) (gdc
0.12, using dmd 0.125)
Input #0, mp3, from '/tmp/radio/input.mp3':
Duration: 00:02:40.7, start: 0.000000, bitrate: 128 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, 128 kb/s
File '/tmp/radio/test.mp3' already exists. Overwrite ? [y/N] y
Output #0, mp3, to '/tmp/radio/test.mp3':
Stream #0.0: Audio: mp3, 44100 Hz, stereo, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
size= 1256kB time=160.8 bitrate= 64.0kbits/s
video:0kB audio:1256kB global headers:0kB muxing overhead 0.000000%

3. Convert a mp3 to a aac file is fine
----------------------------------------------
$> ffmpeg -i /tmp/radio/input.mp3 -acodec aac /tmp/radio/output.aac
FFmpeg version SVN-r5804, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-mp3lame --enable-faac
libavutil version: 49.0.0
libavcodec version: 51.10.0
libavformat version: 50.5.0
built on Jul 23 2006 02:11:46, gcc: 3.4.4 (cygming special) (gdc
0.12, using dmd 0.125)
Input #0, mp3, from '/tmp/radio/input.mp3':
Duration: 00:02:40.7, start: 0.000000, bitrate: 128 kb/s
Stream #0.0: Audio: mp3, 44100 Hz, stereo, 128 kb/s
Output #0, adts, to '/tmp/radio/output.aac':
Stream #0.0: Audio: aac, 44100 Hz, stereo, 64 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Press [q] to stop encoding
size= 1255kB time=160.6 bitrate= 64.0kbits/s
video:0kB audio:1255kB global headers:0kB muxing overhead 0.000000%

4. Converting the aac input file to a mp3 file fails!!
------------------------------------------------------------------------
$> ffmpeg -i /tmp/radio/input.aac -acodec mp3 -ac 2 -ab 128
/tmp/radio/test.mp3
FFmpeg version SVN-r5804, Copyright (c) 2000-2004 Fabrice Bellard
configuration: --enable-mp3lame --enable-faac
libavutil version: 49.0.0
libavcodec version: 51.10.0
libavformat version: 50.5.0
built on Jul 23 2006 02:11:46, gcc: 3.4.4 (cygming special) (gdc
0.12, using dmd 0.125)
Input #0, aac, from '/tmp/radio/input.aac':
Duration: 00:20:33.4, start: 0.000000, bitrate: 35 kb/s
Stream #0.0: Audio: 0x0000, 22050 Hz, stereo, 35 kb/s
File '/tmp/radio/test.mp3' already exists. Overwrite ? [y/N] y
Output #0, mp3, to '/tmp/radio/test.mp3':
Stream #0.0: Audio: mp3, 22050 Hz, stereo, 128 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Unsupported codec (id=86018) for input stream #0.0

(E-Mail Removed) wrote (in part):

> I'm trying to convert a aac file (retrieved from a online radio
> station, using fPls) into a mp3 file (to listen to it on my MP3
> player).
> I've compiled 'mplayer' and 'ffmpeg' (on Windows, using cygwin)...

Mplayer in MS-Windows will use the Windows codecs already
installed in your system. ffmpeg will not. I see that libfaad2
is not listed in your sample output. The following links say
that you must recompile ffmpeg with --enable-faad to include
libfaad2 for aac decoding, and explain how to do that:

<http://lists.mplayerhq.hu/pipermail/ffmpeg-user/2006-April/002746.html>
<http://forum.rivavx.com/viewtopic.php?t=96&view=next&sid=ece5c4b38ae70af86f37da198151ba2b>

I prefer to use "faad 2" for command-line AAC decoding (faac is
an aac encoder, faad is the matching decoder). You can download
a precompiled Win32 binary of faad 2 at
http://rarewares.org/aac.html
For source code or more information about faac and faad, see
http://www.audiocoding.com
http://sourceforge.net/projects/faac

Thanks for help!
I run in other problems when tryin to compile faad2 library:

1. I've dowloaded faad2 from here:

http://prdownloads.sourceforge.net/f...ar.gz?download

2. Expanded and configured it (got a warning here!!!):
$> tar xvfz faad2-2.0.tar.gz
//..
$> cd faad2
$> ./bootstrap
configure.in:38: warning: underquoted definition of
MY_CHECK_TYPEDEF_FROM_INCLUDE
run info '(automake)Extending aclocal'
or see
http://sources.redhat.com/automake/a...ending-aclocal
configure.in: installing `./install-sh'
configure.in: installing `./missing'
common/mp4ff/Makefile.am: installing `./depcomp'
plugins/xmms/src/Makefile.am: installing `./compile'
Makefile.am: installing `./INSTALL'
Ready to run ./configure
$>./configure
//...

3. Build the lib (got the first error):
$> make
Makefile:624: *** missing separator. Stop.

The line 624 is
//...
632> rpm: Makefile
624> make dist
625> $(RPMBUILD) -ta $(PACKAGE)-$(VERSION).tar.gz
626> rm $(PACKAGE)-$(VERSION).tar.gz
4. I manually removed the whole 'rpm' target above, and tried to 'make'
again, but got an error about "conflicting types for 'lrintf'...

$> make
make all-recursive
make[1]: Entering directory `/usr/src/faad2'
Making all in libfaad
make[2]: Entering directory `/usr/src/faad2/libfaad'
if /bin/sh ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I.
-I. -I.. -g -O2 -MT bits.lo -MD -MP -MF ".deps
/bits.Tpo" -c -o bits.lo bits.c; \
then mv -f ".deps/bits.Tpo" ".deps/bits.Plo"; else rm -f
".deps/bits.Tpo"; exit 1; fi
mkdir .libs
gcc -DHAVE_CONFIG_H -I. -I. -I.. -g -O2 -MT bits.lo -MD -MP -MF
..deps/bits.Tpo -c bits.c -DDLL_EXPORT -DPIC -o .libs/b
its.o
In file included from common.h:347,
from bits.c:28:
/usr/include/math.h:278: error: conflicting types for 'lrintf'
common.h:314: error: previous definition of 'lrintf' was here
make[2]: *** [bits.lo] Error 1
make[2]: Leaving directory `/usr/src/faad2/libfaad'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/usr/src/faad2'
make: *** [all] Error 2


Now I'm stuck again.
Any idea?

Regards,
Petru





Mark Blain wrote:
> (E-Mail Removed) wrote (in part):
>
> > I'm trying to convert a aac file (retrieved from a online radio
> > station, using fPls) into a mp3 file (to listen to it on my MP3
> > player).
> > I've compiled 'mplayer' and 'ffmpeg' (on Windows, using cygwin)...
>
> Mplayer in MS-Windows will use the Windows codecs already
> installed in your system. ffmpeg will not. I see that libfaad2
> is not listed in your sample output. The following links say
> that you must recompile ffmpeg with --enable-faad to include
> libfaad2 for aac decoding, and explain how to do that:
>
> <http://lists.mplayerhq.hu/pipermail/ffmpeg-user/2006-April/002746.html>
> <http://forum.rivavx.com/viewtopic.php?t=96&view=next&sid=ece5c4b38ae70af86f37da198151ba2b>
>
> I prefer to use "faad 2" for command-line AAC decoding (faac is
> an aac encoder, faad is the matching decoder). You can download
> a precompiled Win32 binary of faad 2 at
> http://rarewares.org/aac.html
> For source code or more information about faac and faad, see
> http://www.audiocoding.com
> http://sourceforge.net/projects/faac

Rich_on 23-Jul-2006, (E-Mail Removed) wrote:

> I'm trying to convert a aac file (retrieved from a online radio
> station, using fPls) into a mp3 file (to listen to it on my MP3
> player).

<snip>

Might not be the way you want to go, however Switch

http://www.nch.com.au/switch/faq.html

will convert .aac format.

--
rich

Hey,

"Switch" is great: I installed it, and it's working just fine. Thanks
for sharing this info with me.

However I need to do the convertion from the command-line. I plan to
get the online radio station file, convert it and transfer it to my MP3
player using a script (that run overnight).
With Switch (GUI) I will have to click-click-click forever.
For that reason I chose "ffmpeg". But now I'm stuck compiling the faad2
on Cygwin.
Any idea?

Best regards,
Petru

rich wrote:
> Rich_on 23-Jul-2006, (E-Mail Removed) wrote:
>
> > I'm trying to convert a aac file (retrieved from a online radio
> > station, using fPls) into a mp3 file (to listen to it on my MP3
> > player).
>
> <snip>
>
> Might not be the way you want to go, however Switch
>
> http://www.nch.com.au/switch/faq.html
>
> will convert .aac format.
>
> --
> rich

Rich_on 24-Jul-2006, (E-Mail Removed) wrote:

> "Switch" is great: I installed it, and it's working just fine. Thanks
> for sharing this info with me.
>
> However I need to do the convertion from the command-line. I plan to
> get the online radio station file, convert it and transfer it to my MP3
> player using a script (that run overnight).
> With Switch (GUI) I will have to click-click-click forever.
> For that reason I chose "ffmpeg". But now I'm stuck compiling the faad2
> on Cygwin.
> Any idea?
<snipped a bit>

I only once needed to convert a .aac file which was when I found switch.

You say that you are compiling faad - why not use the already compiled faad
from rarewares.org that Mark Blain pointed out.
You would have to go .aac to .wav then use the output and convert to .mp3
I have just been looking at Super(c) from eRightsoft which uses ffmpeg and
will convert much multimedia all ways, but not .aac to .mp3. Perhaps it is
not posible?


--
rich

Probably I don't know how these things work: I was thinking by
compiling faad2 I would end up with some header files (*.h) and
libraries (*.a) that ffmpeg would use (and it seems it looks for when
configure with the --enable-faad flag)
Please correct me if I'm wrong.

>From the website http://rarewares.org/aac.html I can download the
faad.exe file or the libfaad.dll file. I'm not sure how to use them.

What I want is to be able to configure ffmpeg to use faad:
$> ./configure --enable-faad ...

Best regards,
Petru


rich wrote:
> Rich_on 24-Jul-2006, (E-Mail Removed) wrote:
>
> > "Switch" is great: I installed it, and it's working just fine. Thanks
> > for sharing this info with me.
> >
> > However I need to do the convertion from the command-line. I plan to
> > get the online radio station file, convert it and transfer it to my MP3
> > player using a script (that run overnight).
> > With Switch (GUI) I will have to click-click-click forever.
> > For that reason I chose "ffmpeg". But now I'm stuck compiling the faad2
> > on Cygwin.
> > Any idea?
> <snipped a bit>
>
> I only once needed to convert a .aac file which was when I found switch.
>
> You say that you are compiling faad - why not use the already compiled faad
> from rarewares.org that Mark Blain pointed out.
> You would have to go .aac to .wav then use the output and convert to .mp3
> I have just been looking at Super(c) from eRightsoft which uses ffmpeg and
> will convert much multimedia all ways, but not .aac to .mp3. Perhaps it is
> not posible?
>
>
> --
> rich

I dowloaded the percompiled faad utility and tried to get the *.wav
from the *.aac file. I get this error:

$> faad -o test.wav KissMusic.aac
*********** Ahead Software MPEG-4 AAC Decoder V2.1 beta
******************
Build: Jul 9 2004
Copyright 2002-2004: Ahead Software AG
http://www.audiocoding.com
Floating point version
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License.

**************************************************************************
KissMusic.aac file info:
RAW
Error: Invalid number of channels

Any idea what's wrong?

Regards,
Petru

rich wrote:
> Rich_on 24-Jul-2006, (E-Mail Removed) wrote:
>
> > "Switch" is great: I installed it, and it's working just fine. Thanks
> > for sharing this info with me.
> >
> > However I need to do the convertion from the command-line. I plan to
> > get the online radio station file, convert it and transfer it to my MP3
> > player using a script (that run overnight).
> > With Switch (GUI) I will have to click-click-click forever.
> > For that reason I chose "ffmpeg". But now I'm stuck compiling the faad2
> > on Cygwin.
> > Any idea?
> <snipped a bit>
>
> I only once needed to convert a .aac file which was when I found switch.
>
> You say that you are compiling faad - why not use the already compiled faad
> from rarewares.org that Mark Blain pointed out.
> You would have to go .aac to .wav then use the output and convert to .mp3
> I have just been looking at Super(c) from eRightsoft which uses ffmpeg and
> will convert much multimedia all ways, but not .aac to .mp3. Perhaps it is
> not posible?
>
>
> --
> rich

I want to also specify that the input file is playable by both mplayer
and WinAmp.
You can see here the 'Properties' screen from WinAmp for this specific
input file:

http://www.petrum.net/tmp/WinAmp-screen.JPG

Regards,
Petru

petru.margin...@gmail.com wrote:
> I dowloaded the percompiled faad utility and tried to get the *.wav
> from the *.aac file. I get this error:
>
> $> faad -o test.wav KissMusic.aac
> *********** Ahead Software MPEG-4 AAC Decoder V2.1 beta
> ******************
> Build: Jul 9 2004
> Copyright 2002-2004: Ahead Software AG
> http://www.audiocoding.com
> Floating point version
> This program is free software; you can redistribute it and/or modify
> it under the terms of the GNU General Public License.
>
> **************************************************************************
> KissMusic.aac file info:
> RAW
> Error: Invalid number of channels
>
> Any idea what's wrong?
>
> Regards,
> Petru
>
> rich wrote:
> > Rich_on 24-Jul-2006, (E-Mail Removed) wrote:
> >
> > > "Switch" is great: I installed it, and it's working just fine. Thanks
> > > for sharing this info with me.
> > >
> > > However I need to do the convertion from the command-line. I plan to
> > > get the online radio station file, convert it and transfer it to my MP3
> > > player using a script (that run overnight).
> > > With Switch (GUI) I will have to click-click-click forever.
> > > For that reason I chose "ffmpeg". But now I'm stuck compiling the faad2
> > > on Cygwin.
> > > Any idea?
> > <snipped a bit>
> >
> > I only once needed to convert a .aac file which was when I found switch.
> >
> > You say that you are compiling faad - why not use the already compiled faad
> > from rarewares.org that Mark Blain pointed out.
> > You would have to go .aac to .wav then use the output and convert to .mp3
> > I have just been looking at Super(c) from eRightsoft which uses ffmpeg and
> > will convert much multimedia all ways, but not .aac to .mp3. Perhaps it is
> > not posible?
> >
> >
> > --
> > rich

Rich_on 25-Jul-2006, (E-Mail Removed) wrote:

<snip>
> > > > However I need to do the convert ion from the command-line. I plan
> > > >to
> > > > get the online radio station file, convert it and transfer it to my
> > > > MP3
> > > > player using a script (that run overnight).
> > > > With Switch (GUI) I will have to click-click-click forever.
> > > > For that reason I chose "ffmpeg". But now I'm stuck compiling the
> > > > faad2
> > > > on Cygwin.
> > > > Any idea?

> > > <snipped a bit>
> > >
> > > I only once needed to convert a .aac file which was when I found
> > > switch.
> > >
> > > You say that you are compiling faad - why not use the already compiled
> > > faad
> > > from rarewares.org that Mark Blain pointed out.
> > > You would have to go .aac to .wav then use the output and convert to
> > > .mp3
> > > I have just been looking at Super(c) from eRightsoft which uses ffmpeg
> > > and
> > > will convert much multimedia all ways, but not .aac to .mp3. Perhaps
> > > it is
> > > not possible?

<moved a bit to keep the thread>

> I want to also specify that the input file is playable by both mplayer
> and WinAmp.
> You can see here the 'Properties' screen from WinAmp for this specific
> input file:
>
> http://www.petrum.net/tmp/WinAmp-screen.JPG
>


> > I dowloaded the percompiled faad utility and tried to get the *.wav
> > from the *.aac file. I get this error:
> >
> > $> faad -o test.wav KissMusic.aac
> > *********** Ahead Software MPEG-4 AAC Decoder V2.1 beta
> > ******************
> > Build: Jul 9 2004
> > Copyright 2002-2004: Ahead Software AG
> > http://www.audiocoding.com
> > Floating point version
> > This program is free software; you can redistribute it and/or modify
> > it under the terms of the GNU General Public License.
> >
> > **************************************************************************
> > KissMusic.aac file info:
> > RAW
> > Error: Invalid number of channels
> >
> > Any idea what's wrong?

I dont think that I can add much more as I do not normally need to convert
..aac files
however I have tried a few simple tests to see what works and does not.

1. faad
Not having any .aac files I had to make one so I could decode it.
Using belight - a front end for besweet, and the aac conversion option,
there is the first problem - what codec?
There is a faac codec which will make a .aac file which faad will decode.
There is a nero codec which make a .mp4 file which produces the error you
get - RAW - error: invalid number of channels.
I do not know how to fix this problem.

2. Super(c) converter
I mentioned this before because I thought it might convert a .aac file but
didn't.
That was on my new-ish computer which I use for video and graphics.
I installed Super on my old internet box (this one) and for some reason it
will convert the .mp4 file ( the RAW file) but still does not convert the
..aac file.
The point of this is: Super is only a front end for ffmpeg and other things.
By stopping Super half way through a .mp4 to .mp3 conversion I was able to
"rescue" the avisynth script it used.

this is the content of sample.avs
Clip=DirectShowSource("E:\sample.mp4", fps=25, convertfps=true)
ConvertToYV12(Clip)
Subtitle("Me ; )", font="verdana", size=14, text_color=$1afffa,
halo_color=$1f4afa, align=3)

As you see it uses directx in the conversion.
You can use this script as input to ffmpeg bypassing the Superfront end.

ffmpeg -i sample.avs -acodec mp3 test.mp3

Suggestion: Get Super(c) from eRightSoft and see if it will convert your
..aac file, try renaming to .mp4 as .aac is not a valid imput.
If it does, taskkill Super and retrieve the avisynth script from your temp
folder.

Suggestion: Try one of the video forums, possibly doom9.org, they know all
there is to know about ffmpeg. Have a look at avisynth and how it interacts
with ffmpeg.

Suggestion: stop top posting.

Best of luck

--
rich